update for FFmpeg 2.0.3

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
avutil/log: skip IO calls on empty strings
2013-12-24 03:22:00 +01:00 · 2013-12-24 01:05:47 +01:00 · 2013-12-24 01:05:47 +01:00 · 2013-12-24 01:05:47 +01:00 · 2013-12-24 01:05:47 +01:00 · 2013-12-24 01:05:47 +01:00
1582 changed files with 99132 additions and 58967 deletions
--- a/.gitignore
+++ b/.gitignore
@@ -6,6 +6,8 @@
 *.dylib
 *.exe
 *.exp
+*.gcda
+*.gcno
 *.h.c
 *.ilk
 *.lib
@@ -24,11 +26,13 @@
 /ffprobe
 /ffserver
 /config.*
+/coverage.info
 /version.h
 /doc/*.1
 /doc/*.3
 /doc/*.html
 /doc/*.pod
+/doc/config.texi
 /doc/avoptions_codec.texi
 /doc/avoptions_format.texi
 /doc/examples/decoding_encoding
@@ -43,6 +47,7 @@
 /doc/fate.txt
 /doc/doxy/html/
 /doc/print_options
+/lcov/
 /libavcodec/*_tablegen
 /libavcodec/*_tables.c
 /libavcodec/*_tables.h
@@ -52,6 +57,7 @@
 /tests/data/
 /tests/rotozoom
 /tests/tiny_psnr
+/tests/tiny_ssim
 /tests/videogen
 /tests/vsynth1/
 /tools/aviocat
@@ -61,6 +67,7 @@
 /tools/fourcc2pixfmt
 /tools/ffescape
 /tools/ffeval
+/tools/ffhash
 /tools/graph2dot
 /tools/ismindex
 /tools/pktdumper
@@ -68,3 +75,4 @@
 /tools/qt-faststart
 /tools/trasher
 /tools/seek_print
+/tools/zmqsend
--- a/74
+++ b/74
@@ -1,7 +1,79 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.

-version <next>:
+version 2.0:
+
+- curves filter
+- reference-counting for AVFrame and AVPacket data
+- ffmpeg now fails when input options are used for output file
+  or vice versa
+- support for Monkey's Audio versions from 3.93
+- perms and aperms filters
+- audio filtering support in ffplay
+- 10% faster aac encoding on x86 and MIPS
+- sine audio filter source
+- WebP demuxing and decoding support
+- new ffmpeg options -filter_script and -filter_complex_script, which allow a
+  filtergraph description to be read from a file
+- OpenCL support
+- audio phaser filter
+- separatefields filter
+- libquvi demuxer
+- uniform options syntax across all filters
+- telecine filter
+- new interlace filter
+- smptehdbars source
+- inverse telecine filters (fieldmatch and decimate)
+- colorbalance filter
+- colorchannelmixer filter
+- The matroska demuxer can now output proper verbatim ASS packets. It will
+  become the default at the next libavformat major bump.
+- decent native animated GIF encoding
+- asetrate filter
+- interleave filter
+- timeline editing with filters
+- vidstabdetect and vidstabtransform filters for video stabilization using
+  the vid.stab library
+- astats filter
+- trim and atrim filters
+- ffmpeg -t and -ss (output-only) options are now sample-accurate when
+  transcoding audio
+- Matroska muxer can now put the index at the beginning of the file.
+- extractplanes filter
+- avectorscope filter
+- ADPCM DTK decoder
+- ADP demuxer
+- RSD demuxer
+- RedSpark demuxer
+- ADPCM IMA Radical decoder
+- zmq filters
+- DCT denoiser filter (dctdnoiz)
+- Wavelet denoiser filter ported from libmpcodecs as owdenoise (formerly "ow")
+- Apple Intermediate Codec decoder
+- Escape 130 video decoder
+- FTP protocol support
+- V4L2 output device
+- 3D LUT filter (lut3d)
+- SMPTE 302M audio encoder
+- support for slice multithreading in libavfilter
+- Hald CLUT support (generation and filtering)
+- VC-1 interlaced B-frame support
+- support for WavPack muxing (raw and in Matroska)
+- XVideo output device
+- vignette filter
+- True Audio (TTA) encoder
+- Go2Webinar decoder
+- mcdeint filter ported from libmpcodecs
+- sab filter ported from libmpcodecs
+- ffprobe -show_chapters option
+- WavPack encoding through libwavpack
+- rotate filter
+- spp filter ported from libmpcodecs
+- libgme support
+- psnr filter
+
+
+version 1.2:

 - VDPAU hardware acceleration through normal hwaccel
 - SRTP support
--- a/15
+++ b/15
@@ -35,10 +35,14 @@ Specifically, the GPL parts of FFmpeg are
    - vf_hqdn3d.c
    - vf_hue.c
    - vf_kerndeint.c
+    - vf_mcdeint.c
    - vf_mp.c
    - vf_noise.c
+    - vf_owdenoise.c
    - vf_pp.c
+    - vf_sab.c
    - vf_smartblur.c
+    - vf_spp.c
    - vf_stereo3d.c
    - vf_super2xsai.c
    - vf_tinterlace.c
@@ -69,8 +73,15 @@ affect the licensing of binaries resulting from the combination.
 compatible libraries
 --------------------

-The libcdio, libx264, libxavs and libxvid libraries are under GPL. When
-combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
+The following libraries are under GPL:
+    - frei0r
+    - libcdio
+    - libutvideo
+    - libvidstab
+    - libx264
+    - libxavs
+    - libxvid
+When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
 passing --enable-gpl to configure.

 The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
--- a/65
+++ b/65
@@ -7,8 +7,8 @@ FFmpeg code.
 Please try to keep entries where you are the maintainer up to date!

 Names in () mean that the maintainer currently has no time to maintain the code.
-A CC after the name means that the maintainer prefers to be CC-ed on patches
-and related discussions.
+A (CC <address>) after the name means that the maintainer prefers to be CC-ed on
+patches and related discussions.


 Project Leader
@@ -46,7 +46,7 @@ Miscellaneous Areas
 documentation                           Mike Melanson
 website                                 Robert Swain, Lou Logan
 build system (configure,Makefiles)      Diego Biurrun, Mans Rullgard
-project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger
+project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger, Alexander Strasser
 mailinglists                            Michael Niedermayer, Baptiste Coudurier, Lou Logan
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
@@ -62,11 +62,20 @@ Internal Interfaces:
  libavutil/common.h                    Michael Niedermayer

 Other:
-  intfloat*                             Michael Niedermayer
-  rational.c, rational.h                Michael Niedermayer
-  mathematics.c, mathematics.h          Michael Niedermayer
-  integer.c, integer.h                  Michael Niedermayer
+  bprint                                Nicolas George
  bswap.h
+  des                                   Reimar Doeffinger
+  float_dsp                             Loren Merritt
+  hash                                  Reimar Doeffinger
+  intfloat*                             Michael Niedermayer
+  integer.c, integer.h                  Michael Niedermayer
+  lzo                                   Reimar Doeffinger
+  mathematics.c, mathematics.h          Michael Niedermayer
+  opencl.c, opencl.h                    Wei Gao
+  rational.c, rational.h                Michael Niedermayer
+  rc4                                   Reimar Doeffinger
+  ripemd.c, ripemd.h                    James Almer
+  timecode                              Clément Bœsch


 libavcodec
@@ -129,8 +138,8 @@ Codecs:
  binkaudio.c                           Peter Ross
  bmp.c                                 Mans Rullgard, Kostya Shishkov
  cavs*                                 Stefan Gehrer
-  celp_filters.*                        Vitor Sessak
  cdxl.c                                Paul B Mahol
+  celp_filters.*                        Vitor Sessak
  cinepak.c                             Roberto Togni
  cljr                                  Alex Beregszaszi
  cllc.c                                Derek Buitenhuis
@@ -141,8 +150,8 @@ Codecs:
  dca.c                                 Kostya Shishkov, Benjamin Larsson
  dnxhd*                                Baptiste Coudurier
  dpcm.c                                Mike Melanson
-  dxa.c                                 Kostya Shishkov
  dv.c                                  Roman Shaposhnik
+  dxa.c                                 Kostya Shishkov
  eacmv*, eaidct*, eat*                 Peter Ross
  ffv1.c                                Michael Niedermayer
  ffwavesynth.c                         Nicolas George
@@ -152,9 +161,9 @@ Codecs:
  g722.c                                Martin Storsjo
  g726.c                                Roman Shaposhnik
  gifdec.c                              Baptiste Coudurier
-  h264*                                 Loren Merritt, Michael Niedermayer
  h261*                                 Michael Niedermayer
  h263*                                 Michael Niedermayer
+  h264*                                 Loren Merritt, Michael Niedermayer
  huffyuv.c                             Michael Niedermayer
  idcinvideo.c                          Mike Melanson
  imc*                                  Benjamin Larsson
@@ -163,13 +172,14 @@ Codecs:
  interplayvideo.c                      Mike Melanson
  ivi*                                  Kostya Shishkov
  jacosub*                              Clément Bœsch
+  jpeg2000*                             Nicolas Bertrand
  jpeg_ls.c                             Kostya Shishkov
  jvdec.c                               Peter Ross
  kmvc.c                                Kostya Shishkov
  lcl*.c                                Roberto Togni, Reimar Doeffinger
  libcelt_dec.c                         Nicolas George
-  libgsm.c                              Michel Bardiaux
  libdirac*                             David Conrad
+  libgsm.c                              Michel Bardiaux
  libopenjpeg.c                         Jaikrishnan Menon
  libopenjpegenc.c                      Michael Bradshaw
  libschroedinger*                      David Conrad
@@ -177,8 +187,8 @@ Codecs:
  libtheoraenc.c                        David Conrad
  libutvideo*                           Derek Buitenhuis
  libvorbis.c                           David Conrad
-  libxavs.c                             Stefan Gehrer
  libx264.c                             Mans Rullgard, Jason Garrett-Glaser
+  libxavs.c                             Stefan Gehrer
  loco.c                                Kostya Shishkov
  lzo.h, lzo.c                          Reimar Doeffinger
  mdec.c                                Michael Niedermayer
@@ -189,6 +199,7 @@ Codecs:
  mpc*                                  Kostya Shishkov
  mpeg12.c, mpeg12data.h                Michael Niedermayer
  mpegvideo.c, mpegvideo.h              Michael Niedermayer
+  mqc*                                  Nicolas Bertrand
  msmpeg4.c, msmpeg4data.h              Michael Niedermayer
  msrle.c                               Mike Melanson
  msvideo1.c                            Mike Melanson
@@ -214,6 +225,7 @@ Codecs:
  s3tc*                                 Ivo van Poorten
  smacker.c                             Kostya Shishkov
  smc.c                                 Mike Melanson
+  smvjpegdec.c                          Ash Hughes
  snow.c                                Michael Niedermayer, Loren Merritt
  sonic.c                               Alex Beregszaszi
  srt*                                  Aurelien Jacobs
@@ -227,6 +239,7 @@ Codecs:
  truespeech.c                          Kostya Shishkov
  tscc.c                                Kostya Shishkov
  tta.c                                 Alex Beregszaszi, Jaikrishnan Menon
+  ttaenc.c                              Paul B Mahol
  txd.c                                 Ivo van Poorten
  ulti*                                 Kostya Shishkov
  v410*.c                               Derek Buitenhuis
@@ -237,8 +250,8 @@ Codecs:
  vda_h264_dec.c                        Xidorn Quan
  vima.c                                Paul B Mahol
  vmnc.c                                Kostya Shishkov
-  vorbis_enc.c                          Oded Shimon
  vorbis_dec.c                          Denes Balatoni, David Conrad
+  vorbis_enc.c                          Oded Shimon
  vp3*                                  Mike Melanson
  vp5                                   Aurelien Jacobs
  vp6                                   Aurelien Jacobs
@@ -272,11 +285,11 @@ libavdevice
    libavdevice/avdevice.h


+  dshow.c                               Roger Pack
  iec61883.c                            Georg Lippitsch
  libdc1394.c                           Roman Shaposhnik
  v4l2.c                                Luca Abeni
  vfwcap.c                              Ramiro Polla
-  dshow.c                               Roger Pack

 libavfilter
 ===========
@@ -286,9 +299,13 @@ Generic parts:

 Filters:
  af_amerge.c                           Nicolas George
+  af_aresample.c                        Michael Niedermayer
  af_astreamsync.c                      Nicolas George
  af_atempo.c                           Pavel Koshevoy
  af_pan.c                              Nicolas George
+  vf_delogo.c                           Jean Delvare (CC <khali@linux-fr.org>)
+  vf_drawbox.c/drawgrid                 Andrey Utkin
+  vf_scale.c                            Michael Niedermayer
  vf_yadif.c                            Michael Niedermayer

 Sources:
@@ -308,12 +325,14 @@ Muxers/Demuxers:
  4xm.c                                 Mike Melanson
  adtsenc.c                             Robert Swain
  afc.c                                 Paul B Mahol
-  aiff.c                                Baptiste Coudurier
+  aiffdec.c                             Baptiste Coudurier, Matthieu Bouron
+  aiffenc.c                             Baptiste Coudurier, Matthieu Bouron
  ape.c                                 Kostya Shishkov
  ass*                                  Aurelien Jacobs
  astdec.c                              Paul B Mahol
  astenc.c                              James Almer
  avi*                                  Michael Niedermayer
+  avisynth.c                            AvxSynth Team (avxsynth.testing at gmail dot com)
  avr.c                                 Paul B Mahol
  bink.c                                Peter Ross
  brstm.c                               Paul B Mahol
@@ -335,8 +354,8 @@ Muxers/Demuxers:
  idcin.c                               Mike Melanson
  idroqdec.c                            Mike Melanson
  iff.c                                 Jaikrishnan Menon
-  ipmovie.c                             Mike Melanson
  img2*.c                               Michael Niedermayer
+  ipmovie.c                             Mike Melanson
  ircam*                                Paul B Mahol
  iss.c                                 Stefan Gehrer
  jacosub*                              Clément Bœsch
@@ -350,11 +369,11 @@ Muxers/Demuxers:
  matroskadec.c                         Aurelien Jacobs
  matroskaenc.c                         David Conrad
  metadata*                             Aurelien Jacobs
-  microdvd*                             Aurelien Jacobs
  mgsts.c                               Paul B Mahol
+  microdvd*                             Aurelien Jacobs
  mm.c                                  Peter Ross
  mov.c                                 Michael Niedermayer, Baptiste Coudurier
-  movenc.c                              Michael Niedermayer, Baptiste Coudurier
+  movenc.c                              Baptiste Coudurier, Matthieu Bouron
  mpc.c                                 Kostya Shishkov
  mpeg.c                                Michael Niedermayer
  mpegenc.c                             Michael Niedermayer
@@ -399,6 +418,7 @@ Muxers/Demuxers:
  voc.c                                 Aurelien Jacobs
  wav.c                                 Michael Niedermayer
  wc3movie.c                            Mike Melanson
+  webvtt*                               Matthew J Heaney
  westwood.c                            Mike Melanson
  wtv.c                                 Peter Ross
  wv.c                                  Kostya Shishkov
@@ -406,6 +426,7 @@ Muxers/Demuxers:

 Protocols:
  bluray.c                              Petri Hintukainen
+  ftp.c                                 Lukasz Marek
  http.c                                Ronald S. Bultje
  mms*.c                                Ronald S. Bultje
  udp.c                                 Luca Abeni
@@ -445,10 +466,10 @@ x86                                     Michael Niedermayer
 Releases
 ========

+2.0                                     Michael Niedermayer
 1.2                                     Michael Niedermayer
-1.1                                     Michael Niedermayer
-1.0                                     Michael Niedermayer

+If you want to maintain an older release, please contact us


 GnuPG Fingerprints of maintainers and contributors
@@ -456,6 +477,7 @@ GnuPG Fingerprints of maintainers and contributors

 Anssi Hannula                 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
 Anton Khirnov                 6D0C 6625 56F8 65D1 E5F5 814B B50A 1241 C067 07AB
+Ash Hughes                    694D 43D2 D180 C7C7 6421 ABD3 A641 D0B7 623D 6029
 Attila Kinali                 11F0 F9A6 A1D2 11F6 C745 D10C 6520 BCDD F2DF E765
 Baptiste Coudurier            8D77 134D 20CC 9220 201F C5DB 0AC9 325C 5C1A BAAA
 Ben Littler                   3EE3 3723 E560 3214 A8CD 4DEB 2CDB FCE7 768C 8D2C
@@ -463,8 +485,10 @@ Benoit Fouet                  B22A 4F4F 43EF 636B BB66 FCDC 0023 AE1E 2985 49C8
 Bœsch Clément                 52D0 3A82 D445 F194 DB8B 2B16 87EE 2CB8 F4B8 FCF9
 Daniel Verkamp                78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
 Diego Biurrun                 8227 1E31 B6D9 4994 7427 E220 9CAE D6CC 4757 FCC5
+FFmpeg release signing key    FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
 Gwenole Beauchesne            2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
 Jaikrishnan Menon             61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
+Jean Delvare                  7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
 Justin Ruggles                3136 ECC0 C10D 6C04 5F43 CA29 FCBE CD2A 3787 1EBF
 Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
 Lou Logan                     7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
@@ -481,3 +505,4 @@ Sascha Sommer                 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
 Stefano Sabatini              0D0B AD6B 5330 BBAD D3D6 6A0C 719C 2839 FC43 2D5F
 Stephan Hilb                  4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
 Tomas Härdin                  A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
+Wei Gao                       4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9
--- a/18
+++ b/18
@@ -20,7 +20,7 @@ INSTPROGS   = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))

 OBJS        = cmdutils.o $(EXEOBJS)
 OBJS-ffmpeg = ffmpeg_opt.o ffmpeg_filter.o
-TESTTOOLS   = audiogen videogen rotozoom tiny_psnr base64
+TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
 HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
 TOOLS       = qt-faststart trasher
 TOOLS-$(CONFIG_ZLIB) += cws2fws
@@ -28,7 +28,6 @@ TOOLS-$(CONFIG_ZLIB) += cws2fws
 BASENAMES   = ffmpeg ffplay ffprobe ffserver
 ALLPROGS    = $(BASENAMES:%=%$(PROGSSUF)$(EXESUF))
 ALLPROGS_G  = $(BASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
-ALLMANPAGES = $(BASENAMES:%=%.1)

 FFLIBS-$(CONFIG_AVDEVICE) += avdevice
 FFLIBS-$(CONFIG_AVFILTER) += avfilter
@@ -44,7 +43,7 @@ FFLIBS := avutil
 DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
 EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile $(SRC_PATH)/doc/examples/README

-SKIPHEADERS = cmdutils_common_opts.h
+SKIPHEADERS = cmdutils_common_opts.h compat/w32pthreads.h

 include $(SRC_PATH)/common.mak

@@ -154,25 +153,16 @@ clean::
 	$(RM) $(ALLPROGS) $(ALLPROGS_G)
 	$(RM) $(CLEANSUFFIXES)
 	$(RM) $(CLEANSUFFIXES:%=tools/%)
-	$(RM) coverage.info
 	$(RM) -r coverage-html
+	$(RM) -rf coverage.info lcov

 distclean::
 	$(RM) $(DISTCLEANSUFFIXES)
-	$(RM) config.* .version version.h libavutil/avconfig.h libavcodec/codec_names.h
+	$(RM) config.* .config libavutil/avconfig.h .version version.h libavcodec/codec_names.h

 config:
 	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)

-# Without the sed genthml thinks "libavutil" and "./libavutil" are two different things
-coverage.info: $(wildcard *.gcda *.gcno */*.gcda */*.gcno */*/*.gcda */*/*.gcno)
-	$(Q)lcov -c -d . -b . | sed -e 's#/./#/#g' > $@
-
-coverage-html: coverage.info
-	$(Q)mkdir -p $@
-	$(Q)genhtml -o $@ $<
-	$(Q)touch $@
-
 check: all alltools examples testprogs fate

 include $(SRC_PATH)/doc/Makefile
--- a/2
+++ b/2
@@ -1 +1 @@
-1.1.git
+2.0.3
--- a/1
+++ b/1
@@ -0,0 +1 @@
+2.0.3
--- a/cmdutils.c
+++ b/cmdutils.c
@@ -36,9 +36,7 @@
 #include "libavresample/avresample.h"
 #include "libswscale/swscale.h"
 #include "libswresample/swresample.h"
-#if CONFIG_POSTPROC
 #include "libpostproc/postprocess.h"
-#endif
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
 #include "libavutil/bprint.h"
@@ -58,6 +56,10 @@
 #include <sys/time.h>
 #include <sys/resource.h>
 #endif
+#if CONFIG_OPENCL
+#include "libavutil/opencl.h"
+#endif
+

 static int init_report(const char *env);

@@ -109,6 +111,21 @@ static void log_callback_report(void *ptr, int level, const char *fmt, va_list v
    fflush(report_file);
 }

+static void (*program_exit)(int ret);
+
+void register_exit(void (*cb)(int ret))
+{
+    program_exit = cb;
+}
+
+void exit_program(int ret)
+{
+    if (program_exit)
+        program_exit(ret);
+
+    exit(ret);
+}
+
 double parse_number_or_die(const char *context, const char *numstr, int type,
                           double min, double max)
 {
@@ -126,7 +143,7 @@ double parse_number_or_die(const char *context, const char *numstr, int type,
    else
        return d;
    av_log(NULL, AV_LOG_FATAL, error, context, numstr, min, max);
-    exit(1);
+    exit_program(1);
    return 0;
 }

@@ -137,7 +154,7 @@ int64_t parse_time_or_die(const char *context, const char *timestr,
    if (av_parse_time(&us, timestr, is_duration) < 0) {
        av_log(NULL, AV_LOG_FATAL, "Invalid %s specification for %s: %s\n",
               is_duration ? "duration" : "date", context, timestr);
-        exit(1);
+        exit_program(1);
    }
    return us;
 }
@@ -280,7 +297,7 @@ static int write_option(void *optctx, const OptionDef *po, const char *opt,
    if (po->flags & OPT_STRING) {
        char *str;
        str = av_strdup(arg);
-//         av_freep(dst);
+        av_freep(dst);
        *(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);
@@ -296,12 +313,13 @@ static int write_option(void *optctx, const OptionDef *po, const char *opt,
        int ret = po->u.func_arg(optctx, opt, arg);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR,
-                   "Failed to set value '%s' for option '%s'\n", arg, opt);
+                   "Failed to set value '%s' for option '%s': %s\n",
+                   arg, opt, av_err2str(ret));
            return ret;
        }
    }
    if (po->flags & OPT_EXIT)
-        exit(0);
+        exit_program(0);

    return 0;
 }
@@ -361,7 +379,7 @@ void parse_options(void *optctx, int argc, char **argv, const OptionDef *options
            opt++;

            if ((ret = parse_option(optctx, opt, argv[optindex], options)) < 0)
-                exit(1);
+                exit_program(1);
            optindex += ret;
        } else {
            if (parse_arg_function)
@@ -380,6 +398,16 @@ int parse_optgroup(void *optctx, OptionGroup *g)
    for (i = 0; i < g->nb_opts; i++) {
        Option *o = &g->opts[i];

+        if (g->group_def->flags &&
+            !(g->group_def->flags & o->opt->flags)) {
+            av_log(NULL, AV_LOG_ERROR, "Option %s (%s) cannot be applied to "
+                   "%s %s -- you are trying to apply an input option to an "
+                   "output file or vice versa. Move this option before the "
+                   "file it belongs to.\n", o->key, o->opt->help,
+                   g->group_def->name, g->arg);
+            return AVERROR(EINVAL);
+        }
+
        av_log(NULL, AV_LOG_DEBUG, "Applying option %s (%s) with argument %s.\n",
               o->key, o->opt->help, o->val);

@@ -494,10 +522,10 @@ int opt_default(void *optctx, const char *opt, const char *arg)
        consumed = 1;
    }
    if ((o = av_opt_find(&fc, opt, NULL, 0,
-                              AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
+                         AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
        av_dict_set(&format_opts, opt, arg, FLAGS);
-        if(consumed)
-            av_log(NULL, AV_LOG_VERBOSE, "Routing %s to codec and muxer layer\n", opt);
+        if (consumed)
+            av_log(NULL, AV_LOG_VERBOSE, "Routing option %s to both codec and muxer layer\n", opt);
        consumed = 1;
    }
 #if CONFIG_SWSCALE
@@ -624,7 +652,7 @@ static void init_parse_context(OptionParseContext *octx,
    octx->nb_groups = nb_groups;
    octx->groups    = av_mallocz(sizeof(*octx->groups) * octx->nb_groups);
    if (!octx->groups)
-        exit(1);
+        exit_program(1);

    for (i = 0; i < octx->nb_groups; i++)
        octx->groups[i].group_def = &groups[i];
@@ -784,6 +812,13 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg)
    int level;
    int i;

+    tail = strstr(arg, "repeat");
+    av_log_set_flags(tail ? 0 : AV_LOG_SKIP_REPEATED);
+    if (tail == arg)
+        arg += 6 + (arg[6]=='+');
+    if(tail && !*arg)
+        return 0;
+
    for (i = 0; i < FF_ARRAY_ELEMS(log_levels); i++) {
        if (!strcmp(log_levels[i].name, arg)) {
            av_log_set_level(log_levels[i].level);
@@ -797,7 +832,7 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg)
               "Possible levels are numbers or:\n", arg);
        for (i = 0; i < FF_ARRAY_ELEMS(log_levels); i++)
            av_log(NULL, AV_LOG_FATAL, "\"%s\"\n", log_levels[i].name);
-        exit(1);
+        exit_program(1);
    }
    av_log_set_level(level);
    return 0;
@@ -890,7 +925,6 @@ static int init_report(const char *env)
           tm->tm_year + 1900, tm->tm_mon + 1, tm->tm_mday,
           tm->tm_hour, tm->tm_min, tm->tm_sec,
           filename.str);
-    av_log_set_level(FFMAX(av_log_get_level(), AV_LOG_VERBOSE));
    av_bprint_finalize(&filename, NULL);
    return 0;
 }
@@ -908,7 +942,7 @@ int opt_max_alloc(void *optctx, const char *opt, const char *arg)
    max = strtol(arg, &tail, 10);
    if (*tail) {
        av_log(NULL, AV_LOG_FATAL, "Invalid max_alloc \"%s\".\n", arg);
-        exit(1);
+        exit_program(1);
    }
    av_max_alloc(max);
    return 0;
@@ -939,6 +973,26 @@ int opt_timelimit(void *optctx, const char *opt, const char *arg)
    return 0;
 }

+#if CONFIG_OPENCL
+int opt_opencl(void *optctx, const char *opt, const char *arg)
+{
+    char *key, *value;
+    const char *opts = arg;
+    int ret = 0;
+    while (*opts) {
+        ret = av_opt_get_key_value(&opts, "=", ":", 0, &key, &value);
+        if (ret < 0)
+            return ret;
+        ret = av_opencl_set_option(key, value);
+        if (ret < 0)
+            return ret;
+        if (*opts)
+            opts++;
+    }
+    return ret;
+}
+#endif
+
 void print_error(const char *filename, int err)
 {
    char errbuf[128];
@@ -991,12 +1045,10 @@ static void print_all_libs_info(int flags, int level)
    PRINT_LIB_INFO(avformat, AVFORMAT, flags, level);
    PRINT_LIB_INFO(avdevice, AVDEVICE, flags, level);
    PRINT_LIB_INFO(avfilter, AVFILTER, flags, level);
-//    PRINT_LIB_INFO(avresample, AVRESAMPLE, flags, level);
+    PRINT_LIB_INFO(avresample, AVRESAMPLE, flags, level);
    PRINT_LIB_INFO(swscale,  SWSCALE,  flags, level);
    PRINT_LIB_INFO(swresample,SWRESAMPLE,  flags, level);
-#if CONFIG_POSTPROC
    PRINT_LIB_INFO(postproc, POSTPROC, flags, level);
-#endif
 }

 static void print_program_info(int flags, int level)
@@ -1176,7 +1228,8 @@ static void print_codec(const AVCodec *c)
    printf("%s %s [%s]:\n", encoder ? "Encoder" : "Decoder", c->name,
           c->long_name ? c->long_name : "");

-    if (c->type == AVMEDIA_TYPE_VIDEO) {
+    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)) {
@@ -1257,7 +1310,7 @@ static unsigned get_codecs_sorted(const AVCodecDescriptor ***rcodecs)
        nb_codecs++;
    if (!(codecs = av_calloc(nb_codecs, sizeof(*codecs)))) {
        av_log(NULL, AV_LOG_ERROR, "Out of memory\n");
-        exit(1);
+        exit_program(1);
    }
    desc = NULL;
    while ((desc = avcodec_descriptor_next(desc)))
@@ -1410,31 +1463,41 @@ int show_protocols(void *optctx, const char *opt, const char *arg)

 int show_filters(void *optctx, const char *opt, const char *arg)
 {
-    AVFilter av_unused(**filter) = NULL;
+    const AVFilter av_unused(*filter) = NULL;
    char descr[64], *descr_cur;
    int i, j;
    const AVFilterPad *pad;

-    printf("Filters:\n");
+    printf("Filters:\n"
+           "  T. = Timeline support\n"
+           "  .S = Slice threading\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 = av_filter_next(filter)) && *filter) {
+    while ((filter = avfilter_next(filter))) {
        descr_cur = descr;
        for (i = 0; i < 2; i++) {
            if (i) {
                *(descr_cur++) = '-';
                *(descr_cur++) = '>';
            }
-            pad = i ? (*filter)->outputs : (*filter)->inputs;
+            pad = i ? filter->outputs : filter->inputs;
            for (j = 0; pad && pad[j].name; j++) {
                if (descr_cur >= descr + sizeof(descr) - 4)
                    break;
                *(descr_cur++) = get_media_type_char(pad[j].type);
            }
            if (!j)
-                *(descr_cur++) = '|';
+                *(descr_cur++) = ((!i && (filter->flags & AVFILTER_FLAG_DYNAMIC_INPUTS)) ||
+                                  ( i && (filter->flags & AVFILTER_FLAG_DYNAMIC_OUTPUTS))) ? 'N' : '|';
        }
        *descr_cur = 0;
-        printf("%-16s %-10s %s\n", (*filter)->name, descr, (*filter)->description);
+        printf(" %c%c %-16s %-10s %s\n",
+               filter->flags & AVFILTER_FLAG_SUPPORT_TIMELINE ? 'T' : '.',
+               filter->flags & AVFILTER_FLAG_SLICE_THREADS    ? 'S' : '.',
+               filter->name, descr, filter->description);
    }
 #endif
    return 0;
@@ -1461,11 +1524,11 @@ int show_pix_fmts(void *optctx, const char *opt, const char *arg)
    while ((pix_desc = av_pix_fmt_desc_next(pix_desc))) {
        enum AVPixelFormat pix_fmt = av_pix_fmt_desc_get_id(pix_desc);
        printf("%c%c%c%c%c %-16s       %d            %2d\n",
-               sws_isSupportedInput (pix_fmt)      ? 'I' : '.',
-               sws_isSupportedOutput(pix_fmt)      ? 'O' : '.',
-               pix_desc->flags & PIX_FMT_HWACCEL   ? 'H' : '.',
-               pix_desc->flags & PIX_FMT_PAL       ? 'P' : '.',
-               pix_desc->flags & PIX_FMT_BITSTREAM ? 'B' : '.',
+               sws_isSupportedInput (pix_fmt)              ? 'I' : '.',
+               sws_isSupportedOutput(pix_fmt)              ? 'O' : '.',
+               pix_desc->flags & AV_PIX_FMT_FLAG_HWACCEL   ? 'H' : '.',
+               pix_desc->flags & AV_PIX_FMT_FLAG_PAL       ? 'P' : '.',
+               pix_desc->flags & AV_PIX_FMT_FLAG_BITSTREAM ? 'B' : '.',
               pix_desc->name,
               pix_desc->nb_components,
               av_get_bits_per_pixel(pix_desc));
@@ -1597,6 +1660,62 @@ static void show_help_muxer(const char *name)
        show_help_children(fmt->priv_class, AV_OPT_FLAG_ENCODING_PARAM);
 }

+#if CONFIG_AVFILTER
+static void show_help_filter(const char *name)
+{
+#if CONFIG_AVFILTER
+    const AVFilter *f = avfilter_get_by_name(name);
+    int i, count;
+
+    if (!name) {
+        av_log(NULL, AV_LOG_ERROR, "No filter name specified.\n");
+        return;
+    } else if (!f) {
+        av_log(NULL, AV_LOG_ERROR, "Unknown filter '%s'.\n", name);
+        return;
+    }
+
+    printf("Filter %s\n", f->name);
+    if (f->description)
+        printf("  %s\n", f->description);
+
+    if (f->flags & AVFILTER_FLAG_SLICE_THREADS)
+        printf("    slice threading supported\n");
+
+    printf("    Inputs:\n");
+    count = avfilter_pad_count(f->inputs);
+    for (i = 0; i < count; i++) {
+        printf("       #%d: %s (%s)\n", i, avfilter_pad_get_name(f->inputs, i),
+               media_type_string(avfilter_pad_get_type(f->inputs, i)));
+    }
+    if (f->flags & AVFILTER_FLAG_DYNAMIC_INPUTS)
+        printf("        dynamic (depending on the options)\n");
+    else if (!count)
+        printf("        none (source filter)\n");
+
+    printf("    Outputs:\n");
+    count = avfilter_pad_count(f->outputs);
+    for (i = 0; i < count; i++) {
+        printf("       #%d: %s (%s)\n", i, avfilter_pad_get_name(f->outputs, i),
+               media_type_string(avfilter_pad_get_type(f->outputs, i)));
+    }
+    if (f->flags & AVFILTER_FLAG_DYNAMIC_OUTPUTS)
+        printf("        dynamic (depending on the options)\n");
+    else if (!count)
+        printf("        none (sink filter)\n");
+
+    if (f->priv_class)
+        show_help_children(f->priv_class, AV_OPT_FLAG_VIDEO_PARAM | AV_OPT_FLAG_FILTERING_PARAM |
+                                          AV_OPT_FLAG_AUDIO_PARAM);
+    if (f->flags & AVFILTER_FLAG_SUPPORT_TIMELINE)
+        printf("This filter has support for timeline through the 'enable' option.\n");
+#else
+    av_log(NULL, AV_LOG_ERROR, "Build without libavfilter; "
+           "can not to satisfy request\n");
+#endif
+}
+#endif
+
 int show_help(void *optctx, const char *opt, const char *arg)
 {
    char *topic, *par;
@@ -1617,6 +1736,10 @@ int show_help(void *optctx, const char *opt, const char *arg)
        show_help_demuxer(par);
    } else if (!strcmp(topic, "muxer")) {
        show_help_muxer(par);
+#if CONFIG_AVFILTER
+    } else if (!strcmp(topic, "filter")) {
+        show_help_filter(par);
+#endif
    } else {
        show_help_default(topic, par);
    }
@@ -1816,13 +1939,13 @@ void *grow_array(void *array, int elem_size, int *size, int new_size)
 {
    if (new_size >= INT_MAX / elem_size) {
        av_log(NULL, AV_LOG_ERROR, "Array too big.\n");
-        exit(1);
+        exit_program(1);
    }
    if (*size < new_size) {
        uint8_t *tmp = av_realloc(array, new_size*elem_size);
        if (!tmp) {
            av_log(NULL, AV_LOG_ERROR, "Could not alloc buffer.\n");
-            exit(1);
+            exit_program(1);
        }
        memset(tmp + *size*elem_size, 0, (new_size-*size) * elem_size);
        *size = new_size;
@@ -1830,144 +1953,3 @@ void *grow_array(void *array, int elem_size, int *size, int new_size)
    }
    return array;
 }
-
-static int alloc_buffer(FrameBuffer **pool, AVCodecContext *s, FrameBuffer **pbuf)
-{
-    const AVPixFmtDescriptor *desc = av_pix_fmt_desc_get(s->pix_fmt);
-    FrameBuffer *buf;
-    int i, ret;
-    int pixel_size;
-    int h_chroma_shift, v_chroma_shift;
-    int edge = 32; // XXX should be avcodec_get_edge_width(), but that fails on svq1
-    int w = s->width, h = s->height;
-
-    if (!desc)
-        return AVERROR(EINVAL);
-    pixel_size = desc->comp[0].step_minus1 + 1;
-
-    buf = av_mallocz(sizeof(*buf));
-    if (!buf)
-        return AVERROR(ENOMEM);
-
-    avcodec_align_dimensions(s, &w, &h);
-
-    if (!(s->flags & CODEC_FLAG_EMU_EDGE)) {
-        w += 2*edge;
-        h += 2*edge;
-    }
-
-    if ((ret = av_image_alloc(buf->base, buf->linesize, w, h,
-                              s->pix_fmt, 32)) < 0) {
-        av_freep(&buf);
-        av_log(s, AV_LOG_ERROR, "alloc_buffer: av_image_alloc() failed\n");
-        return ret;
-    }
-
-    avcodec_get_chroma_sub_sample(s->pix_fmt, &h_chroma_shift, &v_chroma_shift);
-    for (i = 0; i < FF_ARRAY_ELEMS(buf->data); i++) {
-        const int h_shift = i==0 ? 0 : h_chroma_shift;
-        const int v_shift = i==0 ? 0 : v_chroma_shift;
-        if ((s->flags & CODEC_FLAG_EMU_EDGE) || !buf->linesize[i] || !buf->base[i])
-            buf->data[i] = buf->base[i];
-        else
-            buf->data[i] = buf->base[i] +
-                           FFALIGN((buf->linesize[i]*edge >> v_shift) +
-                                   (pixel_size*edge >> h_shift), 32);
-    }
-    buf->w       = s->width;
-    buf->h       = s->height;
-    buf->pix_fmt = s->pix_fmt;
-    buf->pool    = pool;
-
-    *pbuf = buf;
-    return 0;
-}
-
-int codec_get_buffer(AVCodecContext *s, AVFrame *frame)
-{
-    FrameBuffer **pool = s->opaque;
-    FrameBuffer *buf;
-    int ret, i;
-
-    if(av_image_check_size(s->width, s->height, 0, s) || s->pix_fmt<0) {
-        av_log(s, AV_LOG_ERROR, "codec_get_buffer: image parameters invalid\n");
-        return -1;
-    }
-
-    if (!*pool && (ret = alloc_buffer(pool, s, pool)) < 0)
-        return ret;
-
-    buf              = *pool;
-    *pool            = buf->next;
-    buf->next        = NULL;
-    if (buf->w != s->width || buf->h != s->height || buf->pix_fmt != s->pix_fmt) {
-        av_freep(&buf->base[0]);
-        av_free(buf);
-        if ((ret = alloc_buffer(pool, s, &buf)) < 0)
-            return ret;
-    }
-    av_assert0(!buf->refcount);
-    buf->refcount++;
-
-    frame->opaque        = buf;
-    frame->type          = FF_BUFFER_TYPE_USER;
-    frame->extended_data = frame->data;
-
-    for (i = 0; i < FF_ARRAY_ELEMS(buf->data); i++) {
-        frame->base[i]     = buf->base[i];  // XXX h264.c uses base though it shouldn't
-        frame->data[i]     = buf->data[i];
-        frame->linesize[i] = buf->linesize[i];
-    }
-
-    return 0;
-}
-
-static void unref_buffer(FrameBuffer *buf)
-{
-    FrameBuffer **pool = buf->pool;
-
-    av_assert0(buf->refcount > 0);
-    buf->refcount--;
-    if (!buf->refcount) {
-        FrameBuffer *tmp;
-        for(tmp= *pool; tmp; tmp= tmp->next)
-            av_assert1(tmp != buf);
-
-        buf->next = *pool;
-        *pool = buf;
-    }
-}
-
-void codec_release_buffer(AVCodecContext *s, AVFrame *frame)
-{
-    FrameBuffer *buf = frame->opaque;
-    int i;
-
-    if(frame->type!=FF_BUFFER_TYPE_USER) {
-        avcodec_default_release_buffer(s, frame);
-        return;
-    }
-
-    for (i = 0; i < FF_ARRAY_ELEMS(frame->data); i++)
-        frame->data[i] = NULL;
-
-    unref_buffer(buf);
-}
-
-void filter_release_buffer(AVFilterBuffer *fb)
-{
-    FrameBuffer *buf = fb->priv;
-    av_free(fb);
-    unref_buffer(buf);
-}
-
-void free_buffer_pool(FrameBuffer **pool)
-{
-    FrameBuffer *buf = *pool;
-    while (buf) {
-        *pool = buf->next;
-        av_freep(&buf->base[0]);
-        av_free(buf);
-        buf = *pool;
-    }
-}
--- a/cmdutils.h
+++ b/cmdutils.h
@@ -54,6 +54,16 @@ extern struct SwsContext *sws_opts;
 extern AVDictionary *swr_opts;
 extern AVDictionary *format_opts, *codec_opts, *resample_opts;

+/**
+ * Register a program-specific cleanup routine.
+ */
+void register_exit(void (*cb)(int ret));
+
+/**
+ * Wraps exit with a program-specific cleanup routine.
+ */
+void exit_program(int ret);
+
 /**
 * Initialize the cmdutils option system, in particular
 * allocate the *_opts contexts.
@@ -90,6 +100,8 @@ int opt_cpuflags(void *optctx, const char *opt, const char *arg);

 int opt_codec_debug(void *optctx, const char *opt, const char *arg);

+int opt_opencl(void *optctx, const char *opt, const char *arg);
+
 /**
 * Limit the execution time.
 */
@@ -162,6 +174,8 @@ typedef struct OptionDef {
                                   an int containing element count in the array. */
 #define OPT_TIME  0x10000
 #define OPT_DOUBLE 0x20000
+#define OPT_INPUT  0x40000
+#define OPT_OUTPUT 0x80000
     union {
        void *dst_ptr;
        int (*func_arg)(void *, const char *, const char *);
@@ -190,13 +204,13 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags,
 void show_help_children(const AVClass *class, int flags);

 /**
- * Per-avtool specific help handler. Implemented in each
- * avtool, called by show_help().
+ * Per-fftool specific help handler. Implemented in each
+ * fftool, called by show_help().
 */
 void show_help_default(const char *opt, const char *arg);

 /**
- * Generic -h handler common to all avtools.
+ * Generic -h handler common to all fftools.
 */
 int show_help(void *optctx, const char *opt, const char *arg);

@@ -242,6 +256,11 @@ typedef struct OptionGroupDef {
     * are terminated by a non-option argument (e.g. ffmpeg output files)
     */
    const char *sep;
+    /**
+     * Option flags that must be set on each option that is
+     * applied to this group
+     */
+    int flags;
 } OptionGroupDef;

 typedef struct OptionGroup {
@@ -517,52 +536,11 @@ FILE *get_preset_file(char *filename, size_t filename_size,
 */
 void *grow_array(void *array, int elem_size, int *size, int new_size);

+#define media_type_string av_get_media_type_string
+
 #define GROW_ARRAY(array, nb_elems)\
    array = grow_array(array, sizeof(*array), &nb_elems, nb_elems + 1)

-typedef struct FrameBuffer {
-    uint8_t *base[4];
-    uint8_t *data[4];
-    int  linesize[4];
-
-    int h, w;
-    enum AVPixelFormat pix_fmt;
-
-    int refcount;
-    struct FrameBuffer **pool;  ///< head of the buffer pool
-    struct FrameBuffer *next;
-} FrameBuffer;
-
-/**
- * Get a frame from the pool. This is intended to be used as a callback for
- * AVCodecContext.get_buffer.
- *
- * @param s codec context. s->opaque must be a pointer to the head of the
- *          buffer pool.
- * @param frame frame->opaque will be set to point to the FrameBuffer
- *              containing the frame data.
- */
-int codec_get_buffer(AVCodecContext *s, AVFrame *frame);
-
-/**
- * A callback to be used for AVCodecContext.release_buffer along with
- * codec_get_buffer().
- */
-void codec_release_buffer(AVCodecContext *s, AVFrame *frame);
-
-/**
- * A callback to be used for AVFilterBuffer.free.
- * @param fb buffer to free. fb->priv must be a pointer to the FrameBuffer
- *           containing the buffer data.
- */
-void filter_release_buffer(AVFilterBuffer *fb);
-
-/**
- * Free all the buffers in the pool. This must be called after all the
- * buffers have been released.
- */
-void free_buffer_pool(FrameBuffer **pool);
-
 #define GET_PIX_FMT_NAME(pix_fmt)\
    const char *name = av_get_pix_fmt_name(pix_fmt);

--- a/cmdutils_common_opts.h
+++ b/cmdutils_common_opts.h
@@ -14,8 +14,11 @@
    { "pix_fmts"   , OPT_EXIT, {.func_arg = show_pix_fmts },    "show available pixel formats" },
    { "layouts"    , OPT_EXIT, {.func_arg = show_layouts  },    "show standard channel layouts" },
    { "sample_fmts", OPT_EXIT, {.func_arg = show_sample_fmts }, "show available audio sample formats" },
-    { "loglevel"   , HAS_ARG,  {.func_arg = opt_loglevel},      "set libav* logging level", "loglevel" },
-    { "v",           HAS_ARG,  {.func_arg = opt_loglevel},      "set libav* logging level", "loglevel" },
+    { "loglevel"   , HAS_ARG,  {.func_arg = opt_loglevel},      "set logging level", "loglevel" },
+    { "v",           HAS_ARG,  {.func_arg = opt_loglevel},      "set logging level", "loglevel" },
    { "report"     , 0,        {(void*)opt_report}, "generate a report" },
    { "max_alloc"  , HAS_ARG,  {.func_arg = opt_max_alloc},     "set maximum size of a single allocated block", "bytes" },
    { "cpuflags"   , HAS_ARG | OPT_EXPERT, {.func_arg = opt_cpuflags}, "force specific cpu flags", "flags" },
+#if CONFIG_OPENCL
+    { "opencl_options", HAS_ARG, {.func_arg = opt_opencl},      "set OpenCL environment options" },
+#endif
--- a/common.mak
+++ b/common.mak
@@ -101,6 +101,7 @@ HEADERS   += $(HEADERS-yes)

 DEP_LIBS := $(foreach NAME,$(FFLIBS),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))

+SRC_DIR    := $(SRC_PATH)/lib$(NAME)
 ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
 SKIPHEADERS += $(ARCH_HEADERS:%=$(ARCH)/%) $(SKIPHEADERS-)
 SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
--- a/compat/aix/math.h
+++ b/compat/aix/math.h
@@ -0,0 +1,14 @@
+/*
+ * Workaround aix-specific class() function clashing with ffmpeg class usage
+ */
+
+#ifndef COMPAT_AIX_MATH_H
+#define COMPAT_AIX_MATH_H
+
+#define class class_in_math_h_causes_problems
+
+#include_next <math.h>
+
+#undef class
+
+#endif /* COMPAT_AIX_MATH_H */
--- a/compat/avisynth/avisynth_c.h
+++ b/compat/avisynth/avisynth_c.h
@@ -0,0 +1,879 @@
+// 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.
+
+// NOTE: this is a partial update of the Avisynth C interface to recognize
+// new color spaces added in Avisynth 2.60. By no means is this document
+// completely Avisynth 2.60 compliant.
+
+#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
+
+
+/////////////////////////////////////////////////////////////////////
+//
+// Constants
+//
+
+#ifndef __AVISYNTH_H__
+enum { AVISYNTH_INTERFACE_VERSION = 4 };
+#endif
+
+enum {AVS_SAMPLE_INT8  = 1<<0,
+      AVS_SAMPLE_INT16 = 1<<1,
+      AVS_SAMPLE_INT24 = 1<<2,
+      AVS_SAMPLE_INT32 = 1<<3,
+      AVS_SAMPLE_FLOAT = 1<<4};
+
+enum {AVS_PLANAR_Y=1<<0,
+      AVS_PLANAR_U=1<<1,
+      AVS_PLANAR_V=1<<2,
+      AVS_PLANAR_ALIGNED=1<<3,
+      AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
+      AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
+      AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED,
+      AVS_PLANAR_A=1<<4,
+      AVS_PLANAR_R=1<<5,
+      AVS_PLANAR_G=1<<6,
+      AVS_PLANAR_B=1<<7,
+      AVS_PLANAR_A_ALIGNED=AVS_PLANAR_A|AVS_PLANAR_ALIGNED,
+      AVS_PLANAR_R_ALIGNED=AVS_PLANAR_R|AVS_PLANAR_ALIGNED,
+      AVS_PLANAR_G_ALIGNED=AVS_PLANAR_G|AVS_PLANAR_ALIGNED,
+      AVS_PLANAR_B_ALIGNED=AVS_PLANAR_B|AVS_PLANAR_ALIGNED};
+
+  // Colorspace properties.
+enum {AVS_CS_BGR = 1<<28,
+      AVS_CS_YUV = 1<<29,
+      AVS_CS_INTERLEAVED = 1<<30,
+      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_SUB_WIDTH_MASK    = 7 << AVS_CS_SHIFT_SUB_WIDTH,
+      AVS_CS_SUB_WIDTH_1       = 3 << AVS_CS_SHIFT_SUB_WIDTH, // YV24
+      AVS_CS_SUB_WIDTH_2       = 0 << AVS_CS_SHIFT_SUB_WIDTH, // YV12, I420, YV16
+      AVS_CS_SUB_WIDTH_4       = 1 << AVS_CS_SHIFT_SUB_WIDTH, // YUV9, YV411
+
+      AVS_CS_VPLANEFIRST       = 1 << 3, // YV12, YV16, YV24, YV411, YUV9
+      AVS_CS_UPLANEFIRST       = 1 << 4, // I420
+
+      AVS_CS_SUB_HEIGHT_MASK   = 7 << AVS_CS_SHIFT_SUB_HEIGHT,
+      AVS_CS_SUB_HEIGHT_1      = 3 << AVS_CS_SHIFT_SUB_HEIGHT, // YV16, YV24, YV411
+      AVS_CS_SUB_HEIGHT_2      = 0 << AVS_CS_SHIFT_SUB_HEIGHT, // YV12, I420
+      AVS_CS_SUB_HEIGHT_4      = 1 << AVS_CS_SHIFT_SUB_HEIGHT, // YUV9
+
+      AVS_CS_SAMPLE_BITS_MASK  = 7 << AVS_CS_SHIFT_SAMPLE_BITS,
+      AVS_CS_SAMPLE_BITS_8     = 0 << AVS_CS_SHIFT_SAMPLE_BITS,
+      AVS_CS_SAMPLE_BITS_16    = 1 << AVS_CS_SHIFT_SAMPLE_BITS,
+      AVS_CS_SAMPLE_BITS_32    = 2 << AVS_CS_SHIFT_SAMPLE_BITS,
+
+      AVS_CS_PLANAR_MASK       = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_BGR | AVS_CS_SAMPLE_BITS_MASK | AVS_CS_SUB_HEIGHT_MASK | AVS_CS_SUB_WIDTH_MASK,
+      AVS_CS_PLANAR_FILTER     = ~( AVS_CS_VPLANEFIRST | AVS_CS_UPLANEFIRST )};
+
+  // Specific colorformats
+enum {
+  AVS_CS_UNKNOWN = 0,
+  AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
+  AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
+  AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
+  //  AVS_CS_YV12  = 1<<3  Reserved
+  //  AVS_CS_I420  = 1<<4  Reserved
+  AVS_CS_RAW32 = 1<<5 | AVS_CS_INTERLEAVED,
+
+  AVS_CS_YV24  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_1,  // YVU 4:4:4 planar
+  AVS_CS_YV16  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_2,  // YVU 4:2:2 planar
+  AVS_CS_YV12  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2,  // YVU 4:2:0 planar
+  AVS_CS_I420  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_UPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2,  // YUV 4:2:0 planar
+  AVS_CS_IYUV  = AVS_CS_I420,
+  AVS_CS_YV411 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_4,  // YVU 4:1:1 planar
+  AVS_CS_YUV9  = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_4 | AVS_CS_SUB_WIDTH_4,  // YVU 4:1:0 planar
+  AVS_CS_Y8    = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8                                              // Y   4:0:0 planar
+};
+
+enum {
+  AVS_IT_BFF = 1<<0,
+  AVS_IT_TFF = 1<<1,
+  AVS_IT_FIELDBASED = 1<<2};
+
+enum {
+  AVS_FILTER_TYPE=1,
+  AVS_FILTER_INPUT_COLORSPACE=2,
+  AVS_FILTER_OUTPUT_TYPE=9,
+  AVS_FILTER_NAME=4,
+  AVS_FILTER_AUTHOR=5,
+  AVS_FILTER_VERSION=6,
+  AVS_FILTER_ARGS=7,
+  AVS_FILTER_ARGS_INFO=8,
+  AVS_FILTER_ARGS_DESCRIPTION=10,
+  AVS_FILTER_DESCRIPTION=11};
+
+enum {  //SUBTYPES
+  AVS_FILTER_TYPE_AUDIO=1,
+  AVS_FILTER_TYPE_VIDEO=2,
+  AVS_FILTER_OUTPUT_TYPE_SAME=3,
+  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
+  };
+
+#define AVS_FRAME_ALIGN 16
+
+typedef struct AVS_Clip AVS_Clip;
+typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_VideoInfo
+//
+
+// AVS_VideoInfo is layed out identicly to VideoInfo
+typedef struct AVS_VideoInfo {
+  int width, height;    // width=0 means no video
+  unsigned fps_numerator, fps_denominator;
+  int num_frames;
+
+  int pixel_type;
+
+  int audio_samples_per_second;   // 0 means no audio
+  int sample_type;
+  INT64 num_audio_samples;
+  int nchannels;
+
+  // Imagetype properties
+
+  int image_type;
+} AVS_VideoInfo;
+
+// useful functions of the above
+AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
+        { return (p->width!=0); }
+
+AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
+        { return (p->audio_samples_per_second!=0); }
+
+AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
+        { return !!(p->pixel_type&AVS_CS_BGR); }
+
+AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
+        { return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
+
+AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
+        { return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
+
+AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
+        { return !!(p->pixel_type&AVS_CS_YUV ); }
+
+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_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_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_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_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_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
+        { return ((p->pixel_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_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
+        { return !!(p->image_type & AVS_IT_FIELDBASED); }
+
+AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
+        { return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
+
+AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
+        { return !!(p->image_type & AVS_IT_BFF); }
+
+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_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_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_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
+        { return p->audio_samples_per_second; }
+
+
+AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
+{
+    switch (p->sample_type) {
+      case AVS_SAMPLE_INT8:  return sizeof(signed char);
+      case AVS_SAMPLE_INT16: return sizeof(signed short);
+      case AVS_SAMPLE_INT24: return 3;
+      case AVS_SAMPLE_INT32: return sizeof(signed int);
+      case AVS_SAMPLE_FLOAT: return sizeof(float);
+      default: return 0;
+    }
+}
+AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
+        { return p->nchannels*avs_bytes_per_channel_sample(p);}
+
+AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
+        { return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
+
+AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
+        { return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
+
+AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
+        { return bytes / avs_bytes_per_audio_sample(p); }
+
+AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
+        { return samples * avs_bytes_per_audio_sample(p); }
+
+AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
+        { return p->nchannels; }
+
+AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
+        { return p->sample_type;}
+
+// useful mutator
+AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
+        { p->image_type|=property; }
+
+AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
+        { p->image_type&=~property; }
+
+AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
+        { if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
+
+AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
+{
+    unsigned x=numerator, y=denominator;
+    while (y) {   // find gcd
+      unsigned t = x%y; x = y; y = t;
+    }
+    p->fps_numerator = numerator/x;
+    p->fps_denominator = denominator/x;
+}
+
+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));
+}
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_VideoFrame
+//
+
+// VideoFrameBuffer holds information about a memory block which is used
+// for video data.  For efficiency, instances of this class are not deleted
+// when the refcount reaches zero; instead they're stored in a linked list
+// to be reused.  The instances are deleted when the corresponding AVS
+// file is closed.
+
+// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
+// DO NOT USE THIS STRUCTURE DIRECTLY
+typedef struct AVS_VideoFrameBuffer {
+  BYTE * data;
+  int data_size;
+  // sequence_number is incremented every time the buffer is changed, so
+  // that stale views can tell they're no longer valid.
+  volatile long sequence_number;
+
+  volatile long refcount;
+} AVS_VideoFrameBuffer;
+
+// VideoFrame holds a "window" into a VideoFrameBuffer.
+
+// AVS_VideoFrame is layed out identicly to IVideoFrame
+// DO NOT USE THIS STRUCTURE DIRECTLY
+typedef struct AVS_VideoFrame {
+  volatile long refcount;
+  AVS_VideoFrameBuffer * vfb;
+  int offset, pitch, row_size, height, offsetU, offsetV, pitchUV;  // U&V offsets are from top of picture.
+  int row_sizeUV, heightUV;
+} AVS_VideoFrame;
+
+// Access functions for AVS_VideoFrame
+AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
+        return p->pitch;}
+
+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;}
+
+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_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_INLINE const BYTE* avs_get_read_ptr(const AVS_VideoFrame * p) {
+        return p->vfb->data + p->offset;}
+
+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_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(void, avs_release_video_frame)(AVS_VideoFrame *);
+// makes a shallow copy of a video frame
+AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
+
+#ifndef AVSC_NO_DECLSPEC
+AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
+  {avs_release_video_frame(f);}
+AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
+  {return avs_copy_video_frame(f);}
+#endif
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_Value
+//
+
+// Treat AVS_Value as a fat pointer.  That is use avs_copy_value
+// and avs_release_value appropiaty as you would if AVS_Value was
+// a pointer.
+
+// To maintain source code compatibility with future versions of the
+// avisynth_c API don't use the AVS_Value directly.  Use the helper
+// functions below.
+
+// AVS_Value is layed out identicly to AVSValue
+typedef struct AVS_Value AVS_Value;
+struct AVS_Value {
+  short type;  // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
+               // for some function e'rror
+  short array_size;
+  union {
+    void * clip; // do not use directly, use avs_take_clip
+    char boolean;
+    int integer;
+    float floating_pt;
+    const char * string;
+    const AVS_Value * array;
+  } d;
+};
+
+// AVS_Value should be initilized with avs_void.
+// Should also set to avs_void after the value is released
+// with avs_copy_value.  Consider it the equalvent of setting
+// a pointer to NULL
+static const AVS_Value avs_void = {'v'};
+
+AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
+AVSC_API(void, avs_release_value)(AVS_Value);
+
+AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
+AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
+AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
+AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
+AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
+AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
+AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
+AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
+
+AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
+AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
+
+AVSC_INLINE int avs_as_bool(AVS_Value v)
+        { return v.d.boolean; }
+AVSC_INLINE int avs_as_int(AVS_Value v)
+        { return v.d.integer; }
+AVSC_INLINE const char * avs_as_string(AVS_Value v)
+        { return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
+AVSC_INLINE double avs_as_float(AVS_Value v)
+        { return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
+AVSC_INLINE const char * avs_as_error(AVS_Value v)
+        { return avs_is_error(v) ? v.d.string : 0; }
+AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
+        { return v.d.array; }
+AVSC_INLINE int avs_array_size(AVS_Value v)
+        { return avs_is_array(v) ? v.array_size : 1; }
+AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
+        { return avs_is_array(v) ? v.d.array[index] : v; }
+
+// only use these functions on an 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; }
+AVSC_INLINE AVS_Value avs_new_value_int(int v0)
+        { AVS_Value v; 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; }
+AVSC_INLINE AVS_Value avs_new_value_float(float v0)
+        { AVS_Value v; 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; }
+#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; }
+#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_Clip
+//
+
+AVSC_API(void, avs_release_clip)(AVS_Clip *);
+AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
+
+AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
+
+AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
+
+AVSC_API(int, avs_get_version)(AVS_Clip *);
+
+AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
+// The returned video frame must be released with avs_release_video_frame
+
+AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
+// return field parity if field_based, else parity of first field in frame
+
+AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
+                             INT64 start, INT64 count);
+// start and count are in samples
+
+AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
+                                   int cachehints, int frame_range);
+
+// This is the callback type used by avs_add_function
+typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
+                        (AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
+
+typedef struct AVS_FilterInfo AVS_FilterInfo;
+struct AVS_FilterInfo
+{
+  // these members should not be modified outside of the AVS_ApplyFunc callback
+  AVS_Clip * child;
+  AVS_VideoInfo vi;
+  AVS_ScriptEnvironment * env;
+  AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
+  int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
+  int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
+                                  INT64 start, INT64 count);
+  int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
+                                        int frame_range);
+  void (AVSC_CC * free_filter)(AVS_FilterInfo *);
+
+  // Should be set when ever there is an error to report.
+  // It is cleared before any of the above methods are called
+  const char * error;
+  // this is to store whatever and may be modified at will
+  void * user_data;
+};
+
+// Create a new filter
+// fi is set to point to the AVS_FilterInfo so that you can
+//   modify it once it is initilized.
+// store_child should generally be set to true.  If it is not
+//    set than ALL methods (the function pointers) must be defined
+// If it is set than you do not need to worry about freeing the child
+//    clip.
+AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
+                                       AVS_FilterInfo * * fi,
+                                       AVS_Value child, int store_child);
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_ScriptEnvironment
+//
+
+// For GetCPUFlags.  These are backwards-compatible with those in VirtualDub.
+enum {
+                                /* slowest CPU to support extension */
+  AVS_CPU_FORCE        = 0x01,   // N/A
+  AVS_CPU_FPU          = 0x02,   // 386/486DX
+  AVS_CPU_MMX          = 0x04,   // P55C, K6, PII
+  AVS_CPU_INTEGER_SSE  = 0x08,   // PIII, Athlon
+  AVS_CPU_SSE          = 0x10,   // PIII, Athlon XP/MP
+  AVS_CPU_SSE2         = 0x20,   // PIV, Hammer
+  AVS_CPU_3DNOW        = 0x40,   // K6-2
+  AVS_CPU_3DNOW_EXT    = 0x80,   // Athlon
+  AVS_CPU_X86_64       = 0xA0,   // Hammer (note: equiv. to 3DNow + SSE2,
+                                 // which only Hammer will have anyway)
+  AVS_CPUF_SSE3       = 0x100,   //  PIV+, K8 Venice
+  AVS_CPUF_SSSE3      = 0x200,   //  Core 2
+  AVS_CPUF_SSE4       = 0x400,   //  Penryn, Wolfdale, Yorkfield
+  AVS_CPUF_SSE4_1     = 0x400,
+  AVS_CPUF_SSE4_2     = 0x800,   //  Nehalem
+};
+
+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_check_version)(AVS_ScriptEnvironment *, int version);
+
+AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
+AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
+
+AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, void* val);
+ // note: val is really a va_list; I hope everyone typedefs va_list to a pointer
+
+AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
+                                const char * name, const char * params,
+                                AVS_ApplyFunc apply, void * user_data);
+
+AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
+
+AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
+                               AVS_Value args, const char** arg_names);
+// The returned value must be be released with avs_release_value
+
+AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
+// The returned value must be be released with avs_release_value
+
+AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
+
+AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
+
+//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
+//void avs_pop_context(AVS_ScriptEnvironment *);
+
+AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
+                                          const AVS_VideoInfo * vi, int align);
+// align should be at least 16
+
+#ifndef AVSC_NO_DECLSPEC
+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);}
+
+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);}
+#endif
+
+
+AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
+
+AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, BYTE* dstp, int dst_pitch, const BYTE* srcp, int src_pitch, int row_size, int height);
+
+typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
+AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
+
+AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
+// The returned video frame must be be released
+
+AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
+
+AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
+
+// avisynth.dll exports this; it's a way to use it as a library, without
+// writing an AVS script or without going through AVIFile.
+AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
+
+// this symbol is the entry point for the plugin and must
+// be defined
+AVSC_EXPORT
+const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
+
+
+AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
+
+
+AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
+// The returned video frame must be be released
+
+#ifdef AVSC_NO_DECLSPEC
+// use LoadLibrary and related functions to dynamically load Avisynth instead of declspec(dllimport)
+/*
+  The following functions needs to have been declared, probably from windows.h
+
+  void* malloc(size_t)
+  void free(void*);
+
+  HMODULE LoadLibrary(const char*);
+  void* GetProcAddress(HMODULE, const char*);
+  FreeLibrary(HMODULE);
+*/
+
+
+typedef struct AVS_Library AVS_Library;
+
+#define AVSC_DECLARE_FUNC(name) name##_func name
+
+struct AVS_Library {
+  HMODULE handle;
+
+  AVSC_DECLARE_FUNC(avs_add_function);
+  AVSC_DECLARE_FUNC(avs_at_exit);
+  AVSC_DECLARE_FUNC(avs_bit_blt);
+  AVSC_DECLARE_FUNC(avs_check_version);
+  AVSC_DECLARE_FUNC(avs_clip_get_error);
+  AVSC_DECLARE_FUNC(avs_copy_clip);
+  AVSC_DECLARE_FUNC(avs_copy_value);
+  AVSC_DECLARE_FUNC(avs_copy_video_frame);
+  AVSC_DECLARE_FUNC(avs_create_script_environment);
+  AVSC_DECLARE_FUNC(avs_delete_script_environment);
+  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);
+  AVSC_DECLARE_FUNC(avs_get_version);
+  AVSC_DECLARE_FUNC(avs_get_video_info);
+  AVSC_DECLARE_FUNC(avs_invoke);
+  AVSC_DECLARE_FUNC(avs_make_writable);
+  AVSC_DECLARE_FUNC(avs_new_c_filter);
+  AVSC_DECLARE_FUNC(avs_new_video_frame_a);
+  AVSC_DECLARE_FUNC(avs_release_clip);
+  AVSC_DECLARE_FUNC(avs_release_value);
+  AVSC_DECLARE_FUNC(avs_release_video_frame);
+  AVSC_DECLARE_FUNC(avs_save_string);
+  AVSC_DECLARE_FUNC(avs_set_cache_hints);
+  AVSC_DECLARE_FUNC(avs_set_global_var);
+  AVSC_DECLARE_FUNC(avs_set_memory_max);
+  AVSC_DECLARE_FUNC(avs_set_to_clip);
+  AVSC_DECLARE_FUNC(avs_set_var);
+  AVSC_DECLARE_FUNC(avs_set_working_dir);
+  AVSC_DECLARE_FUNC(avs_sprintf);
+  AVSC_DECLARE_FUNC(avs_subframe);
+  AVSC_DECLARE_FUNC(avs_subframe_planar);
+  AVSC_DECLARE_FUNC(avs_take_clip);
+  AVSC_DECLARE_FUNC(avs_vsprintf);
+};
+
+#undef AVSC_DECLARE_FUNC
+
+
+AVSC_INLINE AVS_Library * avs_load_library() {
+  AVS_Library *library = (AVS_Library *)malloc(sizeof(AVS_Library));
+  if (library == NULL)
+    return NULL;
+  library->handle = LoadLibrary("avisynth");
+  if (library->handle == NULL)
+    goto fail;
+
+#define __AVSC_STRINGIFY(x) #x
+#define AVSC_STRINGIFY(x) __AVSC_STRINGIFY(x)
+#define AVSC_LOAD_FUNC(name) {\
+  library->name = (name##_func) GetProcAddress(library->handle, AVSC_STRINGIFY(name));\
+  if (library->name == NULL)\
+    goto fail;\
+}
+
+  AVSC_LOAD_FUNC(avs_add_function);
+  AVSC_LOAD_FUNC(avs_at_exit);
+  AVSC_LOAD_FUNC(avs_bit_blt);
+  AVSC_LOAD_FUNC(avs_check_version);
+  AVSC_LOAD_FUNC(avs_clip_get_error);
+  AVSC_LOAD_FUNC(avs_copy_clip);
+  AVSC_LOAD_FUNC(avs_copy_value);
+  AVSC_LOAD_FUNC(avs_copy_video_frame);
+  AVSC_LOAD_FUNC(avs_create_script_environment);
+  AVSC_LOAD_FUNC(avs_delete_script_environment);
+  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);
+  AVSC_LOAD_FUNC(avs_get_version);
+  AVSC_LOAD_FUNC(avs_get_video_info);
+  AVSC_LOAD_FUNC(avs_invoke);
+  AVSC_LOAD_FUNC(avs_make_writable);
+  AVSC_LOAD_FUNC(avs_new_c_filter);
+  AVSC_LOAD_FUNC(avs_new_video_frame_a);
+  AVSC_LOAD_FUNC(avs_release_clip);
+  AVSC_LOAD_FUNC(avs_release_value);
+  AVSC_LOAD_FUNC(avs_release_video_frame);
+  AVSC_LOAD_FUNC(avs_save_string);
+  AVSC_LOAD_FUNC(avs_set_cache_hints);
+  AVSC_LOAD_FUNC(avs_set_global_var);
+  AVSC_LOAD_FUNC(avs_set_memory_max);
+  AVSC_LOAD_FUNC(avs_set_to_clip);
+  AVSC_LOAD_FUNC(avs_set_var);
+  AVSC_LOAD_FUNC(avs_set_working_dir);
+  AVSC_LOAD_FUNC(avs_sprintf);
+  AVSC_LOAD_FUNC(avs_subframe);
+  AVSC_LOAD_FUNC(avs_subframe_planar);
+  AVSC_LOAD_FUNC(avs_take_clip);
+  AVSC_LOAD_FUNC(avs_vsprintf);
+
+#undef __AVSC_STRINGIFY
+#undef AVSC_STRINGIFY
+#undef AVSC_LOAD_FUNC
+
+  return library;
+
+fail:
+  free(library);
+  return NULL;
+}
+
+AVSC_INLINE void avs_free_library(AVS_Library *library) {
+  if (library == NULL)
+    return;
+  FreeLibrary(library->handle);
+  free(library);
+}
+#endif
+
+#endif
--- a/compat/avisynth/avisynth_c_25.h
+++ b/compat/avisynth/avisynth_c_25.h
@@ -0,0 +1,68 @@
+//  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); }
--- a/compat/avisynth/avxsynth_c.h
+++ b/compat/avisynth/avxsynth_c.h
@@ -0,0 +1,727 @@
+// 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 __AVXSYNTH_C__
+#define __AVXSYNTH_C__
+
+#include "windowsPorts/windows2linux.h"
+#include <stdarg.h>
+
+#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
+
+#ifdef __GNUC__
+typedef long long int INT64;
+#else
+typedef __int64 INT64;
+#endif
+
+
+/////////////////////////////////////////////////////////////////////
+//
+// Constants
+//
+
+#ifndef __AVXSYNTH_H__
+enum { AVISYNTH_INTERFACE_VERSION = 3 };
+#endif
+
+enum {AVS_SAMPLE_INT8  = 1<<0,
+      AVS_SAMPLE_INT16 = 1<<1,
+      AVS_SAMPLE_INT24 = 1<<2,
+      AVS_SAMPLE_INT32 = 1<<3,
+      AVS_SAMPLE_FLOAT = 1<<4};
+
+enum {AVS_PLANAR_Y=1<<0,
+      AVS_PLANAR_U=1<<1,
+      AVS_PLANAR_V=1<<2,
+      AVS_PLANAR_ALIGNED=1<<3,
+      AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED,
+      AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED,
+      AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED};
+
+  // Colorspace properties.
+enum {AVS_CS_BGR = 1<<28,
+      AVS_CS_YUV = 1<<29,
+      AVS_CS_INTERLEAVED = 1<<30,
+      AVS_CS_PLANAR = 1<<31};
+
+  // Specific colorformats
+enum {
+  AVS_CS_UNKNOWN = 0,
+  AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
+  AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED,
+  AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED,
+  AVS_CS_YV12 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-v-u, planar
+  AVS_CS_I420 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-u-v, planar
+  AVS_CS_IYUV = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR  // same as above
+};
+
+enum {
+  AVS_IT_BFF = 1<<0,
+  AVS_IT_TFF = 1<<1,
+  AVS_IT_FIELDBASED = 1<<2};
+
+enum {
+  AVS_FILTER_TYPE=1,
+  AVS_FILTER_INPUT_COLORSPACE=2,
+  AVS_FILTER_OUTPUT_TYPE=9,
+  AVS_FILTER_NAME=4,
+  AVS_FILTER_AUTHOR=5,
+  AVS_FILTER_VERSION=6,
+  AVS_FILTER_ARGS=7,
+  AVS_FILTER_ARGS_INFO=8,
+  AVS_FILTER_ARGS_DESCRIPTION=10,
+  AVS_FILTER_DESCRIPTION=11};
+
+enum {  //SUBTYPES
+  AVS_FILTER_TYPE_AUDIO=1,
+  AVS_FILTER_TYPE_VIDEO=2,
+  AVS_FILTER_OUTPUT_TYPE_SAME=3,
+  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
+};
+
+#define AVS_FRAME_ALIGN 16
+
+typedef struct AVS_Clip AVS_Clip;
+typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_VideoInfo
+//
+
+// AVS_VideoInfo is layed out identicly to VideoInfo
+typedef struct AVS_VideoInfo {
+  int width, height;    // width=0 means no video
+  unsigned fps_numerator, fps_denominator;
+  int num_frames;
+
+  int pixel_type;
+
+  int audio_samples_per_second;   // 0 means no audio
+  int sample_type;
+  INT64 num_audio_samples;
+  int nchannels;
+
+  // Imagetype properties
+
+  int image_type;
+} AVS_VideoInfo;
+
+// useful functions of the above
+AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p)
+        { return (p->width!=0); }
+
+AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p)
+        { return (p->audio_samples_per_second!=0); }
+
+AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p)
+        { return !!(p->pixel_type&AVS_CS_BGR); }
+
+AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p)
+        { return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties
+
+AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p)
+        { return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; }
+
+AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
+        { return !!(p->pixel_type&AVS_CS_YUV ); }
+
+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_yv12(const AVS_VideoInfo * p)
+        { return ((p->pixel_type & AVS_CS_YV12) == AVS_CS_YV12)||((p->pixel_type & AVS_CS_I420) == AVS_CS_I420); }
+
+AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
+        { return ((p->pixel_type & c_space) == c_space); }
+
+AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
+        { return ((p->pixel_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_field_based(const AVS_VideoInfo * p)
+        { return !!(p->image_type & AVS_IT_FIELDBASED); }
+
+AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p)
+        { return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); }
+
+AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
+        { return !!(p->image_type & AVS_IT_BFF); }
+
+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_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_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_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
+        { return p->audio_samples_per_second; }
+
+
+AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p)
+{
+    switch (p->sample_type) {
+      case AVS_SAMPLE_INT8:  return sizeof(signed char);
+      case AVS_SAMPLE_INT16: return sizeof(signed short);
+      case AVS_SAMPLE_INT24: return 3;
+      case AVS_SAMPLE_INT32: return sizeof(signed int);
+      case AVS_SAMPLE_FLOAT: return sizeof(float);
+      default: return 0;
+    }
+}
+AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p)
+        { return p->nchannels*avs_bytes_per_channel_sample(p);}
+
+AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames)
+        { return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); }
+
+AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
+        { return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); }
+
+AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes)
+        { return bytes / avs_bytes_per_audio_sample(p); }
+
+AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples)
+        { return samples * avs_bytes_per_audio_sample(p); }
+
+AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p)
+        { return p->nchannels; }
+
+AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p)
+        { return p->sample_type;}
+
+// useful mutator
+AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property)
+        { p->image_type|=property; }
+
+AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property)
+        { p->image_type&=~property; }
+
+AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased)
+        { if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; }
+
+AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator)
+{
+    unsigned x=numerator, y=denominator;
+    while (y) {   // find gcd
+      unsigned t = x%y; x = y; y = t;
+    }
+    p->fps_numerator = numerator/x;
+    p->fps_denominator = denominator/x;
+}
+
+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));
+}
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_VideoFrame
+//
+
+// VideoFrameBuffer holds information about a memory block which is used
+// for video data.  For efficiency, instances of this class are not deleted
+// when the refcount reaches zero; instead they're stored in a linked list
+// to be reused.  The instances are deleted when the corresponding AVS
+// file is closed.
+
+// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer
+// DO NOT USE THIS STRUCTURE DIRECTLY
+typedef struct AVS_VideoFrameBuffer {
+  unsigned char * data;
+  int data_size;
+  // sequence_number is incremented every time the buffer is changed, so
+  // that stale views can tell they're no longer valid.
+  long sequence_number;
+
+  long refcount;
+} AVS_VideoFrameBuffer;
+
+// VideoFrame holds a "window" into a VideoFrameBuffer.
+
+// AVS_VideoFrame is layed out identicly to IVideoFrame
+// DO NOT USE THIS STRUCTURE DIRECTLY
+typedef struct AVS_VideoFrame {
+  int refcount;
+  AVS_VideoFrameBuffer * vfb;
+  int offset, pitch, row_size, height, offsetU, offsetV, pitchUV;  // U&V offsets are from top of picture.
+} AVS_VideoFrame;
+
+// Access functions for AVS_VideoFrame
+AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
+        return p->pitch;}
+
+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;}
+
+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_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_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->height>>1;
+                        return 0;
+        }
+        return p->height;}
+
+AVSC_INLINE const unsigned char* avs_get_read_ptr(const AVS_VideoFrame * p) {
+        return p->vfb->data + p->offset;}
+
+AVSC_INLINE const unsigned char* 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_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
+        return (p->refcount == 1 && p->vfb->refcount == 1);}
+
+AVSC_INLINE unsigned char* 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 unsigned char* 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;
+                }
+        }
+}
+
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
+// makes a shallow copy of a video frame
+AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *);
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+#ifndef AVSC_NO_DECLSPEC
+AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f)
+  {avs_release_video_frame(f);}
+AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f)
+  {return avs_copy_video_frame(f);}
+#endif
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_Value
+//
+
+// Treat AVS_Value as a fat pointer.  That is use avs_copy_value
+// and avs_release_value appropiaty as you would if AVS_Value was
+// a pointer.
+
+// To maintain source code compatibility with future versions of the
+// avisynth_c API don't use the AVS_Value directly.  Use the helper
+// functions below.
+
+// AVS_Value is layed out identicly to AVSValue
+typedef struct AVS_Value AVS_Value;
+struct AVS_Value {
+  short type;  // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong
+               // for some function e'rror
+  short array_size;
+  union {
+    void * clip; // do not use directly, use avs_take_clip
+    char boolean;
+    int integer;
+    INT64 integer64; // match addition of __int64 to avxplugin.h
+    float floating_pt;
+    const char * string;
+    const AVS_Value * array;
+  } d;
+};
+
+// AVS_Value should be initilized with avs_void.
+// Should also set to avs_void after the value is released
+// with avs_copy_value.  Consider it the equalvent of setting
+// a pointer to NULL
+static const AVS_Value avs_void = {'v'};
+
+AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src);
+AVSC_API(void, avs_release_value)(AVS_Value);
+
+AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; }
+AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; }
+AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; }
+AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; }
+AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; }
+AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; }
+AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; }
+AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; }
+
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *);
+AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *);
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+AVSC_INLINE int avs_as_bool(AVS_Value v)
+        { return v.d.boolean; }
+AVSC_INLINE int avs_as_int(AVS_Value v)
+        { return v.d.integer; }
+AVSC_INLINE const char * avs_as_string(AVS_Value v)
+        { return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; }
+AVSC_INLINE double avs_as_float(AVS_Value v)
+        { return avs_is_int(v) ? v.d.integer : v.d.floating_pt; }
+AVSC_INLINE const char * avs_as_error(AVS_Value v)
+        { return avs_is_error(v) ? v.d.string : 0; }
+AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v)
+        { return v.d.array; }
+AVSC_INLINE int avs_array_size(AVS_Value v)
+        { return avs_is_array(v) ? v.array_size : 1; }
+AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
+        { return avs_is_array(v) ? v.d.array[index] : v; }
+
+// 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; }
+AVSC_INLINE AVS_Value avs_new_value_int(int v0)
+        { AVS_Value v; 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; }
+AVSC_INLINE AVS_Value avs_new_value_float(float v0)
+        { AVS_Value v; 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; }
+#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; }
+#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_Clip
+//
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(void, avs_release_clip)(AVS_Clip *);
+AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *);
+
+AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error
+
+AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *);
+
+AVSC_API(int, avs_get_version)(AVS_Clip *);
+
+AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n);
+// The returned video frame must be released with avs_release_video_frame
+
+AVSC_API(int, avs_get_parity)(AVS_Clip *, int n);
+// return field parity if field_based, else parity of first field in frame
+
+AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf,
+                                  INT64 start, INT64 count);
+// start and count are in samples
+
+AVSC_API(int, avs_set_cache_hints)(AVS_Clip *,
+                                        int cachehints, size_t frame_range);
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+// This is the callback type used by avs_add_function
+typedef AVS_Value (AVSC_CC * AVS_ApplyFunc)
+                        (AVS_ScriptEnvironment *, AVS_Value args, void * user_data);
+
+typedef struct AVS_FilterInfo AVS_FilterInfo;
+struct AVS_FilterInfo
+{
+  // these members should not be modified outside of the AVS_ApplyFunc callback
+  AVS_Clip * child;
+  AVS_VideoInfo vi;
+  AVS_ScriptEnvironment * env;
+  AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n);
+  int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n);
+  int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf,
+                                  INT64 start, INT64 count);
+  int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints,
+                                        int frame_range);
+  void (AVSC_CC * free_filter)(AVS_FilterInfo *);
+
+  // Should be set when ever there is an error to report.
+  // It is cleared before any of the above methods are called
+  const char * error;
+  // this is to store whatever and may be modified at will
+  void * user_data;
+};
+
+// Create a new filter
+// fi is set to point to the AVS_FilterInfo so that you can
+//   modify it once it is initilized.
+// store_child should generally be set to true.  If it is not
+//    set than ALL methods (the function pointers) must be defined
+// If it is set than you do not need to worry about freeing the child
+//    clip.
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e,
+                                      AVS_FilterInfo * * fi,
+                                      AVS_Value child, int store_child);
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+
+/////////////////////////////////////////////////////////////////////
+//
+// AVS_ScriptEnvironment
+//
+
+// For GetCPUFlags.  These are backwards-compatible with those in VirtualDub.
+enum {
+                                /* slowest CPU to support extension */
+  AVS_CPU_FORCE        = 0x01,   // N/A
+  AVS_CPU_FPU          = 0x02,   // 386/486DX
+  AVS_CPU_MMX          = 0x04,   // P55C, K6, PII
+  AVS_CPU_INTEGER_SSE  = 0x08,   // PIII, Athlon
+  AVS_CPU_SSE          = 0x10,   // PIII, Athlon XP/MP
+  AVS_CPU_SSE2         = 0x20,   // PIV, Hammer
+  AVS_CPU_3DNOW        = 0x40,   // K6-2
+  AVS_CPU_3DNOW_EXT    = 0x80,   // Athlon
+  AVS_CPU_X86_64       = 0xA0,   // Hammer (note: equiv. to 3DNow + SSE2,
+                                 // which only Hammer will have anyway)
+};
+
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+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_check_version)(AVS_ScriptEnvironment *, int version);
+
+AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
+AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...);
+
+AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, va_list val);
+ // note: val is really a va_list; I hope everyone typedefs va_list to a pointer
+
+AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *,
+                                     const char * name, const char * params,
+                                     AVS_ApplyFunc apply, void * user_data);
+
+AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name);
+
+AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name,
+                               AVS_Value args, const char** arg_names);
+// The returned value must be be released with avs_release_value
+
+AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name);
+// The returned value must be be released with avs_release_value
+
+AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val);
+
+AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val);
+
+//void avs_push_context(AVS_ScriptEnvironment *, int level=0);
+//void avs_pop_context(AVS_ScriptEnvironment *);
+
+AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
+                                          const AVS_VideoInfo * vi, int align);
+// align should be at least 16
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+#ifndef AVSC_NO_DECLSPEC
+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);}
+
+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);}
+#endif
+
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf);
+
+AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, unsigned char* dstp, int dst_pitch, const unsigned char* srcp, int src_pitch, int row_size, int height);
+
+typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env);
+AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data);
+
+AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height);
+// The returned video frame must be be released
+
+AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem);
+
+AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir);
+
+// avisynth.dll exports this; it's a way to use it as a library, without
+// writing an AVS script or without going through AVIFile.
+AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version);
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+// this symbol is the entry point for the plugin and must
+// be defined
+AVSC_EXPORT
+const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env);
+
+
+#if defined __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *);
+
+
+AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV);
+// The returned video frame must be be released
+#if defined __cplusplus
+}
+#endif // __cplusplus
+
+#endif //__AVXSYNTH_C__
--- a/compat/avisynth/windowsPorts/basicDataTypeConversions.h
+++ b/compat/avisynth/windowsPorts/basicDataTypeConversions.h
@@ -0,0 +1,85 @@
+#ifndef __DATA_TYPE_CONVERSIONS_H__
+#define __DATA_TYPE_CONVERSIONS_H__
+
+#include <stdint.h>
+#include <wchar.h>
+
+#ifdef __cplusplus
+namespace avxsynth {
+#endif // __cplusplus
+
+typedef int64_t __int64;
+typedef int32_t __int32;
+#ifdef __cplusplus
+typedef bool BOOL;
+#else
+typedef uint32_t BOOL;
+#endif // __cplusplus
+typedef void* HMODULE;
+typedef void* LPVOID;
+typedef void* PVOID;
+typedef PVOID HANDLE;
+typedef HANDLE HWND;
+typedef HANDLE HINSTANCE;
+typedef void* HDC;
+typedef void* HBITMAP;
+typedef void* HICON;
+typedef void* HFONT;
+typedef void* HGDIOBJ;
+typedef void* HBRUSH;
+typedef void* HMMIO;
+typedef void* HACMSTREAM;
+typedef void* HACMDRIVER;
+typedef void* HIC;
+typedef void* HACMOBJ;
+typedef HACMSTREAM* LPHACMSTREAM;
+typedef void* HACMDRIVERID;
+typedef void* LPHACMDRIVER;
+typedef unsigned char BYTE;
+typedef BYTE* LPBYTE;
+typedef char TCHAR;
+typedef TCHAR* LPTSTR;
+typedef const TCHAR* LPCTSTR;
+typedef char* LPSTR;
+typedef LPSTR LPOLESTR;
+typedef const char* LPCSTR;
+typedef LPCSTR LPCOLESTR;
+typedef wchar_t WCHAR;
+typedef unsigned short WORD;
+typedef unsigned int UINT;
+typedef UINT MMRESULT;
+typedef uint32_t DWORD;
+typedef DWORD COLORREF;
+typedef DWORD FOURCC;
+typedef DWORD HRESULT;
+typedef DWORD* LPDWORD;
+typedef DWORD* DWORD_PTR;
+typedef int32_t LONG;
+typedef int32_t* LONG_PTR;
+typedef LONG_PTR LRESULT;
+typedef uint32_t ULONG;
+typedef uint32_t* ULONG_PTR;
+//typedef __int64_t intptr_t;
+typedef uint64_t _fsize_t;
+
+
+//
+// Structures
+//
+
+typedef struct _GUID {
+  DWORD Data1;
+  WORD  Data2;
+  WORD  Data3;
+  BYTE  Data4[8];
+} GUID;
+
+typedef GUID REFIID;
+typedef GUID CLSID;
+typedef CLSID* LPCLSID;
+typedef GUID IID;
+
+#ifdef __cplusplus
+}; // namespace avxsynth
+#endif // __cplusplus
+#endif //  __DATA_TYPE_CONVERSIONS_H__
--- a/compat/avisynth/windowsPorts/windows2linux.h
+++ b/compat/avisynth/windowsPorts/windows2linux.h
@@ -0,0 +1,77 @@
+#ifndef __WINDOWS2LINUX_H__
+#define __WINDOWS2LINUX_H__
+
+/*
+ * LINUX SPECIFIC DEFINITIONS
+*/
+//
+// Data types conversions
+//
+#include <stdlib.h>
+#include <string.h>
+#include "basicDataTypeConversions.h"
+
+#ifdef __cplusplus
+namespace avxsynth {
+#endif // __cplusplus
+//
+// purposefully define the following MSFT definitions
+// to mean nothing (as they do not mean anything on Linux)
+//
+#define __stdcall
+#define __cdecl
+#define noreturn
+#define __declspec(x)
+#define STDAPI       extern "C" HRESULT
+#define STDMETHODIMP HRESULT __stdcall
+#define STDMETHODIMP_(x) x __stdcall
+
+#define STDMETHOD(x)    virtual HRESULT x
+#define STDMETHOD_(a, x) virtual a x
+
+#ifndef TRUE
+#define TRUE  true
+#endif
+
+#ifndef FALSE
+#define FALSE false
+#endif
+
+#define S_OK                (0x00000000)
+#define S_FALSE             (0x00000001)
+#define E_NOINTERFACE       (0X80004002)
+#define E_POINTER           (0x80004003)
+#define E_FAIL              (0x80004005)
+#define E_OUTOFMEMORY       (0x8007000E)
+
+#define INVALID_HANDLE_VALUE    ((HANDLE)((LONG_PTR)-1))
+#define FAILED(hr)              ((hr) & 0x80000000)
+#define SUCCEEDED(hr)           (!FAILED(hr))
+
+
+//
+// Functions
+//
+#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
+#define lstrcmpi                            strcasecmp
+#define _stricmp                            strcasecmp
+#define InterlockedIncrement(x)             __sync_fetch_and_add((x), 1)
+#define InterlockedDecrement(x)             __sync_fetch_and_sub((x), 1)
+// Windows uses (new, old) ordering but GCC has (old, new)
+#define InterlockedCompareExchange(x,y,z)   __sync_val_compare_and_swap(x,z,y)
+
+#define UInt32x32To64(a, b)                 ( (uint64_t) ( ((uint64_t)((uint32_t)(a))) * ((uint32_t)(b))  ) )
+#define Int64ShrlMod32(a, b)                ( (uint64_t) ( (uint64_t)(a) >> (b) ) )
+#define Int32x32To64(a, b)                  ((__int64)(((__int64)((long)(a))) * ((long)(b))))
+
+#define MulDiv(nNumber, nNumerator, nDenominator)   (int32_t) (((int64_t) (nNumber) * (int64_t) (nNumerator) + (int64_t) ((nDenominator)/2)) / (int64_t) (nDenominator))
+
+#ifdef __cplusplus
+}; // namespace avxsynth
+#endif // __cplusplus
+
+#endif //  __WINDOWS2LINUX_H__
--- a/libavcodec/os2threads.h
+++ b/libavcodec/os2threads.h
--- a/libavcodec/w32pthreads.h
+++ b/libavcodec/w32pthreads.h
@@ -4,20 +4,20 @@
 * Authors: Steven Walters <kemuri9@gmail.com>
 *          Pegasys Inc. <http://www.pegasys-inc.com>
 *
- * This file is part of Libav.
+ * This file is part of FFmpeg.
 *
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

@@ -26,8 +26,8 @@
 * w32threads to pthreads wrapper
 */

-#ifndef AVCODEC_W32PTHREADS_H
-#define AVCODEC_W32PTHREADS_H
+#ifndef LIBAV_W32PTHREADS_H
+#define LIBAV_W32PTHREADS_H

 /* Build up a pthread-like API using underlying Windows API. Have only static
 * methods so as to not conflict with a potentially linked in pthread-win32
@@ -257,6 +257,7 @@ static void pthread_cond_signal(pthread_cond_t *cond)

 static void w32thread_init(void)
 {
+#if _WIN32_WINNT < 0x0600
    HANDLE kernel_dll = GetModuleHandle(TEXT("kernel32.dll"));
    /* if one is available, then they should all be available */
    cond_init      =
@@ -267,6 +268,13 @@ static void w32thread_init(void)
        (void*)GetProcAddress(kernel_dll, "WakeConditionVariable");
    cond_wait      =
        (void*)GetProcAddress(kernel_dll, "SleepConditionVariableCS");
+#else
+    cond_init      = InitializeConditionVariable;
+    cond_broadcast = WakeAllConditionVariable;
+    cond_signal    = WakeConditionVariable;
+    cond_wait      = SleepConditionVariableCS;
+#endif
+
 }

-#endif /* AVCODEC_W32PTHREADS_H */
+#endif /* LIBAV_W32PTHREADS_H */
--- a/717
+++ b/717
--- a/doc/APIchanges
+++ b/doc/APIchanges
@@ -2,10 +2,10 @@ 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:    2012-01-27
-libavdevice:   2011-04-18
+libavcodec:    2013-03-xx
+libavdevice:   2013-03-xx
 libavfilter:   2012-06-22
-libavformat:   2012-01-27
+libavformat:   2013-03-xx
 libavresample: 2012-10-05
 libpostproc:   2011-04-18
 libswresample: 2011-09-19
@@ -15,6 +15,78 @@ libavutil:     2012-10-22

 API changes, most recent first:

+2013-07-03 - xxxxxxx - lavfi 3.78.100 - avfilter.h
+  Deprecate avfilter_graph_parse() in favor of the equivalent
+  avfilter_graph_parse_ptr().
+
+2013-06-xx - xxxxxxx - lavc 55.10.0 - avcodec.h
+  Add MPEG-2 AAC profiles
+
+2013-06-xx - xxxxxxx - lavf 55.10.100 - avformat.h
+  Add AV_DISPOSITION_* flags to indicate text track kind.
+
+2013-06-xx - xxxxxxx - lavu 52.36.100
+  Add AVRIPEMD:
+   av_ripemd_alloc()
+   av_ripemd_init()
+   av_ripemd_update()
+   av_ripemd_final()
+
+2013-06-05 - fc962d4 - lavu 52.13.0 - mem.h
+  Add av_realloc_array and av_reallocp_array
+
+2013-05-30 - 682b227 - lavu 52.35.100
+  Add AVSHA512:
+   av_sha512_alloc()
+   av_sha512_init()
+   av_sha512_update()
+   av_sha512_final()
+
+2013-05-24 - xxxxxxx - lavfi 3.70.100 - avfilter.h
+  Add support for slice multithreading to lavfi. Filters supporting threading
+  are marked with AVFILTER_FLAG_SLICE_THREADS.
+  New fields AVFilterContext.thread_type, AVFilterGraph.thread_type and
+  AVFilterGraph.nb_threads (accessible directly or through AVOptions) may be
+  used to configure multithreading.
+
+2013-05-24 - xxxxxxx - lavu 52.34.100 - cpu.h
+  Add av_cpu_count() function for getting the number of logical CPUs.
+
+2013-05-24 - xxxxxxx - lavc 55.12.100 - avcodec.h
+  Add picture_structure to AVCodecParserContext.
+
+2013-05-17 - xxxxxxx - lavu 52.33.100 - opt.h
+  Add AV_OPT_TYPE_COLOR value to AVOptionType enum.
+
+2013-05-13 - xxxxxxx - lavu 52.31.100 - mem.h
+  Add av_dynarray2_add().
+
+2013-05-12 - xxxxxxx - lavfi 3.65.100
+  Add AVFILTER_FLAG_SUPPORT_TIMELINE* filter flags.
+
+2013-04-19 - xxxxxxx - lavc 55.4.100
+  Add AV_CODEC_PROP_TEXT_SUB property for text based subtitles codec.
+
+2013-04-18 - xxxxxxx - lavf 55.3.100
+  The matroska demuxer can now output proper verbatim ASS packets. It will
+  become the default starting lavf 56.0.100.
+
+2013-04-10 - xxxxxxx - lavu 25.26.100 - avutil.h,opt.h
+  Add av_int_list_length()
+  and av_opt_set_int_list().
+
+2013-03-30 - xxxxxxx - lavu 52.24.100 - samplefmt.h
+  Add av_samples_alloc_array_and_samples().
+
+2013-03-29 - xxxxxxx - lavf 55.1.100 - avformat.h
+  Add av_guess_frame_rate()
+
+2013-03-20 - xxxxxxx - lavu 52.22.100 - opt.h
+  Add AV_OPT_TYPE_DURATION value to AVOptionType enum.
+
+2013-03-17 - xxxxxx - lavu 52.20.100 - opt.h
+  Add AV_OPT_TYPE_VIDEO_RATE value to AVOptionType enum.
+
 2013-03-07 - xxxxxx - lavu 52.18.100 - avstring.h,bprint.h
  Add av_escape() and av_bprint_escape() API.

@@ -144,21 +216,93 @@ API changes, most recent first:
 2012-03-26 - a67d9cf - lavfi 2.66.100
  Add avfilter_fill_frame_from_{audio_,}buffer_ref() functions.

-2013-xx-xx - xxxxxxx - lavfi 3.4.0 - avfiltergraph.h
+2013-05-xx - xxxxxxx - lavu 52.11.0 - pixdesc.h
+  Replace PIX_FMT_* flags with AV_PIX_FMT_FLAG_*.
+
+2013-04-xx - xxxxxxx - lavc 55.4.0 - avcodec.h
+  Add field_order to AVCodecParserContext.
+
+2013-03-xx - xxxxxxx - lavc 55.2.0 - avcodec.h
+  Add CODEC_FLAG_UNALIGNED to allow decoders to produce unaligned output.
+
+2013-04-11 - lavfi 3.8.0
+  38f0c07 - Move all content from avfiltergraph.h to avfilter.h. Deprecate
+            avfilterhraph.h, user applications should include just avfilter.h
+  bc1a985 - Add avfilter_graph_alloc_filter(), deprecate avfilter_open() and
+            avfilter_graph_add_filter().
+  1113672 - Add AVFilterContext.graph pointing to the AVFilterGraph that contains the
+            filter.
+  48a5ada - Add avfilter_init_str(), deprecate avfilter_init_filter().
+  1ba95a9 - Add avfilter_init_dict().
+  7cdd737 - Add AVFilter.flags field and AVFILTER_FLAG_DYNAMIC_{INPUTS,OUTPUTS} flags.
+  7e8fe4b - Add avfilter_pad_count() for counting filter inputs/outputs.
+  fa2a34c - Add avfilter_next(), deprecate av_filter_next().
+            Deprecate avfilter_uninit().
+
+2013-04-09 - lavfi 3.7.0 - avfilter.h
+  b439c99 - Add AVFilter.priv_class for exporting filter options through the
+            AVOptions API in the similar way private options work in lavc and lavf.
+  8114c10 - Add avfilter_get_class().
+  Switch all filters to use AVOptions.
+
+2013-03-19 - 2c328a9 - lavu 52.9.0 - pixdesc.h
+  Add av_pix_fmt_count_planes() function for counting planes in a pixel format.
+
+2013-03-16 - 42c7c61 - lavfi 3.6.0
+  Add AVFilterGraph.nb_filters, deprecate AVFilterGraph.filter_count.
+
+2013-03-08 - Reference counted buffers - lavu 52.8.0, lavc 55.0.0, lavf 55.0.0,
+lavd 54.0.0, lavfi 3.5.0
+  8e401db, 1cec062 - add a new API for reference counted buffers and buffer
+                     pools (new header libavutil/buffer.h).
+  1afddbe - add AVPacket.buf to allow reference counting for the AVPacket data.
+            Add av_packet_from_data() function for constructing packets from
+            av_malloc()ed data.
+  7ecc2d4 - move AVFrame from lavc to lavu (new header libavutil/frame.h), add
+            AVFrame.buf/extended_buf to allow reference counting for the AVFrame
+            data. Add new API for working with reference-counted AVFrames.
+  759001c - add the refcounted_frames field to AVCodecContext to make audio and
+            video decoders return reference-counted frames. Add get_buffer2()
+            callback to AVCodecContext which allocates reference-counted frames.
+            Add avcodec_default_get_buffer2() as the default get_buffer2()
+            implementation.
+            Deprecate AVCodecContext.get_buffer() / release_buffer() /
+            reget_buffer(), avcodec_default_get_buffer(),
+            avcodec_default_reget_buffer(), avcodec_default_release_buffer().
+            Remove avcodec_default_free_buffers(), which should not have ever
+            been called from outside of lavc.
+            Deprecate the following AVFrame fields:
+                * base -- is now stored in AVBufferRef
+                * reference, type, buffer_hints -- are unnecessary in the new API
+                * hwaccel_picture_private, owner, thread_opaque -- should not
+                  have been acessed from outside of lavc
+                * qscale_table, qstride, qscale_type, mbskip_table, motion_val,
+                  mb_type, dct_coeff, ref_index -- mpegvideo-specific tables,
+                  which are not exported anymore.
+  7e35037 - switch libavfilter to use AVFrame instead of AVFilterBufferRef. Add
+            av_buffersrc_add_frame(), deprecate av_buffersrc_buffer().
+            Add av_buffersink_get_frame() and av_buffersink_get_samples(),
+            deprecate av_buffersink_read() and av_buffersink_read_samples().
+            Deprecate AVFilterBufferRef and all functions for working with it.
+
+2013-03-17 - 12c5c1d - lavu 52.8.0 - avstring.h
+  Add av_isdigit, av_isgraph, av_isspace, av_isxdigit.
+
+2013-02-23 - 9f12235 - lavfi 3.4.0 - avfiltergraph.h
  Add resample_lavr_opts to AVFilterGraph for setting libavresample options
  for auto-inserted resample filters.

-2013-xx-xx - xxxxxxx - lavu 52.7.0 - dict.h
+2013-01-25 - 38c1466 - lavu 52.7.0 - dict.h
  Add av_dict_parse_string() to set multiple key/value pairs at once from a
  string.

-2013-01-xx - xxxxxxx - lavu 52.6.0 - avstring.h
+2013-01-25 - b85a5e8 - lavu 52.6.0 - avstring.h
  Add av_strnstr()

-2013-01-xx - xxxxxxx - lavu 52.5.0 - hmac.h
+2013-01-15 - 8ee288d - lavu 52.5.0 - hmac.h
  Add AVHMAC.

-2013-01-13 - xxxxxxx - lavc 54.87.100 / 54.36.0 - vdpau.h
+2013-01-13 - 44e065d - lavc 54.87.100 / 54.36.0 - vdpau.h
  Add AVVDPAUContext struct for VDPAU hardware-accelerated decoding.

 2013-01-12 - dae382b / 169fb94 - lavu 52.14.100 / 52.4.0 - pixdesc.h
--- a/doc/Doxyfile
+++ b/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         = 2.0.3

 # With the PROJECT_LOGO tag one can specify an logo or icon that is included
 # in the documentation. The maximum height of the logo should not exceed 55
@@ -277,7 +277,7 @@ SUBGROUPING            = YES
 # be useful for C code in case the coding convention dictates that all compound
 # types are typedef'ed and only the typedef is referenced, never the tag name.

-TYPEDEF_HIDES_STRUCT   = NO
+TYPEDEF_HIDES_STRUCT   = YES

 # The SYMBOL_CACHE_SIZE determines the size of the internal cache use to
 # determine which symbols to keep in memory and which to flush to disk.
@@ -409,7 +409,7 @@ INLINE_INFO            = YES
 # alphabetically by member name. If set to NO the members will appear in
 # declaration order.

-SORT_MEMBER_DOCS       = YES
+SORT_MEMBER_DOCS       = NO

 # If the SORT_BRIEF_DOCS tag is set to YES then doxygen will sort the
 # brief documentation of file, namespace and class members alphabetically
@@ -709,7 +709,7 @@ INLINE_SOURCES         = NO
 # doxygen to hide any special comment blocks from generated source code
 # fragments. Normal C and C++ comments will always remain visible.

-STRIP_CODE_COMMENTS    = YES
+STRIP_CODE_COMMENTS    = NO

 # If the REFERENCED_BY_RELATION tag is set to YES
 # then for each documented function all documented
--- a/doc/Makefile
+++ b/doc/Makefile
@@ -6,7 +6,6 @@ LIBRARIES-$(CONFIG_AVFORMAT)   += libavformat
 LIBRARIES-$(CONFIG_AVDEVICE)   += libavdevice
 LIBRARIES-$(CONFIG_AVFILTER)   += libavfilter

-COMPONENTS-yes = $(PROGS-yes)
 COMPONENTS-$(CONFIG_AVUTIL)     += ffmpeg-utils
 COMPONENTS-$(CONFIG_SWSCALE)    += ffmpeg-scaler
 COMPONENTS-$(CONFIG_SWRESAMPLE) += ffmpeg-resampler
@@ -15,9 +14,11 @@ COMPONENTS-$(CONFIG_AVFORMAT)   += ffmpeg-formats ffmpeg-protocols
 COMPONENTS-$(CONFIG_AVDEVICE)   += ffmpeg-devices
 COMPONENTS-$(CONFIG_AVFILTER)   += ffmpeg-filters

-MANPAGES    = $(COMPONENTS-yes:%=doc/%.1)    $(LIBRARIES-yes:%=doc/%.3)
-PODPAGES    = $(COMPONENTS-yes:%=doc/%.pod)  $(LIBRARIES-yes:%=doc/%.pod)
-HTMLPAGES   = $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html)     \
+MANPAGES1   = $(PROGS-yes:%=doc/%.1)    $(PROGS-yes:%=doc/%-all.1)    $(COMPONENTS-yes:%=doc/%.1)
+MANPAGES3   = $(LIBRARIES-yes:%=doc/%.3)
+MANPAGES    = $(MANPAGES1) $(MANPAGES3)
+PODPAGES    = $(PROGS-yes:%=doc/%.pod)  $(PROGS-yes:%=doc/%-all.pod)  $(COMPONENTS-yes:%=doc/%.pod)  $(LIBRARIES-yes:%=doc/%.pod)
+HTMLPAGES   = $(PROGS-yes:%=doc/%.html) $(PROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
              doc/developer.html                                        \
              doc/faq.html                                              \
              doc/fate.html                                             \
@@ -59,12 +60,22 @@ $(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
 doc/%.html: TAG = HTML
 doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
 	$(Q)$(TEXIDEP)
-	$(M)texi2html -I doc -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
+	$(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 $@ $<

 doc/%.pod: TAG = POD
 doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
 	$(Q)$(TEXIDEP)
-	$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Idoc $< $@
+	$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-not-all=yes -Idoc $< $@
+
+doc/%-all.pod: TAG = POD
+doc/%-all.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
+	$(Q)$(TEXIDEP)
+	$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-all=yes -Idoc $< $@

 doc/%.1 doc/%.3: TAG = MAN
 doc/%.1: doc/%.pod $(GENTEXI)
@@ -84,16 +95,22 @@ install-progs-$(CONFIG_DOC): install-man

 install-man: $(MANPAGES)
 	$(Q)mkdir -p "$(MANDIR)/man1"
-	$(INSTALL) -m 644 $(MANPAGES) "$(MANDIR)/man1"
+	$(INSTALL) -m 644 $(MANPAGES1) "$(MANDIR)/man1"
+	$(Q)mkdir -p "$(MANDIR)/man3"
+	$(INSTALL) -m 644 $(MANPAGES3) "$(MANDIR)/man3"
 endif

 uninstall: uninstall-man

 uninstall-man:
-	$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
+	$(RM) $(addprefix "$(MANDIR)/man1/",$(PROGS-yes:%=%.1) $(PROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
+	$(RM) $(addprefix "$(MANDIR)/man3/",$(LIBRARIES-yes:%=%.3))

 clean:: docclean

+distclean:: docclean
+	$(RM) doc/config.texi
+
 docclean:
 	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 $(CLEANSUFFIXES:%=doc/%) doc/avoptions_*.texi
 	$(RM) -r doc/doxy/html
--- a/doc/RELEASE_NOTES
+++ b/doc/RELEASE_NOTES
@@ -1,7 +1,7 @@
 Release Notes
 =============

-* 0.10 "Freedom"  January, 2012
+* 2.0 "Nameless"  July, 2013


 General notes
@@ -14,9 +14,3 @@ accepted. If you are experiencing issues with any formally released version of
 FFmpeg, please try git master to check if the issue still exists. If it does,
 make your report against the development code following the usual bug reporting
 guidelines.
-
-Of big interest to our Windows users, FFmpeg now supports building with the MSVC
-compiler. Since MSVC does not support C99 features used extensively by FFmpeg,
-this has been accomplished using a converter that turns C99 code to C89. See the
-platform-specific documentation for more detailed documentation on building
-FFmpeg with MSVC.
--- a/doc/avtools-common-opts.texi
+++ b/doc/avtools-common-opts.texi
@@ -1,32 +1,33 @@
-All the numerical options, if not specified otherwise, accept in input
-a string representing a number, which may contain one of the
-SI unit prefixes, for example 'K', 'M', 'G'.
-If 'i' is appended after the prefix, binary prefixes are used,
-which are based on powers of 1024 instead of powers of 1000.
-The 'B' postfix multiplies the value by 8, and can be
-appended after a unit prefix or used alone. This allows using for
-example 'KB', 'MiB', 'G' and 'B' as number postfix.
+All the numerical options, if not specified otherwise, accept a string
+representing a number as input, which may be followed by one of the SI
+unit prefixes, for example: 'K', 'M', or 'G'.
+
+If 'i' is appended to the SI unit prefix, the complete prefix will be
+interpreted as a unit prefix for binary multiplies, which are based on
+powers of 1024 instead of powers of 1000. Appending 'B' to the SI unit
+prefix multiplies the value by 8. This allows using, for example:
+'KB', 'MiB', 'G' and 'B' as number suffixes.

 Options which do not take arguments are boolean options, and set the
 corresponding value to true. They can be set to false by prefixing
-with "no" the option name, for example using "-nofoo" in the
-command line will set to false the boolean option with name "foo".
+the option name with "no". For example using "-nofoo"
+will set the boolean option with name "foo" to false.

@anchor{Stream specifiers}
@section Stream specifiers
 Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers
-are used to precisely specify which stream(s) does a given option belong to.
+are used to precisely specify which stream(s) a given option belongs to.

 A stream specifier is a string generally appended to the option name and
-separated from it by a colon. E.g. @code{-codec:a:1 ac3} option contains
-@code{a:1} stream specifier, which matches the second audio stream. Therefore it
+separated from it by a colon. E.g. @code{-codec:a:1 ac3} contains the
+@code{a:1} stream specifier, which matches the second audio stream. Therefore, it
 would select the ac3 codec for the second audio stream.

-A stream specifier can match several streams, the option is then applied to all
+A stream specifier can match several streams, so that the option is applied to all
 of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio
 streams.

-An empty stream specifier matches all streams, for example @code{-codec copy}
+An empty stream specifier matches all streams. For example, @code{-codec copy}
 or @code{-codec: copy} would copy all the streams without reencoding.

 Possible forms of stream specifiers are:
@@ -35,20 +36,21 @@ 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: 'v' for video, 'a' for audio, 's' for subtitle,
-'d' for data and 't' for attachments. If @var{stream_index} is given, then
-matches stream number @var{stream_index} of this type. Otherwise matches all
+@var{stream_type} is one of following: 'v' for video, 'a' for audio, 's' for subtitle,
+'d' for data, and 't' for attachments. If @var{stream_index} is given, then it matches
+stream number @var{stream_index} of this type. Otherwise, it matches all
 streams of this type.
@item p:@var{program_id}[:@var{stream_index}]
-If @var{stream_index} is given, then matches stream number @var{stream_index} in
-program with id @var{program_id}. Otherwise matches all streams in this program.
+If @var{stream_index} is given, then it matches the stream with number @var{stream_index}
+in the program with the id @var{program_id}. Otherwise, it matches all streams in the
+program.
@item #@var{stream_id}
-Matches the stream by format-specific ID.
+Matches the stream by a format-specific ID.
@end table

@section Generic options

-These options are shared amongst the av* tools.
+These options are shared amongst the ff* tools.

@table @option

@@ -77,6 +79,10 @@ Print detailed information about the demuxer named @var{demuxer_name}. Use the
 Print detailed information about the muxer named @var{muxer_name}. Use the
@option{-formats} option to get a list of all muxers and demuxers.

+@item filter=@var{filter_name}
+Print detailed information about the filter name @var{filter_name}. Use the
+@option{-filters} option to get a list of all filters.
+
@end table

@item -version
@@ -85,14 +91,6 @@ Show version.
@item -formats
 Show available formats.

-The fields preceding the format names have the following meanings:
-@table @samp
-@item D
-Decoding available
-@item E
-Encoding available
-@end table
-
@item -codecs
 Show all codecs known to libavcodec.

@@ -123,18 +121,36 @@ Show available sample formats.
@item -layouts
 Show channel names and standard channel layouts.

-@item -loglevel @var{loglevel} | -v @var{loglevel}
+@item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel}
 Set the logging level used by the library.
+Adding "repeat+" indicates that repeated log output should not be compressed
+to the first line and the "Last message repeated n times" line will be
+omitted. "repeat" can also be used alone.
+If "repeat" is used alone, and with no prior loglevel set, the default
+loglevel will be used. If multiple loglevel parameters are given, using
+'repeat' will not change the loglevel.
@var{loglevel} is a number or a string containing one of the following values:
@table @samp
@item quiet
+Show nothing at all; be silent.
@item panic
+Only show fatal errors which could lead the process to crash, such as
+and assert failure. This is not currently used for anything.
@item fatal
+Only show fatal errors. These are errors after which the process absolutely
+cannot continue after.
@item error
+Show all errors, including ones which can be recovered from.
@item warning
+Show all warnings and errors. Any message related to possibly
+incorrect or unexpected events will be shown.
@item info
+Show informative messages during processing. This is in addition to
+warnings and errors. This is the default value.
@item verbose
+Same as @code{info}, except more verbose.
@item debug
+Show everything, including debugging information.
@end table

 By default the program logs to stderr, if coloring is supported by the
@@ -176,7 +192,61 @@ ffmpeg -cpuflags -sse+mmx ...
 ffmpeg -cpuflags mmx ...
 ffmpeg -cpuflags 0 ...
@end example
+Possible flags for this option are:
+@table @samp
+@item x86
+@table @samp
+@item mmx
+@item mmxext
+@item sse
+@item sse2
+@item sse2slow
+@item sse3
+@item sse3slow
+@item ssse3
+@item atom
+@item sse4.1
+@item sse4.2
+@item avx
+@item xop
+@item fma4
+@item 3dnow
+@item 3dnowext
+@item cmov
+@end table
+@item ARM
+@table @samp
+@item armv5te
+@item armv6
+@item armv6t2
+@item vfp
+@item vfpv3
+@item neon
+@end table
+@item PowerPC
+@table @samp
+@item altivec
+@end table
+@item Specific Processors
+@table @samp
+@item pentium2
+@item pentium3
+@item pentium4
+@item k6
+@item k62
+@item athlon
+@item athlonxp
+@item k8
+@end table
+@end table

+@item -opencl_options options (@emph{global})
+Set OpenCL environment options. This option is only available when
+FFmpeg has been compiled with @code{--enable-opencl}.
+
+@var{options} must be a list of @var{key}=@var{value} option pairs
+separated by ':'. See the ``OpenCL Options'' section in the
+ffmpeg-utils manual for the list of supported options.
@end table

@section AVOptions
--- a/doc/bitstream_filters.texi
+++ b/doc/bitstream_filters.texi
@@ -17,8 +17,19 @@ Below is a description of the currently available bitstream filters.

@section aac_adtstoasc

+Convert MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration
+bitstream filter.
+
+This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4
+ADTS header and removes the ADTS header.
+
+This is required for example when copying an AAC stream from a raw
+ADTS AAC container to a FLV or a MOV/MP4 file.
+
@section chomp

+Remove zero padding at the end of a packet.
+
@section dump_extradata

@section h264_mp4toannexb
--- a/doc/codecs.texi
+++ b/doc/codecs.texi
--- a/doc/decoders.texi
+++ b/doc/decoders.texi
@@ -60,6 +60,78 @@ This decoder generates wave patterns according to predefined sequences. Its
 use is purely internal and the format of the data it accepts is not publicly
 documented.

+@section libcelt
+
+libcelt decoder wrapper.
+
+libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec.
+Requires the presence of the libcelt headers and library during configuration.
+You need to explicitly configure the build with @code{--enable-libcelt}.
+
+@section libgsm
+
+libgsm decoder wrapper.
+
+libgsm allows libavcodec to decode the GSM full rate audio codec. Requires
+the presence of the libgsm headers and library during configuration. You need
+to explicitly configure the build with @code{--enable-libgsm}.
+
+This decoder supports both the ordinary GSM and the Microsoft variant.
+
+@section libilbc
+
+libilbc decoder wrapper.
+
+libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC)
+audio codec. Requires the presence of the libilbc headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libilbc}.
+
+@subsection Options
+
+The following option is supported by the libilbc wrapper.
+
+@table @option
+@item enhance
+
+Enable the enhancement of the decoded audio when set to 1. The default
+value is 0 (disabled).
+
+@end table
+
+@section libopencore-amrnb
+
+libopencore-amrnb decoder wrapper.
+
+libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate
+Narrowband audio codec. Using it requires the presence of the
+libopencore-amrnb headers and library during configuration. You need to
+explicitly configure the build with @code{--enable-libopencore-amrnb}.
+
+An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB
+without this library.
+
+@section libopencore-amrwb
+
+libopencore-amrwb decoder wrapper.
+
+libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate
+Wideband audio codec. Using it requires the presence of the
+libopencore-amrwb headers and library during configuration. You need to
+explicitly configure the build with @code{--enable-libopencore-amrwb}.
+
+An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB
+without this library.
+
+@section libopus
+
+libopus decoder wrapper.
+
+libopus allows libavcodec to decode the Opus Interactive Audio Codec.
+Requires the presence of the libopus headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libopus}.
+
@c man end AUDIO DECODERS

@chapter Subtitles Decoders
--- a/doc/demuxers.texi
+++ b/doc/demuxers.texi
@@ -78,6 +78,9 @@ Duration of the file. This information can be specified from the file;
 specifying it here may be more efficient or help if the information from the
 file is not available or accurate.

+If the duration is set for all files, then it is possible to seek in the
+whole concatenated video.
+
@end table

@subsection Options
@@ -100,6 +103,31 @@ probed and 0 otherwise.

@end table

+@section libgme
+
+The Game Music Emu library is a collection of video game music file emulators.
+
+See @url{http://code.google.com/p/game-music-emu/} for more information.
+
+Some files have multiple tracks. The demuxer will pick the first track by
+default. The @option{track_index} option can be used to select a different
+track. Track indexes start at 0. The demuxer exports the number of tracks as
+@var{tracks} meta data entry.
+
+For very large files, the @option{max_size} option may have to be adjusted.
+
+@section libquvi
+
+Play media from Internet services using the quvi project.
+
+The demuxer accepts a @option{format} option to request a specific quality. It
+is by default set to @var{best}.
+
+See @url{http://quvi.sourceforge.net/} for more information.
+
+FFmpeg needs to be built with @code{--enable-libquvi} for this demuxer to be
+enabled.
+
@section image2

 Image file demuxer.
@@ -117,7 +145,7 @@ same for all the files in the sequence.
 This demuxer accepts the following options:
@table @option
@item framerate
-Set the framerate for the video stream. It defaults to 25.
+Set the frame rate for the video stream. It defaults to 25.
@item loop
 If set to 1, loop over the input. Default value is 0.
@item pattern_type
@@ -195,6 +223,10 @@ to read from. Default value is 0.
 Set the index interval range to check when looking for the first image
 file in the sequence, starting from @var{start_number}. Default value
 is 5.
+@item ts_from_file
+If set to 1, will set frame timestamp to modification time of image file. Note
+that monotonity of timestamps is not provided: images go in the same order as
+without this option. Default value is 0.
@item video_size
 Set the video size of the images to read. If not specified the video
 size is guessed from the first image file in the sequence.
--- a/doc/developer.texi
+++ b/doc/developer.texi
@@ -157,13 +157,16 @@ There are the following conventions for naming variables and functions:
@item
 For local variables no prefix is required.
@item
-For variables and functions declared as @code{static} no prefix is required.
+For file-scope variables and functions declared as @code{static}, no prefix
+is required.
@item
-For variables and functions used internally by a library an @code{ff_}
-prefix should be used, e.g. @samp{ff_w64_demuxer}.
+For variables and functions visible outside of file scope, but only used
+internally by a library, an @code{ff_} prefix should be used,
+e.g. @samp{ff_w64_demuxer}.
@item
-For variables and functions used internally across multiple libraries, use
-@code{avpriv_}. For example, @samp{avpriv_aac_parse_header}.
+For variables and functions visible outside of file scope, used internally
+across multiple libraries, use @code{avpriv_} as prefix, for example,
+@samp{avpriv_aac_parse_header}.
@item
 Each library has its own prefix for public symbols, in addition to the
 commonly used @code{av_} (@code{avformat_} for libavformat,
@@ -577,4 +580,129 @@ message or introductionary message for the patch series that you post to
 the ffmpeg-devel mailing list, a direct link to download the sample media.


+@subsection Visualizing Test Coverage
+
+The FFmpeg build system allows visualizing the test coverage in an easy
+manner with the coverage tools @code{gcov}/@code{lcov}.  This involves
+the following steps:
+
+@enumerate
+@item
+    Configure to compile with instrumentation enabled:
+    @code{configure --toolchain=gcov}.
+@item
+    Run your test case, either manually or via FATE. This can be either
+    the full FATE regression suite, or any arbitrary invocation of any
+    front-end tool provided by FFmpeg, in any combination.
+@item
+    Run @code{make lcov} to generate coverage data in HTML format.
+@item
+    View @code{lcov/index.html} in your preferred HTML viewer.
+@end enumerate
+
+You can use the command @code{make lcov-reset} to reset the coverage
+measurements. You will need to rerun @code{make lcov} after running a
+new test.
+
+@subsection Using Valgrind
+
+The configure script provides a shortcut for using valgrind to spot bugs
+related to memory handling. Just add the option
+@code{--toolchain=valgrind-memcheck} or @code{--toolchain=valgrind-massif}
+to your configure line, and reasonable defaults will be set for running
+FATE under the supervision of either the @strong{memcheck} or the
+@strong{massif} tool of the valgrind suite.
+
+In case you need finer control over how valgrind is invoked, use the
+@code{--target-exec='valgrind <your_custom_valgrind_options>} option in
+your configure line instead.
+
+@anchor{Release process}
+@section Release process
+
+FFmpeg maintains a set of @strong{release branches}, which are the
+recommended deliverable for system integrators and distributors (such as
+Linux distributions, etc.). At regular times, a @strong{release
+manager} prepares, tests and publishes tarballs on the
+@url{http://ffmpeg.org} website.
+
+There are two kinds of releases:
+
+@enumerate
+@item
+    @strong{Major releases} always include the latest and greatest
+    features and functionality.
+@item
+    @strong{Point releases} are cut from @strong{release} branches,
+    which are named @code{release/X}, with @code{X} being the release
+    version number.
+@end enumerate
+
+Note that we promise to our users that shared libraries from any FFmpeg
+release never break programs that have been @strong{compiled} against
+previous versions of @strong{the same release series} in any case!
+
+However, from time to time, we do make API changes that require adaptations
+in applications. Such changes are only allowed in (new) major releases and
+require further steps such as bumping library version numbers and/or
+adjustments to the symbol versioning file. Please discuss such changes
+on the @strong{ffmpeg-devel} mailing list in time to allow forward planning.
+
+@anchor{Criteria for Point Releases}
+@subsection Criteria for Point Releases
+
+Changes that match the following criteria are valid candidates for
+inclusion into a point release:
+
+@enumerate
+@item
+    Fixes a security issue, preferably identified by a @strong{CVE
+    number} issued by @url{http://cve.mitre.org/}.
+@item
+    Fixes a documented bug in @url{https://trac.ffmpeg.org}.
+@item
+    Improves the included documentation.
+@item
+    Retains both source code and binary compatibility with previous
+    point releases of the same release branch.
+@end enumerate
+
+The order for checking the rules is (1 OR 2 OR 3) AND 4.
+
+
+@subsection Release Checklist
+
+The release process involves the following steps:
+
+@enumerate
+@item
+    Ensure that the @file{RELEASE} file contains the version number for
+    the upcoming release.
+@item
+    Add the release at @url{https://trac.ffmpeg.org/admin/ticket/versions}.
+@item
+    Announce the intent to do a release to the mailing list.
+@item
+    Make sure all relevant security fixes have been backported. See
+    @url{https://ffmpeg.org/security.html}.
+@item
+    Ensure that the FATE regression suite still passes in the release
+    branch on at least @strong{i386} and @strong{amd64}
+    (cf. @ref{Regression tests}).
+@item
+    Prepare the release tarballs in @code{bz2} and @code{gz} formats, and
+    supplementing files that contain @code{gpg} signatures
+@item
+    Publish the tarballs at @url{http://ffmpeg.org/releases}. Create and
+    push an annotated tag in the form @code{nX}, with @code{X}
+    containing the version number.
+@item
+    Propose and send a patch to the @strong{ffmpeg-devel} mailing list
+    with a news entry for the website.
+@item
+    Publish the news entry.
+@item
+    Send announcement to the mailing list.
+@end enumerate
+
@bye
--- a/doc/devices.texi
+++ b/doc/devices.texi
@@ -0,0 +1,21 @@
+@chapter Device Options
+@c man begin DEVICE OPTIONS
+
+The libavdevice library provides the same interface as
+libavformat. Namely, an input device is considered like a demuxer, and
+an output device like a muxer, and the interface and generic device
+options are the same provided by libavformat (see the ffmpeg-formats
+manual).
+
+In addition each input or output device may support so-called private
+options, which are specific for that component.
+
+Options may be set by specifying -@var{option} @var{value} in the
+FFmpeg tools, or by setting the value explicitly in the device
+@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
+for programmatic use.
+
+@c man end DEVICE OPTIONS
+
+@include indevs.texi
+@include outdevs.texi
--- a/doc/encoders.texi
+++ b/doc/encoders.texi
--- a/doc/examples/Makefile
+++ b/doc/examples/Makefile
@@ -7,7 +7,7 @@ FFMPEG_LIBS=    libavdevice                        \
                libswscale                         \
                libavutil                          \

-CFLAGS += -Wall -O2 -g
+CFLAGS += -Wall -g
 CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
 LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)

--- a/doc/examples/decoding_encoding.c
+++ b/doc/examples/decoding_encoding.c
@@ -79,7 +79,7 @@ static int select_channel_layout(AVCodec *codec)
 {
    const uint64_t *p;
    uint64_t best_ch_layout = 0;
-    int best_nb_channells   = 0;
+    int best_nb_channels   = 0;

    if (!codec->channel_layouts)
        return AV_CH_LAYOUT_STEREO;
@@ -88,9 +88,9 @@ static int select_channel_layout(AVCodec *codec)
    while (*p) {
        int nb_channels = av_get_channel_layout_nb_channels(*p);

-        if (nb_channels > best_nb_channells) {
+        if (nb_channels > best_nb_channels) {
            best_ch_layout    = *p;
-            best_nb_channells = nb_channels;
+            best_nb_channels = nb_channels;
        }
        p++;
    }
--- a/doc/examples/filtering_audio.c
+++ b/doc/examples/filtering_audio.c
@@ -36,8 +36,9 @@
 #include <libavfilter/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
+#include <libavutil/opt.h>

-const char *filter_descr = "aresample=8000,aconvert=s16:mono";
+const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
 const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";

 static AVFormatContext *fmt_ctx;
@@ -70,6 +71,7 @@ static int open_input_file(const char *filename)
    }
    audio_stream_index = ret;
    dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
+    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);

    /* init the audio decoder */
    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -85,11 +87,12 @@ static int init_filters(const char *filters_descr)
    char args[512];
    int ret;
    AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
-    AVFilter *abuffersink = avfilter_get_by_name("ffabuffersink");
+    AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
    AVFilterInOut *outputs = avfilter_inout_alloc();
    AVFilterInOut *inputs  = avfilter_inout_alloc();
-    const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
-    AVABufferSinkParams *abuffersink_params;
+    const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
+    const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
+    const int out_sample_rates[] = { 8000, -1 };
    const AVFilterLink *outlink;
    AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;

@@ -110,16 +113,34 @@ static int init_filters(const char *filters_descr)
    }

    /* buffer audio sink: to terminate the filter chain. */
-    abuffersink_params = av_abuffersink_params_alloc();
-    abuffersink_params->sample_fmts     = sample_fmts;
    ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
-                                       NULL, abuffersink_params, filter_graph);
-    av_free(abuffersink_params);
+                                       NULL, NULL, filter_graph);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
        return ret;
    }

+    ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
+                              AV_OPT_SEARCH_CHILDREN);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
+        return ret;
+    }
+
+    ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
+                              AV_OPT_SEARCH_CHILDREN);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
+        return ret;
+    }
+
+    ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
+                              AV_OPT_SEARCH_CHILDREN);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
+        return ret;
+    }
+
    /* Endpoints for the filter graph. */
    outputs->name       = av_strdup("in");
    outputs->filter_ctx = buffersrc_ctx;
@@ -131,7 +152,7 @@ static int init_filters(const char *filters_descr)
    inputs->pad_idx    = 0;
    inputs->next       = NULL;

-    if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
+    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
                                    &inputs, &outputs, NULL)) < 0)
        return ret;

@@ -150,11 +171,10 @@ static int init_filters(const char *filters_descr)
    return 0;
 }

-static void print_samplesref(AVFilterBufferRef *samplesref)
+static void print_frame(const AVFrame *frame)
 {
-    const AVFilterBufferRefAudioProps *props = samplesref->audio;
-    const int n = props->nb_samples * av_get_channel_layout_nb_channels(props->channel_layout);
-    const uint16_t *p     = (uint16_t*)samplesref->data[0];
+    const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame));
+    const uint16_t *p     = (uint16_t*)frame->data[0];
    const uint16_t *p_end = p + n;

    while (p < p_end) {
@@ -169,10 +189,11 @@ int main(int argc, char **argv)
 {
    int ret;
    AVPacket packet;
-    AVFrame *frame = avcodec_alloc_frame();
+    AVFrame *frame = av_frame_alloc();
+    AVFrame *filt_frame = av_frame_alloc();
    int got_frame;

-    if (!frame) {
+    if (!frame || !filt_frame) {
        perror("Could not allocate frame");
        exit(1);
    }
@@ -192,7 +213,6 @@ int main(int argc, char **argv)

    /* read all packets */
    while (1) {
-        AVFilterBufferRef *samplesref;
        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
            break;

@@ -207,22 +227,20 @@ int main(int argc, char **argv)

            if (got_frame) {
                /* push the audio data from decoded frame into the filtergraph */
-                if (av_buffersrc_add_frame(buffersrc_ctx, frame, 0) < 0) {
+                if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, 0) < 0) {
                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
                    break;
                }

                /* pull filtered audio from the filtergraph */
                while (1) {
-                    ret = av_buffersink_get_buffer_ref(buffersink_ctx, &samplesref, 0);
+                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
                    if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                        break;
                    if(ret < 0)
                        goto end;
-                    if (samplesref) {
-                        print_samplesref(samplesref);
-                        avfilter_unref_bufferp(&samplesref);
-                    }
+                    print_frame(filt_frame);
+                    av_frame_unref(filt_frame);
                }
            }
        }
@@ -233,7 +251,8 @@ end:
    if (dec_ctx)
        avcodec_close(dec_ctx);
    avformat_close_input(&fmt_ctx);
-    av_freep(&frame);
+    av_frame_free(&frame);
+    av_frame_free(&filt_frame);

    if (ret < 0 && ret != AVERROR_EOF) {
        char buf[1024];
--- a/doc/examples/filtering_video.c
+++ b/doc/examples/filtering_video.c
@@ -85,7 +85,7 @@ static int init_filters(const char *filters_descr)
    char args[512];
    int ret;
    AVFilter *buffersrc  = avfilter_get_by_name("buffer");
-    AVFilter *buffersink = avfilter_get_by_name("ffbuffersink");
+    AVFilter *buffersink = avfilter_get_by_name("buffersink");
    AVFilterInOut *outputs = avfilter_inout_alloc();
    AVFilterInOut *inputs  = avfilter_inout_alloc();
    enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
@@ -129,7 +129,7 @@ static int init_filters(const char *filters_descr)
    inputs->pad_idx    = 0;
    inputs->next       = NULL;

-    if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
+    if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
                                    &inputs, &outputs, NULL)) < 0)
        return ret;

@@ -138,33 +138,33 @@ static int init_filters(const char *filters_descr)
    return 0;
 }

-static void display_picref(AVFilterBufferRef *picref, AVRational time_base)
+static void display_frame(const AVFrame *frame, AVRational time_base)
 {
    int x, y;
    uint8_t *p0, *p;
    int64_t delay;

-    if (picref->pts != AV_NOPTS_VALUE) {
+    if (frame->pts != AV_NOPTS_VALUE) {
        if (last_pts != AV_NOPTS_VALUE) {
            /* sleep roughly the right amount of time;
             * usleep is in microseconds, just like AV_TIME_BASE. */
-            delay = av_rescale_q(picref->pts - last_pts,
+            delay = av_rescale_q(frame->pts - last_pts,
                                 time_base, AV_TIME_BASE_Q);
            if (delay > 0 && delay < 1000000)
                usleep(delay);
        }
-        last_pts = picref->pts;
+        last_pts = frame->pts;
    }

    /* Trivial ASCII grayscale display. */
-    p0 = picref->data[0];
+    p0 = frame->data[0];
    puts("\033c");
-    for (y = 0; y < picref->video->h; y++) {
+    for (y = 0; y < frame->height; y++) {
        p = p0;
-        for (x = 0; x < picref->video->w; x++)
+        for (x = 0; x < frame->width; x++)
            putchar(" .-+#"[*(p++) / 52]);
        putchar('\n');
-        p0 += picref->linesize[0];
+        p0 += frame->linesize[0];
    }
    fflush(stdout);
 }
@@ -173,10 +173,11 @@ int main(int argc, char **argv)
 {
    int ret;
    AVPacket packet;
-    AVFrame *frame = avcodec_alloc_frame();
+    AVFrame *frame = av_frame_alloc();
+    AVFrame *filt_frame = av_frame_alloc();
    int got_frame;

-    if (!frame) {
+    if (!frame || !filt_frame) {
        perror("Could not allocate frame");
        exit(1);
    }
@@ -196,7 +197,6 @@ int main(int argc, char **argv)

    /* read all packets */
    while (1) {
-        AVFilterBufferRef *picref;
        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
            break;

@@ -213,23 +213,20 @@ int main(int argc, char **argv)
                frame->pts = av_frame_get_best_effort_timestamp(frame);

                /* push the decoded frame into the filtergraph */
-                if (av_buffersrc_add_frame(buffersrc_ctx, frame, 0) < 0) {
+                if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
                    break;
                }

-                /* pull filtered pictures from the filtergraph */
+                /* pull filtered frames from the filtergraph */
                while (1) {
-                    ret = av_buffersink_get_buffer_ref(buffersink_ctx, &picref, 0);
+                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                        break;
                    if (ret < 0)
                        goto end;
-
-                    if (picref) {
-                        display_picref(picref, buffersink_ctx->inputs[0]->time_base);
-                        avfilter_unref_bufferp(&picref);
-                    }
+                    display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
+                    av_frame_unref(filt_frame);
                }
            }
        }
@@ -240,7 +237,8 @@ end:
    if (dec_ctx)
        avcodec_close(dec_ctx);
    avformat_close_input(&fmt_ctx);
-    av_freep(&frame);
+    av_frame_free(&frame);
+    av_frame_free(&filt_frame);

    if (ret < 0 && ret != AVERROR_EOF) {
        char buf[1024];
--- a/doc/examples/muxing.c
+++ b/doc/examples/muxing.c
@@ -34,9 +34,11 @@
 #include <string.h>
 #include <math.h>

+#include <libavutil/opt.h>
 #include <libavutil/mathematics.h>
 #include <libavformat/avformat.h>
 #include <libswscale/swscale.h>
+#include <libswresample/swresample.h>

 /* 5 seconds stream duration */
 #define STREAM_DURATION   200.0
@@ -46,13 +48,6 @@

 static int sws_flags = SWS_BICUBIC;

-/**************************************************************/
-/* audio output */
-
-static float t, tincr, tincr2;
-static int16_t *samples;
-static int audio_input_frame_size;
-
 /* Add an output stream. */
 static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
                            enum AVCodecID codec_id)
@@ -78,8 +73,7 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,

    switch ((*codec)->type) {
    case AVMEDIA_TYPE_AUDIO:
-        st->id = 1;
-        c->sample_fmt  = AV_SAMPLE_FMT_S16;
+        c->sample_fmt  = AV_SAMPLE_FMT_FLTP;
        c->bit_rate    = 64000;
        c->sample_rate = 44100;
        c->channels    = 2;
@@ -127,8 +121,17 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
 /* audio output */

 static float t, tincr, tincr2;
-static int16_t *samples;
-static int audio_input_frame_size;
+
+static uint8_t **src_samples_data;
+static int       src_samples_linesize;
+static int       src_nb_samples;
+
+static int max_dst_nb_samples;
+uint8_t **dst_samples_data;
+int       dst_samples_linesize;
+int       dst_samples_size;
+
+struct SwrContext *swr_ctx = NULL;

 static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
 {
@@ -150,17 +153,51 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
    /* increment frequency by 110 Hz per second */
    tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;

-    if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
-        audio_input_frame_size = 10000;
-    else
-        audio_input_frame_size = c->frame_size;
-    samples = av_malloc(audio_input_frame_size *
-                        av_get_bytes_per_sample(c->sample_fmt) *
-                        c->channels);
-    if (!samples) {
-        fprintf(stderr, "Could not allocate audio samples buffer\n");
+    src_nb_samples = c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE ?
+        10000 : c->frame_size;
+
+    ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels,
+                                             src_nb_samples, c->sample_fmt, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate source samples\n");
        exit(1);
    }
+
+    /* create resampler context */
+    if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
+        swr_ctx = swr_alloc();
+        if (!swr_ctx) {
+            fprintf(stderr, "Could not allocate resampler context\n");
+            exit(1);
+        }
+
+        /* set options */
+        av_opt_set_int       (swr_ctx, "in_channel_count",   c->channels,       0);
+        av_opt_set_int       (swr_ctx, "in_sample_rate",     c->sample_rate,    0);
+        av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
+        av_opt_set_int       (swr_ctx, "out_channel_count",  c->channels,       0);
+        av_opt_set_int       (swr_ctx, "out_sample_rate",    c->sample_rate,    0);
+        av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
+
+        /* initialize the resampling context */
+        if ((ret = swr_init(swr_ctx)) < 0) {
+            fprintf(stderr, "Failed to initialize the resampling context\n");
+            exit(1);
+        }
+    }
+
+    /* compute the number of converted samples: buffering is avoided
+     * ensuring that the output buffer will contain at least all the
+     * converted input samples */
+    max_dst_nb_samples = src_nb_samples;
+    ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels,
+                                             max_dst_nb_samples, c->sample_fmt, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate destination samples\n");
+        exit(1);
+    }
+    dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples,
+                                                  c->sample_fmt, 0);
 }

 /* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
@@ -185,18 +222,45 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
    AVCodecContext *c;
    AVPacket pkt = { 0 }; // data and size must be 0;
    AVFrame *frame = avcodec_alloc_frame();
-    int got_packet, ret;
+    int got_packet, ret, dst_nb_samples;

    av_init_packet(&pkt);
    c = st->codec;

-    get_audio_frame(samples, audio_input_frame_size, c->channels);
-    frame->nb_samples = audio_input_frame_size;
+    get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
+
+    /* convert samples from native format to destination codec format, using the resampler */
+    if (swr_ctx) {
+        /* compute destination number of samples */
+        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
+                                        c->sample_rate, c->sample_rate, AV_ROUND_UP);
+        if (dst_nb_samples > max_dst_nb_samples) {
+            av_free(dst_samples_data[0]);
+            ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
+                                   dst_nb_samples, c->sample_fmt, 0);
+            if (ret < 0)
+                exit(1);
+            max_dst_nb_samples = dst_nb_samples;
+            dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
+                                                          c->sample_fmt, 0);
+        }
+
+        /* convert to destination format */
+        ret = swr_convert(swr_ctx,
+                          dst_samples_data, dst_nb_samples,
+                          (const uint8_t **)src_samples_data, src_nb_samples);
+        if (ret < 0) {
+            fprintf(stderr, "Error while converting\n");
+            exit(1);
+        }
+    } else {
+        dst_samples_data[0] = src_samples_data[0];
+        dst_nb_samples = src_nb_samples;
+    }
+
+    frame->nb_samples = dst_nb_samples;
    avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
-                             (uint8_t *)samples,
-                             audio_input_frame_size *
-                             av_get_bytes_per_sample(c->sample_fmt) *
-                             c->channels, 1);
+                             dst_samples_data[0], dst_samples_size, 0);

    ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
    if (ret < 0) {
@@ -222,8 +286,8 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
 static void close_audio(AVFormatContext *oc, AVStream *st)
 {
    avcodec_close(st->codec);
-
-    av_free(samples);
+    av_free(src_samples_data[0]);
+    av_free(dst_samples_data[0]);
 }

 /**************************************************************/
@@ -342,25 +406,19 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)

        ret = av_interleaved_write_frame(oc, &pkt);
    } else {
-        /* encode the image */
-        AVPacket pkt;
-        int got_output;
-
+        AVPacket pkt = { 0 };
+        int got_packet;
        av_init_packet(&pkt);
-        pkt.data = NULL;    // packet data will be allocated by the encoder
-        pkt.size = 0;

-        ret = avcodec_encode_video2(c, &pkt, frame, &got_output);
+        /* encode the image */
+        ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
        if (ret < 0) {
            fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
            exit(1);
        }
-
        /* If size is zero, it means the image was buffered. */
-        if (got_output) {
-            if (c->coded_frame->key_frame)
-                pkt.flags |= AV_PKT_FLAG_KEY;

+        if (!ret && got_packet && pkt.size) {
            pkt.stream_index = st->index;

            /* Write the compressed frame to the media file. */
@@ -394,7 +452,7 @@ int main(int argc, char **argv)
    AVFormatContext *oc;
    AVStream *audio_st, *video_st;
    AVCodec *audio_codec, *video_codec;
-    double audio_pts, video_pts;
+    double audio_time, video_time;
    int ret;

    /* Initialize libavcodec, and register all codecs and formats. */
@@ -467,23 +525,15 @@ int main(int argc, char **argv)
        frame->pts = 0;
    for (;;) {
        /* Compute current audio and video time. */
-        if (audio_st)
-            audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
-        else
-            audio_pts = 0.0;
+        audio_time = audio_st ? audio_st->pts.val * av_q2d(audio_st->time_base) : 0.0;
+        video_time = video_st ? video_st->pts.val * av_q2d(video_st->time_base) : 0.0;

-        if (video_st)
-            video_pts = (double)video_st->pts.val * video_st->time_base.num /
-                        video_st->time_base.den;
-        else
-            video_pts = 0.0;
-
-        if ((!audio_st || audio_pts >= STREAM_DURATION) &&
-            (!video_st || video_pts >= STREAM_DURATION))
+        if ((!audio_st || audio_time >= STREAM_DURATION) &&
+            (!video_st || video_time >= STREAM_DURATION))
            break;

        /* write interleaved audio and video frames */
-        if (!video_st || (video_st && audio_st && audio_pts < video_pts)) {
+        if (!video_st || (video_st && audio_st && audio_time < video_time)) {
            write_audio_frame(oc, audio_st);
        } else {
            write_video_frame(oc, video_st);
--- a/doc/examples/resampling_audio.c
+++ b/doc/examples/resampling_audio.c
@@ -78,18 +78,6 @@ void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate,
    }
 }

-int alloc_samples_array_and_data(uint8_t ***data, int *linesize, int nb_channels,
-                                    int nb_samples, enum AVSampleFormat sample_fmt, int align)
-{
-    int nb_planes = av_sample_fmt_is_planar(sample_fmt) ? nb_channels : 1;
-
-    *data = av_malloc(sizeof(*data) * nb_planes);
-    if (!*data)
-        return AVERROR(ENOMEM);
-    return av_samples_alloc(*data, linesize, nb_channels,
-                            nb_samples, sample_fmt, align);
-}
-
 int main(int argc, char **argv)
 {
    int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
@@ -149,8 +137,8 @@ int main(int argc, char **argv)
    /* allocate source and destination samples buffers */

    src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
-    ret = alloc_samples_array_and_data(&src_data, &src_linesize, src_nb_channels,
-                                       src_nb_samples, src_sample_fmt, 0);
+    ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
+                                             src_nb_samples, src_sample_fmt, 0);
    if (ret < 0) {
        fprintf(stderr, "Could not allocate source samples\n");
        goto end;
@@ -164,8 +152,8 @@ int main(int argc, char **argv)

    /* buffer is going to be directly written to a rawaudio file, no alignment */
    dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
-    ret = alloc_samples_array_and_data(&dst_data, &dst_linesize, dst_nb_channels,
-                                       dst_nb_samples, dst_sample_fmt, 0);
+    ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
+                                             dst_nb_samples, dst_sample_fmt, 0);
    if (ret < 0) {
        fprintf(stderr, "Could not allocate destination samples\n");
        goto end;
--- a/doc/faq.texi
+++ b/doc/faq.texi
@@ -393,17 +393,17 @@ Appending @code{:v} to it will do exactly that.
 Use @option{-dumpgraph -} to find out exactly where the channel layout is
 lost.

-Most likely, it is through @code{auto-inserted aconvert}. Try to understand
+Most likely, it is through @code{auto-inserted aresample}. Try to understand
 why the converting filter was needed at that place.

 Just before the output is a likely place, as @option{-f lavfi} currently
 only support packed S16.

-Then insert the correct @code{aconvert} explicitly in the filter graph,
+Then insert the correct @code{aformat} explicitly in the filtergraph,
 specifying the exact format.

@example
-aconvert=s16:stereo:packed
+aformat=sample_fmts=s16:channel_layouts=stereo
@end example

@section Why does FFmpeg not see the subtitles in my VOB file?
--- a/doc/fate.texi
+++ b/doc/fate.texi
@@ -27,7 +27,7 @@ by visiting this website:

  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 there recent contribution. This usually happens on the platforms
+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
@@ -131,7 +131,12 @@ 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:

-  b1:31:c8:79:3f:04:1d:f8:f2:23:26:5a:fd:55:fa:92
+@table @option
+@item RSA
+   d3:f1:83:97:a4:75:2b:a6:fb:d6:e8:aa:81:93:97:51
+@item ECDSA
+   76:9f:68:32:04:1e:d5:d4:ec:47:3f:dc:fc:18:17:86
+@end table

  If you have problems connecting to the FATE server, it may help to try out
 the @command{ssh} command with one or more @option{-v} options. You should
@@ -185,6 +190,8 @@ the synchronisation of the samples directory.
    The @var{TARGET_EXEC} option provides a way to run FATE wrapped in
    @command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
    through @command{ssh}.
+@item GEN
+Set to @var{1} to generate the missing or mismatched references.
@end table

@section Examples
--- a/doc/fate_config.sh.template
+++ b/doc/fate_config.sh.template
@@ -4,16 +4,20 @@ samples=                                 # path to samples directory
 workdir=                                 # directory in which to do all the work
 #fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report
 comment=                                 # optional description
+build_only=     # set to "yes" for a compile-only instance that skips tests

 # the following are optional and map to configure options
 arch=
 cpu=
 cross_prefix=
+as=
 cc=
+ld=
 target_os=
 sysroot=
 target_exec=
 target_path=
+target_samples=
 extra_cflags=
 extra_ldflags=
 extra_libs=
--- a/doc/ffmpeg-codecs.texi
+++ b/doc/ffmpeg-codecs.texi
--- a/doc/ffmpeg-devices.texi
+++ b/doc/ffmpeg-devices.texi
@@ -17,27 +17,7 @@ libavdevice library.

@c man end DESCRIPTION

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

@chapter See Also

--- a/doc/ffmpeg-formats.texi
+++ b/doc/ffmpeg-formats.texi
@@ -17,138 +17,7 @@ provided by the libavformat library.

@c man end DESCRIPTION

-@chapter Format Options
-@c man begin FORMAT OPTIONS
-
-The libavformat library provides some generic global options, which
-can be set on all the muxers and demuxers. In addition each muxer or
-demuxer may support so-called private options, which are specific for
-that component.
-
-Options may be set by specifying -@var{option} @var{value} in the
-FFmpeg tools, or by setting the value explicitly in the
-@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
-for programmatic use.
-
-The list of supported options follows:
-
-@table @option
-@item avioflags @var{flags} (@emph{input/output})
-Possible values:
-@table @samp
-@item direct
-Reduce buffering.
-@end table
-
-@item probesize @var{integer} (@emph{input})
-Set probing size in bytes, i.e. the size of the data to analyze to get
-stream information. A higher value will allow to detect more
-information in case it is dispersed into the stream, but will increase
-latency. Must be an integer not lesser than 32. It is 5000000 by default.
-
-@item packetsize @var{integer} (@emph{output})
-Set packet size.
-
-@item fflags @var{flags} (@emph{input/output})
-Set format flags.
-
-Possible values:
-@table @samp
-@item ignidx
-Ignore index.
-@item genpts
-Generate PTS.
-@item nofillin
-Do not fill in missing values that can be exactly calculated.
-@item noparse
-Disable AVParsers, this needs @code{+nofillin} too.
-@item igndts
-Ignore DTS.
-@item discardcorrupt
-Discard corrupted frames.
-@item sortdts
-Try to interleave output packets by DTS.
-@item keepside
-Do not merge side data.
-@item latm
-Enable RTP MP4A-LATM payload.
-@item nobuffer
-Reduce the latency introduced by optional buffering
-@end table
-
-@item analyzeduration @var{integer} (@emph{input})
-Specify how many microseconds are analyzed to probe the input. A
-higher value will allow to detect more accurate information, but will
-increase latency. It defaults to 5,000,000 microseconds = 5 seconds.
-
-@item cryptokey @var{hexadecimal string} (@emph{input})
-Set decryption key.
-
-@item indexmem @var{integer} (@emph{input})
-Set max memory used for timestamp index (per stream).
-
-@item rtbufsize @var{integer} (@emph{input})
-Set max memory used for buffering real-time frames.
-
-@item fdebug @var{flags} (@emph{input/output})
-Print specific debug info.
-
-Possible values:
-@table @samp
-@item ts
-@end table
-
-@item max_delay @var{integer} (@emph{input/output})
-Set maximum muxing or demuxing delay in microseconds.
-
-@item fpsprobesize @var{integer} (@emph{input})
-Set number of frames used to probe fps.
-
-@item audio_preload @var{integer} (@emph{output})
-Set microseconds by which audio packets should be interleaved earlier.
-
-@item chunk_duration @var{integer} (@emph{output})
-Set microseconds for each chunk.
-
-@item chunk_size @var{integer} (@emph{output})
-Set size in bytes for each chunk.
-
-@item err_detect, f_err_detect @var{flags} (@emph{input})
-Set error detection flags. @code{f_err_detect} is deprecated and
-should be used only via the @command{ffmpeg} tool.
-
-Possible values:
-@table @samp
-@item crccheck
-Verify embedded CRCs.
-@item bitstream
-Detect bitstream specification deviations.
-@item buffer
-Detect improper bitstream length.
-@item explode
-Abort decoding on minor error detection.
-@item careful
-Consider things that violate the spec and have not been seen in the
-wild as errors.
-@item compliant
-Consider all spec non compliancies as errors.
-@item aggressive
-Consider things that a sane encoder should not do as an error.
-@end table
-
-@item use_wallclock_as_timestamps @var{integer} (@emph{input})
-Use wallclock as timestamps.
-
-@item avoid_negative_ts @var{integer} (@emph{output})
-Shift timestamps to make them positive. 1 enables, 0 disables, default
-of -1 enables when required by target format.
-@end table
-
-@c man end FORMAT OPTIONS
-
-@include demuxers.texi
-@include muxers.texi
-@include metadata.texi
+@include formats.texi

@chapter See Also

--- a/doc/ffmpeg-resampler.texi
+++ b/doc/ffmpeg-resampler.texi
@@ -19,228 +19,7 @@ and convert audio format and packing layout.

@c man end DESCRIPTION

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

@chapter See Also

--- a/doc/ffmpeg-scaler.texi
+++ b/doc/ffmpeg-scaler.texi
@@ -18,105 +18,7 @@ image rescaling and pixel format conversion.

@c man end DESCRIPTION

-@chapter Scaler Options
-@c man begin SCALER OPTIONS
-
-The video scaler supports the following named options.
-
-Options may be set by specifying -@var{option} @var{value} in the
-FFmpeg tools. For programmatic use, they can be set explicitly in the
-@code{SwsContext} options or through the @file{libavutil/opt.h} API.
-
-@table @option
-
-@item sws_flags
-Set the scaler flags. This is also used to set the scaling
-algorithm. Only a single algorithm should be selected.
-
-It accepts the following values:
-@table @samp
-@item fast_bilinear
-Select fast bilinear scaling algorithm.
-
-@item bilinear
-Select bilinear scaling algorithm.
-
-@item bicubic
-Select bicubic scaling algorithm.
-
-@item experimental
-Select experimental scaling algorithm.
-
-@item neighbor
-Select nearest neighbor rescaling algorithm.
-
-@item area
-Select averaging area rescaling algorithm.
-
-@item bicubiclin
-Select bicubic scaling algorithm for the luma component, bilinear for
-chroma components.
-
-@item gauss
-Select Gaussian rescaling algorithm.
-
-@item sinc
-Select sinc rescaling algorithm.
-
-@item lanczos
-Select lanczos rescaling algorithm.
-
-@item spline
-Select natural bicubic spline rescaling algorithm.
-
-@item print_info
-Enable printing/debug logging.
-
-@item accurate_rnd
-Enable accurate rounding.
-
-@item full_chroma_int
-Enable full chroma interpolation.
-
-@item full_chroma_inp
-Select full chroma input.
-
-@item bitexact
-Enable bitexact output.
-@end table
-
-@item srcw
-Set source width.
-
-@item srch
-Set source height.
-
-@item dstw
-Set destination width.
-
-@item dsth
-Set destination height.
-
-@item src_format
-Set source pixel format (must be expressed as an integer).
-
-@item dst_format
-Set destination pixel format (must be expressed as an integer).
-
-@item src_range
-Select source range.
-
-@item dst_range
-Select destination range.
-
-@item param0, param1
-Set scaling algorithm parameters. The specified values are specific of
-some scaling algorithms and ignored by others. The specified values
-are floating point number values.
-
-@end table
-
-@c man end SCALER OPTIONS
+@include scaler.texi

@chapter See Also

--- a/doc/ffmpeg-utils.texi
+++ b/doc/ffmpeg-utils.texi
@@ -17,8 +17,7 @@ by the libavutil library.

@c man end DESCRIPTION

-@include syntax.texi
-@include eval.texi
+@include utils.texi

@chapter See Also

--- a/doc/ffmpeg.texi
+++ b/doc/ffmpeg.texi
@@ -16,26 +16,26 @@ ffmpeg [@var{global_options}] @{[@var{input_file_options}] -i @file{input_file}@
@chapter Description
@c man begin DESCRIPTION

-ffmpeg is a very fast video and audio converter that can also grab from
+@command{ffmpeg} is a very fast video and audio converter that can also grab from
 a live audio/video source. It can also convert between arbitrary sample
 rates and resize video on the fly with a high quality polyphase filter.

-ffmpeg reads from an arbitrary number of input "files" (which can be regular
+@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.

-Each input or output file can in principle contain any number of streams of
-different types (video/audio/subtitle/attachment/data). Allowed number and/or
-types of streams can be limited by the container format. Selecting, which
-streams from which inputs go into output, is done either automatically or with
-the @code{-map} option (see the Stream selection chapter).
+Each input or output file 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
+or with the @code{-map} option (see the Stream selection chapter).

 To refer to input files in options, you must use their indices (0-based). E.g.
-the first input file is @code{0}, the second is @code{1} etc. Similarly, streams
+the first input file is @code{0}, the second is @code{1}, etc. Similarly, streams
 within a file are referred to by their indices. E.g. @code{2:3} refers to the
-fourth stream in the third input file. See also the Stream specifiers chapter.
+fourth stream in the third input file. Also see the Stream specifiers chapter.

 As a general rule, options are applied to the next specified
 file. Therefore, order is important, and you can have the same
@@ -50,7 +50,7 @@ options apply ONLY to the next input or output file and are reset between files.

@itemize
@item
-To set the video bitrate of the output file to 64kbit/s:
+To set the video bitrate of the output file to 64 kbit/s:
@example
 ffmpeg -i input.avi -b:v 64k -bufsize 64k output.avi
@end example
@@ -96,14 +96,14 @@ tracking lowest timestamp on any active input stream.
 Encoded packets are then passed to the decoder (unless streamcopy is selected
 for the stream, see further for a description). The decoder produces
 uncompressed frames (raw video/PCM audio/...) which can be processed further by
-filtering (see next section). After filtering the frames are passed to the
-encoder, which encodes them and outputs encoded packets again. Finally those are
+filtering (see next section). After filtering, the frames are passed to the
+encoder, which encodes them and outputs encoded packets. Finally those are
 passed to the muxer, which writes the encoded packets to the output file.

@section Filtering
 Before encoding, @command{ffmpeg} can process raw audio and video frames using
 filters from the libavfilter library. Several chained filters form a filter
-graph.  @command{ffmpeg} distinguishes between two types of filtergraphs -
+graph. @command{ffmpeg} distinguishes between two types of filtergraphs:
 simple and complex.

@subsection Simple filtergraphs
@@ -139,7 +139,7 @@ only sets timestamps and otherwise passes the frames unchanged.

@subsection Complex filtergraphs
 Complex filtergraphs are those which cannot be described as simply a linear
-processing chain applied to one stream. This is the case e.g. when the graph has
+processing chain applied to one stream. This is the case, for example, when the graph has
 more than one input and/or output, or when output stream type is different from
 input. They can be represented with the following diagram:

@@ -164,9 +164,11 @@ input. They can be represented with the following diagram:
@end example

 Complex filtergraphs are configured with the @option{-filter_complex} option.
-Note that this option is global, since a complex filtergraph by its nature
+Note that this option is global, since a complex filtergraph, by its nature,
 cannot be unambiguously associated with a single stream or file.

+The @option{-lavfi} option is equivalent to @option{-filter_complex}.
+
 A trivial example of a complex filtergraph is the @code{overlay} filter, which
 has two video inputs and one video output, containing one video overlaid on top
 of the other. Its audio counterpart is the @code{amix} filter.
@@ -176,7 +178,7 @@ Stream copy is a mode selected by supplying the @code{copy} parameter to the
@option{-codec} option. It makes @command{ffmpeg} omit the decoding and encoding
 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:
+diagram above will, in this case, simplify to this:

@example
 _______              ______________            ________
@@ -188,7 +190,7 @@ diagram above will in this case simplify to this:
@end example

 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
+loss. However, it might not work in some cases because of many factors. Applying
 filters is obviously also impossible, since filters work on uncompressed data.

@c man end DETAILED DESCRIPTION
@@ -196,14 +198,14 @@ filters is obviously also impossible, since filters work on uncompressed data.
@chapter Stream selection
@c man begin STREAM SELECTION

-By default ffmpeg includes only one stream of each type (video, audio, subtitle)
+By default, @command{ffmpeg} includes only one stream of each type (video, audio, subtitle)
 present in the input files and adds them to each output file.  It picks the
-"best" of each based upon the following criteria; for video it is the stream
-with the highest resolution, for audio the stream with the most channels, for
-subtitle it's the first subtitle stream. In the case where several streams of
-the same type rate equally, the lowest numbered stream is chosen.
+"best" of each based upon the following criteria: for video, it is the stream
+with the highest resolution, for audio, it is the stream with the most channels, for
+subtitles, it is the first subtitle stream. In the case where several streams of
+the same type rate equally, the stream with the lowest index is chosen.

-You can disable some of those defaults by using @code{-vn/-an/-sn} options. For
+You can disable some of those defaults by using the @code{-vn/-an/-sn} options. For
 full manual control, use the @code{-map} option, which disables the defaults just
 described.

@@ -220,7 +222,7 @@ described.

@item -f @var{fmt} (@emph{input/output})
 Force input or output file format. The format is normally auto detected for input
-files and guessed from file extension for output files, so this option is not
+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})
@@ -230,7 +232,8 @@ input file name
 Overwrite output files without asking.

@item -n (@emph{global})
-Do not overwrite output files but exit if file exists.
+Do not overwrite output files, and exit immediately if a specified
+output file already exists.

@item -c[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
@itemx -codec[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream})
@@ -343,25 +346,31 @@ Use fixed quality scale (VBR). The meaning of @var{q} is
 codec-dependent.

@anchor{filter_option}
-@item -filter[:@var{stream_specifier}] @var{filter_graph} (@emph{output,per-stream})
-Create the filter graph specified by @var{filter_graph} and use it to
+@item -filter[:@var{stream_specifier}] @var{filtergraph} (@emph{output,per-stream})
+Create the filtergraph specified by @var{filtergraph} and use it to
 filter the stream.

-@var{filter_graph} is a description of the filter graph to apply to
+@var{filtergraph} is a description of the filtergraph to apply to
 the stream, and must have a single input and a single output of the
-same type of the stream. In the filter graph, the input is associated
+same type of the stream. In the filtergraph, the input is associated
 to the label @code{in}, and the output to the label @code{out}. See
 the ffmpeg-filters manual for more information about the filtergraph
 syntax.

 See the @ref{filter_complex_option,,-filter_complex option} if you
-want to create filter graphs with multiple inputs and/or outputs.
+want to create filtergraphs with multiple inputs and/or outputs.
+
+@item -filter_script[:@var{stream_specifier}] @var{filename} (@emph{output,per-stream})
+This option is similar to @option{-filter}, the only difference is that its
+argument is the name of the file from which a filtergraph description is to be
+read.

@item -pre[:@var{stream_specifier}] @var{preset_name} (@emph{output,per-stream})
 Specify the preset for matching stream(s).

@item -stats (@emph{global})
-Print encoding progress/statistics. On by default.
+Print encoding progress/statistics. It is on by default, to explicitly
+disable it you need to specify @code{-nostats}.

@item -progress @var{url} (@emph{global})
 Send program-friendly progress information to @var{url}.
@@ -459,6 +468,10 @@ form @var{num}:@var{den}, where @var{num} and @var{den} are the
 numerator and denominator of the aspect ratio. For example "4:3",
 "16:9", "1.3333", and "1.7777" are valid argument values.

+If used together with @option{-vcodec copy}, it will affect the aspect ratio
+stored at container level, but not the aspect ratio stored in encoded
+frames, if it exists.
+
@item -vn (@emph{output})
 Disable video recording.

@@ -487,8 +500,8 @@ stream
@item -vlang @var{code}
 Set the ISO 639 language code (3 letters) of the current video stream.

-@item -vf @var{filter_graph} (@emph{output})
-Create the filter graph specified by @var{filter_graph} and use it to
+@item -vf @var{filtergraph} (@emph{output})
+Create the filtergraph specified by @var{filtergraph} and use it to
 filter the stream.

 This is an alias for @code{-filter:v}, see the @ref{filter_option,,-filter option}.
@@ -504,7 +517,7 @@ If the selected pixel format can not be selected, ffmpeg will print a
 warning and select the best pixel format supported by the encoder.
 If @var{pix_fmt} is prefixed by a @code{+}, ffmpeg will exit with an error
 if the requested pixel format can not be selected, and automatic conversions
-inside filter graphs are disabled.
+inside filtergraphs are disabled.
 If @var{pix_fmt} is a single @code{+}, ffmpeg selects the same pixel format
 as the input (or graph output) and automatic conversions are disabled.

@@ -519,10 +532,6 @@ list separated with slashes. Two first values are the beginning and
 end frame numbers, last one is quantizer to use if positive, or quality
 factor if negative.

-@item -deinterlace
-Deinterlace pictures.
-This option is deprecated since the deinterlacing is very low quality.
-Use the yadif filter with @code{-filter:v yadif}.
@item -ilme
 Force interlacing support in encoder (MPEG-2 and MPEG-4 only).
 Use this option if your input file is interlaced and you want
@@ -628,8 +637,8 @@ Set the audio codec. This is an alias for @code{-codec:a}.
 Set the audio sample format. Use @code{-sample_fmts} to get a list
 of supported sample formats.

-@item -af @var{filter_graph} (@emph{output})
-Create the filter graph specified by @var{filter_graph} and use it to
+@item -af @var{filtergraph} (@emph{output})
+Create the filtergraph specified by @var{filtergraph} and use it to
 filter the stream.

 This is an alias for @code{-filter:a}, see the @ref{filter_option,,-filter option}.
@@ -680,6 +689,9 @@ Note that this option will delay the output of all data until the next
 subtitle packet is decoded: it may increase memory consumption and latency a
 lot.

+@item -canvas_size @var{size}
+Set the size of the canvas used to render subtitles.
+
@end table

@section Advanced options
@@ -884,7 +896,7 @@ Newly added values will have to be specified as strings always.
 Each frame is passed with its timestamp from the demuxer to the muxer.
@item 1, cfr
 Frames will be duplicated and dropped to achieve exactly the requested
-constant framerate.
+constant frame rate.
@item 2, vfr
 Frames are passed through with their timestamp or dropped so as to
 prevent 2 frames from having the same timestamp.
@@ -896,6 +908,10 @@ Chooses between 1 and 2 depending on muxer capabilities. This is the
 default method.
@end table

+Note that the timestamps may be further modified by the muxer, after this.
+For example, in the case that the format option @option{avoid_negative_ts}
+is enabled.
+
 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.
@@ -905,6 +921,11 @@ 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.
 -async 1 is a special case where only the start of the audio stream is corrected
 without any later correction.
+
+Note that the timestamps may be further modified by the muxer, after this.
+For example, in the case that the format option @option{avoid_negative_ts}
+is enabled.
+
 This option has been deprecated. Use the @code{aresample} audio filter instead.

@item -copyts
@@ -913,7 +934,8 @@ to sanitize them. In particular, do not remove the initial start time
 offset value.

 Note that, depending on the @option{vsync} option or on specific muxer
-processing, the output timestamps may mismatch with the input
+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 -copytb @var{mode}
@@ -983,10 +1005,10 @@ ffmpeg -i input.mpg -timecode 01:02:03.04 -r 30000/1001 -s ntsc output.mpg

@anchor{filter_complex_option}
@item -filter_complex @var{filtergraph} (@emph{global})
-Define a complex filter graph, i.e. one with arbitrary number of inputs and/or
+Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or
 outputs. For simple graphs -- those with one input and one output of the same
 type -- see the @option{-filter} options. @var{filtergraph} is a description of
-the filter graph, as described in the ``Filtergraph syntax'' section of the
+the filtergraph, as described in the ``Filtergraph syntax'' section of the
 ffmpeg-filters manual.

 Input link labels must refer to input streams using the
@@ -1028,6 +1050,24 @@ To generate 5 seconds of pure red video using lavfi @code{color} source:
@example
 ffmpeg -filter_complex 'color=c=red' -t 5 out.mkv
@end example
+
+@item -lavfi @var{filtergraph} (@emph{global})
+Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or
+outputs. Equivalent to @option{-filter_complex}.
+
+@item -filter_complex_script @var{filename} (@emph{global})
+This option is similar to @option{-filter_complex}, the only difference is that
+its argument is the name of the file from which a complex filtergraph
+description is to be read.
+
+@item -override_ffserver (@emph{global})
+Overrides the input specifications from ffserver. Using this option you can
+map any input stream to ffserver and control many aspects of the encoding from
+ffmpeg. Without this option ffmpeg will transmit to ffserver what is requested by
+ffserver.
+The option is intended for cases where features are needed that cannot be
+specified to ffserver but can be to ffmpeg.
+
@end table

 As a special exception, you can use a bitmap subtitle stream as input: it
@@ -1330,9 +1370,42 @@ ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
@end itemize
@c man end EXAMPLES

+@include config.texi
+@ifset config-all
+@ifset config-avutil
+@include utils.texi
+@end ifset
+@ifset config-avcodec
+@include codecs.texi
+@include bitstream_filters.texi
+@end ifset
+@ifset config-avformat
+@include formats.texi
+@include protocols.texi
+@end ifset
+@ifset config-avdevice
+@include devices.texi
+@end ifset
+@ifset config-swresample
+@include resampler.texi
+@end ifset
+@ifset config-swscale
+@include scaler.texi
+@end ifset
+@ifset config-avfilter
+@include filters.texi
+@end ifset
+@end ifset
+
@chapter See Also

@ifhtml
+@ifset config-all
+@url{ffmpeg.html,ffmpeg}
+@end ifset
+@ifset config-not-all
+@url{ffmpeg-all.html,ffmpeg-all},
+@end ifset
@url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@@ -1346,6 +1419,12 @@ ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
@end ifhtml

@ifnothtml
+@ifset config-all
+ffmpeg(1),
+@end ifset
+@ifset config-not-all
+ffmpeg-all(1),
+@end ifset
 ffplay(1), ffprobe(1), ffserver(1),
 ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
 ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
--- a/doc/ffplay.texi
+++ b/doc/ffplay.texi
@@ -73,17 +73,23 @@ Default value is "video", if video is not present or cannot be played
 You can interactively cycle through the available show modes by
 pressing the key @key{w}.

-@item -vf @var{filter_graph}
-Create the filter graph specified by @var{filter_graph} and use it to
+@item -vf @var{filtergraph}
+Create the filtergraph specified by @var{filtergraph} and use it to
 filter the video stream.

-@var{filter_graph} is a description of the filter graph to apply to
+@var{filtergraph} is a description of the filtergraph to apply to
 the stream, and must have a single video input and a single video
-output. In the filter graph, the input is associated to the label
+output. In the filtergraph, the input is associated to the label
@code{in}, and the output to the label @code{out}. See the
 ffmpeg-filters manual for more information about the filtergraph
 syntax.

+@item -af @var{filtergraph}
+@var{filtergraph} is a description of the filtergraph to apply to
+the input audio.
+Use the option "-filters" to show all the available filters (including
+sources and sinks).
+
@item -i @var{input_file}
 Read @var{input_file}.
@end table
@@ -93,9 +99,13 @@ Read @var{input_file}.
@item -pix_fmt @var{format}
 Set pixel format.
 This option has been deprecated in favor of private options, try -pixel_format.
+
@item -stats
-Show the stream duration, the codec parameters, the current position in
-the stream and the audio/video synchronisation drift.
+Print several playback statistics, in particular show the stream
+duration, the codec parameters, the current position in the stream and
+the audio/video synchronisation drift. It is on by default, to
+explicitly disable it you need to specify @code{-nostats}.
+
@item -bug
 Work around bugs.
@item -fast
@@ -191,9 +201,42 @@ Seek to percentage in file corresponding to fraction of width.

@c man end

+@include config.texi
+@ifset config-all
+@ifset config-avutil
+@include utils.texi
+@end ifset
+@ifset config-avcodec
+@include codecs.texi
+@include bitstream_filters.texi
+@end ifset
+@ifset config-avformat
+@include formats.texi
+@include protocols.texi
+@end ifset
+@ifset config-avdevice
+@include devices.texi
+@end ifset
+@ifset config-swresample
+@include resampler.texi
+@end ifset
+@ifset config-swscale
+@include scaler.texi
+@end ifset
+@ifset config-avfilter
+@include filters.texi
+@end ifset
+@end ifset
+
@chapter See Also

@ifhtml
+@ifset config-all
+@url{ffplay.html,ffplay},
+@end ifset
+@ifset config-not-all
+@url{ffplay-all.html,ffmpeg-all},
+@end ifset
@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@@ -207,6 +250,12 @@ Seek to percentage in file corresponding to fraction of width.
@end ifhtml

@ifnothtml
+@ifset config-all
+ffplay(1),
+@end ifset
+@ifset config-not-all
+ffplay-all(1),
+@end ifset
 ffmpeg(1), ffprobe(1), ffserver(1),
 ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
 ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
--- a/doc/ffprobe.texi
+++ b/doc/ffprobe.texi
@@ -209,6 +209,11 @@ multimedia stream.
 Each media stream information is printed within a dedicated section
 with name "STREAM".

+@item -show_chapters
+Show information about chapters stored in the format.
+
+Each chapter is printed within a dedicated section with name "CHAPTER".
+
@item -count_frames
 Count the number of frames per stream and report it in the
 corresponding stream section.
@@ -487,10 +492,43 @@ DV, GXF and AVI timecodes are available in format metadata
@end itemize
@c man end TIMECODE

+@include config.texi
+@ifset config-all
+@ifset config-avutil
+@include utils.texi
+@end ifset
+@ifset config-avcodec
+@include codecs.texi
+@include bitstream_filters.texi
+@end ifset
+@ifset config-avformat
+@include formats.texi
+@include protocols.texi
+@end ifset
+@ifset config-avdevice
+@include devices.texi
+@end ifset
+@ifset config-swresample
+@include resampler.texi
+@end ifset
+@ifset config-swscale
+@include scaler.texi
+@end ifset
+@ifset config-avfilter
+@include filters.texi
+@end ifset
+@end ifset
+
@chapter See Also

@ifhtml
-@url{ffplay.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
+@ifset config-all
+@url{ffprobe.html,ffprobe},
+@end ifset
+@ifset config-not-all
+@url{ffprobe-all.html,ffprobe-all},
+@end ifset
+@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@@ -503,6 +541,12 @@ DV, GXF and AVI timecodes are available in format metadata
@end ifhtml

@ifnothtml
+@ifset config-all
+ffprobe(1),
+@end ifset
+@ifset config-not-all
+ffprobe-all(1),
+@end ifset
 ffmpeg(1), ffplay(1), ffserver(1),
 ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
 ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
--- a/doc/ffprobe.xsd
+++ b/doc/ffprobe.xsd
@@ -11,6 +11,7 @@
            <xsd:element name="packets"  type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" />
            <xsd:element name="frames"   type="ffprobe:framesType"  minOccurs="0" maxOccurs="1" />
            <xsd:element name="streams"  type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
+            <xsd:element name="chapters" type="ffprobe:chaptersType" minOccurs="0" maxOccurs="1" />
            <xsd:element name="format"   type="ffprobe:formatType"  minOccurs="0" maxOccurs="1" />
            <xsd:element name="error"    type="ffprobe:errorType"   minOccurs="0" maxOccurs="1" />
            <xsd:element name="program_version"  type="ffprobe:programVersionType"  minOccurs="0" maxOccurs="1" />
@@ -78,7 +79,6 @@
      <xsd:attribute name="interlaced_frame"       type="xsd:int"   />
      <xsd:attribute name="top_field_first"        type="xsd:int"   />
      <xsd:attribute name="repeat_pict"            type="xsd:int"   />
-      <xsd:attribute name="reference"              type="xsd:int"   />
    </xsd:complexType>

    <xsd:complexType name="streamsType">
@@ -182,6 +182,25 @@
      <xsd:attribute name="configuration"    type="xsd:string" use="required"/>
    </xsd:complexType>

+    <xsd:complexType name="chaptersType">
+      <xsd:sequence>
+        <xsd:element name="chapter" type="ffprobe:chapterType" minOccurs="0" maxOccurs="unbounded"/>
+      </xsd:sequence>
+    </xsd:complexType>
+
+    <xsd:complexType name="chapterType">
+      <xsd:sequence>
+        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
+      </xsd:sequence>
+
+      <xsd:attribute name="id"         type="xsd:int" use="required"/>
+      <xsd:attribute name="time_base"  type="xsd:string" use="required"/>
+      <xsd:attribute name="start"      type="xsd:int" use="required"/>
+      <xsd:attribute name="start_time" type="xsd:float"/>
+      <xsd:attribute name="end"        type="xsd:int" use="required"/>
+      <xsd:attribute name="end_time"   type="xsd:float" use="required"/>
+    </xsd:complexType>
+
    <xsd:complexType name="libraryVersionType">
      <xsd:attribute name="name"        type="xsd:string" use="required"/>
      <xsd:attribute name="major"       type="xsd:int"    use="required"/>
--- a/doc/ffserver.texi
+++ b/doc/ffserver.texi
@@ -246,10 +246,43 @@ messages to stdout.
@end table
@c man end

+@include config.texi
+@ifset config-all
+@ifset config-avutil
+@include utils.texi
+@end ifset
+@ifset config-avcodec
+@include codecs.texi
+@include bitstream_filters.texi
+@end ifset
+@ifset config-avformat
+@include formats.texi
+@include protocols.texi
+@end ifset
+@ifset config-avdevice
+@include devices.texi
+@end ifset
+@ifset config-swresample
+@include resampler.texi
+@end ifset
+@ifset config-swscale
+@include scaler.texi
+@end ifset
+@ifset config-avfilter
+@include filters.texi
+@end ifset
+@end ifset
+
@chapter See Also

@ifhtml
-The @file{doc/ffserver.conf} example,
+@ifset config-all
+@url{ffserver.html,ffserver},
+@end ifset
+@ifset config-not-all
+@url{ffserver-all.html,ffserver-all},
+@end ifset
+the @file{doc/ffserver.conf} example,
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@@ -263,7 +296,13 @@ The @file{doc/ffserver.conf} example,
@end ifhtml

@ifnothtml
-The @file{doc/ffserver.conf} example, ffmpeg(1), ffplay(1), ffprobe(1),
+@ifset config-all
+ffserver(1),
+@end ifset
+@ifset config-not-all
+ffserver-all(1),
+@end ifset
+the @file{doc/ffserver.conf} example, ffmpeg(1), ffplay(1), ffprobe(1),
 ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
 ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
 ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
--- a/doc/filter_design.txt
+++ b/doc/filter_design.txt
@@ -29,6 +29,11 @@ Format negotiation
  same format amongst a supported list, all it has to do is use a reference
  to the same list of formats.

+  query_formats can leave some formats unset and return AVERROR(EAGAIN) to
+  cause the negotiation mechanism to try again later. That can be used by
+  filters with complex requirements to use the format negotiated on one link
+  to set the formats supported on another.
+

 Buffer references ownership and permissions
 ===========================================
@@ -199,7 +204,7 @@ Frame scheduling
    filter; these buffered frames must be flushed immediately if a new input
    produces new output.

-    (Example: framerate-doubling filter: filter_frame must (1) flush the
+    (Example: frame rate-doubling filter: filter_frame must (1) flush the
    second copy of the previous frame, if it is still there, (2) push the
    first copy of the incoming frame, (3) keep the second copy for later.)

--- a/doc/filters.texi
+++ b/doc/filters.texi
--- a/doc/formats.texi
+++ b/doc/formats.texi
@@ -0,0 +1,155 @@
+@chapter Format Options
+@c man begin FORMAT OPTIONS
+
+The libavformat library provides some generic global options, which
+can be set on all the muxers and demuxers. In addition each muxer or
+demuxer may support so-called private options, which are specific for
+that component.
+
+Options may be set by specifying -@var{option} @var{value} in the
+FFmpeg tools, or by setting the value explicitly in the
+@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
+for programmatic use.
+
+The list of supported options follows:
+
+@table @option
+@item avioflags @var{flags} (@emph{input/output})
+Possible values:
+@table @samp
+@item direct
+Reduce buffering.
+@end table
+
+@item probesize @var{integer} (@emph{input})
+Set probing size in bytes, i.e. the size of the data to analyze to get
+stream information. A higher value will allow to detect more
+information in case it is dispersed into the stream, but will increase
+latency. Must be an integer not lesser than 32. It is 5000000 by default.
+
+@item packetsize @var{integer} (@emph{output})
+Set packet size.
+
+@item fflags @var{flags} (@emph{input/output})
+Set format flags.
+
+Possible values:
+@table @samp
+@item ignidx
+Ignore index.
+@item genpts
+Generate PTS.
+@item nofillin
+Do not fill in missing values that can be exactly calculated.
+@item noparse
+Disable AVParsers, this needs @code{+nofillin} too.
+@item igndts
+Ignore DTS.
+@item discardcorrupt
+Discard corrupted frames.
+@item sortdts
+Try to interleave output packets by DTS.
+@item keepside
+Do not merge side data.
+@item latm
+Enable RTP MP4A-LATM payload.
+@item nobuffer
+Reduce the latency introduced by optional buffering
+@end table
+
+@item seek2any @var{integer} (@emph{input})
+Forces seeking to enable seek to any mode if set to 1. Default is 0.
+
+@item analyzeduration @var{integer} (@emph{input})
+Specify how many microseconds are analyzed to probe the input. A
+higher value will allow to detect more accurate information, but will
+increase latency. It defaults to 5,000,000 microseconds = 5 seconds.
+
+@item cryptokey @var{hexadecimal string} (@emph{input})
+Set decryption key.
+
+@item indexmem @var{integer} (@emph{input})
+Set max memory used for timestamp index (per stream).
+
+@item rtbufsize @var{integer} (@emph{input})
+Set max memory used for buffering real-time frames.
+
+@item fdebug @var{flags} (@emph{input/output})
+Print specific debug info.
+
+Possible values:
+@table @samp
+@item ts
+@end table
+
+@item max_delay @var{integer} (@emph{input/output})
+Set maximum muxing or demuxing delay in microseconds.
+
+@item fpsprobesize @var{integer} (@emph{input})
+Set number of frames used to probe fps.
+
+@item audio_preload @var{integer} (@emph{output})
+Set microseconds by which audio packets should be interleaved earlier.
+
+@item chunk_duration @var{integer} (@emph{output})
+Set microseconds for each chunk.
+
+@item chunk_size @var{integer} (@emph{output})
+Set size in bytes for each chunk.
+
+@item err_detect, f_err_detect @var{flags} (@emph{input})
+Set error detection flags. @code{f_err_detect} is deprecated and
+should be used only via the @command{ffmpeg} tool.
+
+Possible values:
+@table @samp
+@item crccheck
+Verify embedded CRCs.
+@item bitstream
+Detect bitstream specification deviations.
+@item buffer
+Detect improper bitstream length.
+@item explode
+Abort decoding on minor error detection.
+@item careful
+Consider things that violate the spec and have not been seen in the
+wild as errors.
+@item compliant
+Consider all spec non compliancies as errors.
+@item aggressive
+Consider things that a sane encoder should not do as an error.
+@end table
+
+@item use_wallclock_as_timestamps @var{integer} (@emph{input})
+Use wallclock as timestamps.
+
+@item avoid_negative_ts @var{integer} (@emph{output})
+Shift timestamps to make them non-negative. A value of 1 enables shifting,
+a value of 0 disables it, the default value of -1 enables shifting
+when required by the target format.
+
+When shifting is enabled, all output timestamps are shifted by the
+same amount. Audio, video, and subtitles desynching and relative
+timestamp differences are preserved compared to how they would have
+been without shifting.
+
+Also note that this affects only leading negative timestamps, and not
+non-monotonic negative timestamps.
+
+@item skip_initial_bytes @var{integer} (@emph{input})
+Set number initial bytes to skip. Default is 0.
+
+@item correct_ts_overflow @var{integer} (@emph{input})
+Correct single timestamp overflows if set to 1. Default is 1.
+
+@item flush_packets @var{integer} (@emph{output})
+Flush the underlying I/O stream after each packet. Default 1 enables it, and
+has the effect of reducing the latency; 0 disables it and may slightly
+increase performance in some cases.
+@end table
+
+@c man end FORMAT OPTIONS
+
+@include demuxers.texi
+@include muxers.texi
+@include metadata.texi
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -24,7 +24,7 @@ instructions.  To enable using OpenJPEG in FFmpeg, pass @code{--enable-libopenjp
@file{./configure}.


-@section OpenCORE and VisualOn libraries
+@section OpenCORE, VisualOn, and Fraunhofer libraries

 Spun off Google Android sources, OpenCore, VisualOn and Fraunhofer
 libraries provide encoders for a number of audio codecs.
@@ -32,9 +32,14 @@ libraries provide encoders for a number of audio codecs.
@float NOTE
 OpenCORE and VisualOn libraries are under the Apache License 2.0
 (see @url{http://www.apache.org/licenses/LICENSE-2.0} for details), which is
-incompatible with the LGPL version 2.1 and GPL version 2. You have to
+incompatible to the LGPL version 2.1 and GPL version 2. You have to
 upgrade FFmpeg's license to LGPL version 3 (or if you have enabled
-GPL components, GPL version 3) to use it.
+GPL components, GPL version 3) by passing @code{--enable-version3} to configure in
+order to use it.
+
+The Fraunhofer AAC library is licensed under a license incompatible to the GPL
+and is not known to be compatible to the LGPL. Therefore, you have to pass
+@code{--enable-nonfree} to configure to use it.
@end float

@subsection OpenCORE AMR
@@ -95,6 +100,14 @@ Go to @url{http://www.webmproject.org/} and follow the instructions for
 installing the library. Then pass @code{--enable-libvpx} to configure to
 enable it.

+@section libwavpack
+
+FFmpeg can make use of the libwavpack library for WavPack encoding.
+
+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 x264

 FFmpeg can make use of the x264 library for H.264 encoding.
@@ -147,6 +160,8 @@ library:
    @tab Multimedia format used in game Heart Of Darkness.
@item Apple HTTP Live Streaming @tab   @tab X
@item Artworx Data Format       @tab   @tab X
+@item ADP                       @tab   @tab X
+    @tab Audio format used on the Nintendo Gamecube.
@item AFC                       @tab   @tab X
    @tab Audio format used on the Nintendo Gamecube.
@item ASF                       @tab X @tab X
@@ -322,7 +337,7 @@ library:
@item raw Shorten               @tab   @tab X
@item raw TAK                   @tab   @tab X
@item raw TrueHD                @tab X @tab X
-@item raw VC-1                  @tab   @tab X
+@item raw VC-1                  @tab X @tab X
@item raw PCM A-law             @tab X @tab X
@item raw PCM mu-law            @tab X @tab X
@item raw PCM signed 8 bit      @tab X @tab X
@@ -348,11 +363,13 @@ library:
    @tab File format used by RED Digital cameras, contains JPEG 2000 frames and PCM audio.
@item RealMedia                 @tab X @tab X
@item Redirector                @tab   @tab X
+@item RedSpark                  @tab   @tab X
@item Renderware TeXture Dictionary @tab   @tab X
@item RL2                       @tab   @tab X
    @tab Audio and video format used in some games by Entertainment Software Partners.
@item RPL/ARMovie               @tab   @tab X
@item Lego Mindstorms RSO       @tab X @tab X
+@item RSD                       @tab   @tab X
@item RTMP                      @tab X @tab X
    @tab Output is performed by publishing stream to RTMP server
@item RTP                       @tab X @tab X
@@ -418,7 +435,6 @@ following image formats are supported:
@item .Y.U.V       @tab X @tab X
    @tab one raw file per component
@item animated GIF @tab X @tab X
-    @tab Only uncompressed GIFs are generated.
@item BMP          @tab X @tab X
    @tab Microsoft BMP image
@item PIX          @tab   @tab X
@@ -485,6 +501,7 @@ following image formats are supported:
@item AMV Video              @tab  X  @tab  X
    @tab Used in Chinese MP3 players.
@item ANSI/ASCII art         @tab     @tab  X
+@item Apple Intermediate Codec @tab     @tab  X
@item Apple MJPEG-B          @tab     @tab  X
@item Apple ProRes           @tab  X  @tab  X
@item Apple QuickDraw        @tab     @tab  X
@@ -567,6 +584,8 @@ following image formats are supported:
    @tab Sorenson H.263 used in Flash
@item Forward Uncompressed   @tab     @tab  X
@item Fraps                  @tab     @tab  X
+@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
@@ -769,9 +788,11 @@ following image formats are supported:
    @tab Used in some Sega Saturn console games.
@item ADPCM IMA Duck DK4     @tab     @tab  X
    @tab Used in some Sega Saturn console games.
+@item ADPCM IMA Radical      @tab     @tab  X
@item ADPCM Microsoft        @tab  X  @tab  X
@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 QT IMA           @tab  X  @tab  X
@item ADPCM SEGA CRI ADX     @tab  X  @tab  X
@@ -831,7 +852,6 @@ following image formats are supported:
@item MLP (Meridian Lossless Packing)  @tab     @tab  X
    @tab Used in DVD-Audio discs.
@item Monkey's Audio         @tab     @tab  X
-    @tab Only versions 3.97-3.99 are supported.
@item MP1 (MPEG audio layer 1)  @tab     @tab IX
@item MP2 (MPEG audio layer 2)  @tab IX  @tab IX
    @tab libtwolame can be used alternatively for encoding.
@@ -886,7 +906,7 @@ following image formats are supported:
@item Sierra VMD audio       @tab     @tab  X
    @tab Used in Sierra VMD files.
@item Smacker audio          @tab     @tab  X
-@item SMPTE 302M AES3 audio  @tab     @tab  X
+@item SMPTE 302M AES3 audio  @tab  X  @tab  X
@item Sonic                  @tab  X  @tab  X
    @tab experimental codec
@item Sonic lossless         @tab  X  @tab  X
@@ -894,7 +914,7 @@ following image formats are supported:
@item Speex                  @tab  E  @tab  E
    @tab supported through external library libspeex
@item TAK (Tom's lossless Audio Kompressor)  @tab     @tab  X
-@item True Audio (TTA)       @tab     @tab  X
+@item True Audio (TTA)       @tab  X  @tab  X
@item TrueHD                 @tab     @tab  X
    @tab Used in HD-DVD and Blu-Ray discs.
@item TwinVQ (VQF flavor)    @tab     @tab  X
@@ -902,7 +922,8 @@ following image formats are supported:
    @tab Used in LucasArts SMUSH animations.
@item Vorbis                 @tab  E  @tab  X
    @tab A native but very primitive encoder exists.
-@item WavPack                @tab     @tab  X
+@item WavPack                @tab  E  @tab  X
+    @tab supported through external library libwavpack
@item Westwood Audio (SND1)  @tab     @tab  X
@item Windows Media Audio 1  @tab  X  @tab  X
@item Windows Media Audio 2  @tab  X  @tab  X
@@ -941,7 +962,7 @@ performance on systems without hardware floating point support).
@item TED Talks captions @tab @tab X @tab   @tab X
@item VobSub (IDX+SUB) @tab   @tab X @tab   @tab X
@item VPlayer          @tab   @tab X @tab   @tab X
-@item WebVTT           @tab   @tab X @tab   @tab X
+@item WebVTT           @tab X @tab X @tab   @tab X
@item XSUB             @tab   @tab   @tab X @tab X
@end multitable

@@ -994,7 +1015,7 @@ performance on systems without hardware floating point support).
@item OSS               @tab X      @tab X
@item Pulseaudio        @tab X      @tab
@item SDL               @tab        @tab X
-@item Video4Linux2      @tab X      @tab
+@item Video4Linux2      @tab X      @tab X
@item VfW capture       @tab X      @tab
@item X11 grabbing      @tab X      @tab
@end multitable
--- a/doc/indevs.texi
+++ b/doc/indevs.texi
@@ -86,7 +86,7 @@ fail to open.
 Set the video size in the captured video.

@item framerate
-Set the framerate in the captured video.
+Set the frame rate in the captured video.

@item sample_rate
 Set the sample rate (in Hz) of the captured audio.
@@ -590,8 +590,8 @@ Video4Linux2 input video device.
 "v4l2" can be used as alias for "video4linux2".

 If FFmpeg is built with v4l-utils support (by using the
-@code{--enable-libv4l2} configure option), the device will always rely
-on libv4l2.
+@code{--enable-libv4l2} configure option), it is possible to use it with the
+@code{-use_libv4l2} input device option.

 The name of the device to grab is a file device node, usually Linux
 systems tend to automatically create such nodes when the device
@@ -600,7 +600,7 @@ kind @file{/dev/video@var{N}}, where @var{N} is a number associated to
 the device.

 Video4Linux2 devices usually support a limited set of
-@var{width}x@var{height} sizes and framerates. You can check which are
+@var{width}x@var{height} sizes and frame rates. You can check which are
 supported using @command{-list_formats all} for Video4Linux2 devices.
 Some devices, like TV cards, support one or more standards. It is possible
 to list all the supported standards using @command{-list_standards all}.
@@ -623,7 +623,7 @@ ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0

@item
 Grab and record the input of a video4linux2 device, leave the
-framerate and size as previously set:
+frame rate and size as previously set:
@example
 ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg
@end example
@@ -640,7 +640,8 @@ list of the supported standards, use the @option{list_standards}
 option.

@item channel
-Set the input channel number. Default to 0.
+Set the input channel number. Default to -1, which means using the
+previously selected channel.

@item video_size
 Set the video frame size. The argument must be a string in the form
@@ -655,7 +656,7 @@ This option allows to select the input format, when several are
 available.

@item framerate
-Set the preferred video framerate.
+Set the preferred video frame rate.

@item list_formats
 List available formats (supported pixel formats, codecs, and frame
@@ -771,7 +772,7 @@ ffmpeg -f x11grab -follow_mouse 100 -r 25 -s cif -i :0.0 out.mpg

@item framerate
 Set the grabbing frame rate. Default value is @code{ntsc},
-corresponding to a framerate of @code{30000/1001}.
+corresponding to a frame rate of @code{30000/1001}.

@item show_region
 Show grabbed region on screen.
--- a/doc/issue_tracker.txt
+++ b/doc/issue_tracker.txt
@@ -24,7 +24,7 @@ a mail for every change to every issue.
 The subscription URL for the ffmpeg-trac list is:
 http(s)://ffmpeg.org/mailman/listinfo/ffmpeg-trac
 The URL of the webinterface of the tracker is:
-http(s)://ffmpeg.org/trac/ffmpeg
+http(s)://trac.ffmpeg.org

 Type:
 -----
--- a/doc/metadata.texi
+++ b/doc/metadata.texi
@@ -65,4 +65,20 @@ title=chapter \#1
 title=multi\
 line
@end example
+
+By using the ffmetadata muxer and demuxer it is possible to extract
+metadata from an input file to an ffmetadata file, and then transcode
+the file into an output file with the edited ffmetadata file.
+
+Extracting an ffmetadata file with @file{ffmpeg} goes as follows:
+@example
+ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE
+@end example
+
+Reinserting edited metadata information from the FFMETADATAFILE file can
+be done as:
+@example
+ffmpeg -i INPUT -i FFMETADATAFILE -map_metadata 1 -codec copy OUTPUT
+@end example
+
@c man end METADATA
--- a/doc/mips.txt
+++ b/doc/mips.txt
@@ -48,6 +48,8 @@ Files that have MIPS copyright notice in them:
      float_dsp_mips.c
      libm_mips.h
 * libavcodec/mips/
+      aaccoder_mips.c
+      aacpsy_mips.h
      ac3dsp_mips.c
      acelp_filters_mips.c
      acelp_vectors_mips.c
@@ -63,5 +65,6 @@ Files that have MIPS copyright notice in them:
      fft_table.h
      fft_init_table.c
      fmtconvert_mips.c
+      iirfilter_mips.c
      mpegaudiodsp_mips_fixed.c
      mpegaudiodsp_mips_float.c
--- a/doc/multithreading.txt
+++ b/doc/multithreading.txt
@@ -57,6 +57,11 @@ which re-allocates them for other threads.
 Add CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little
 speed gain at this point but it should work.

+If there are inter-frame dependencies, so the codec calls
+ff_thread_report/await_progress(), set AVCodecInternal.allocate_progress. The
+frames must then be freed with ff_thread_release_buffer().
+Otherwise leave it at zero and decode directly into the user-supplied frames.
+
 Call ff_thread_report_progress() after some part of the current picture has decoded.
 A good place to put this is where draw_horiz_band() is called - add this if it isn't
 called anywhere, as it's useful too and the implementation is trivial when you're
--- a/doc/muxers.texi
+++ b/doc/muxers.texi
@@ -18,6 +18,23 @@ enabled muxers.

 A description of some of the currently available muxers follows.

+@anchor{aiff}
+@section aiff
+
+Audio Interchange File Format muxer.
+
+It accepts the following options:
+
+@table @option
+@item write_id3v2
+Enable ID3v2 tags writing when set to 1. Default is 0 (disabled).
+
+@item id3v2_version
+Select ID3v2 version to write. Currently only version 3 and 4 (aka.
+ID3v2.3 and ID3v2.4) are supported. The default is version 4.
+
+@end table
+
@anchor{crc}
@section crc

@@ -245,9 +262,11 @@ ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg
 Start the sequence from @var{number}. Default value is 1. Must be a
 positive number.

-@item updatefirst 1|0
-If set to 1, update the first written image file again and
-again. Default value is 0.
+@item -update @var{number}
+If @var{number} is nonzero, the filename will always be interpreted as just a
+filename, not a pattern, and this file will be continuously overwritten with new
+images.
+
@end table

 The image muxer supports the .Y.U.V image file format. This format is
@@ -256,6 +275,90 @@ each of the YUV420P components. To read or write this image file format,
 specify the name of the '.Y' file. The muxer will automatically open the
 '.U' and '.V' files as required.

+@section matroska
+
+Matroska container muxer.
+
+This muxer implements the matroska and webm container specs.
+
+The recognized metadata settings in this muxer are:
+
+@table @option
+
+@item title=@var{title name}
+Name provided to a single track
+@end table
+
+@table @option
+
+@item language=@var{language name}
+Specifies the language of the track in the Matroska languages form
+@end table
+
+@table @option
+
+@item stereo_mode=@var{mode}
+Stereo 3D video layout of two views in a single video track
+@table @option
+@item mono
+video is not stereo
+@item left_right
+Both views are arranged side by side, Left-eye view is on the left
+@item bottom_top
+Both views are arranged in top-bottom orientation, Left-eye view is at bottom
+@item top_bottom
+Both views are arranged in top-bottom orientation, Left-eye view is on top
+@item checkerboard_rl
+Each view is arranged in a checkerboard interleaved pattern, Left-eye view being first
+@item checkerboard_lr
+Each view is arranged in a checkerboard interleaved pattern, Right-eye view being first
+@item row_interleaved_rl
+Each view is constituted by a row based interleaving, Right-eye view is first row
+@item row_interleaved_lr
+Each view is constituted by a row based interleaving, Left-eye view is first row
+@item col_interleaved_rl
+Both views are arranged in a column based interleaving manner, Right-eye view is first column
+@item col_interleaved_lr
+Both views are arranged in a column based interleaving manner, Left-eye view is first column
+@item anaglyph_cyan_red
+All frames are in anaglyph format viewable through red-cyan filters
+@item right_left
+Both views are arranged side by side, Right-eye view is on the left
+@item anaglyph_green_magenta
+All frames are in anaglyph format viewable through green-magenta filters
+@item block_lr
+Both eyes laced in one Block, Left-eye view is first
+@item block_rl
+Both eyes laced in one Block, Right-eye view is first
+@end table
+@end table
+
+For example a 3D WebM clip can be created using the following command line:
+@example
+ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm
+@end example
+
+This muxer supports the following options:
+
+@table @option
+
+@item reserve_index_space
+By default, this muxer writes the index for seeking (called cues in Matroska
+terms) at the end of the file, because it cannot know in advance how much space
+to leave for the index at the beginning of the file. However for some use cases
+-- e.g.  streaming where seeking is possible but slow -- it is useful to put the
+index at the beginning of the file.
+
+If this option is set to a non-zero value, the muxer will reserve a given amount
+of space in the file header and then try to write the cues there when the muxing
+finishes. If the available space does not suffice, muxing will fail. A safe size
+for most use cases should be about 50kB per hour of video.
+
+Note that cues are only written if the output is seekable and this option will
+have no effect if it is not.
+
+@end table
+
@anchor{md5}
@section md5

@@ -361,6 +464,42 @@ 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

+@section mp3
+
+The MP3 muxer writes a raw MP3 stream with an ID3v2 header at the beginning and
+optionally an ID3v1 tag at the end. ID3v2.3 and ID3v2.4 are supported, the
+@code{id3v2_version} option controls which one is used. The legacy ID3v1 tag is
+not written by default, but may be enabled with the @code{write_id3v1} option.
+
+For seekable output the muxer also writes a Xing frame at the beginning, which
+contains the number of frames in the file. It is useful for computing duration
+of VBR files.
+
+The muxer supports writing ID3v2 attached pictures (APIC frames). The pictures
+are supplied to the muxer in form of a video stream with a single packet. There
+can be any number of those streams, each will correspond to a single APIC frame.
+The stream metadata tags @var{title} and @var{comment} map to APIC
+@var{description} and @var{picture type} respectively. See
+@url{http://id3.org/id3v2.4.0-frames} for allowed picture types.
+
+Note that the APIC frames must be written at the beginning, so the muxer will
+buffer the audio frames until it gets all the pictures. It is therefore advised
+to provide the pictures as soon as possible to avoid excessive buffering.
+
+Examples:
+
+Write an mp3 with an ID3v2.3 header and an ID3v1 footer:
+@example
+ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3
+@end example
+
+To attach a picture to an mp3 file select both the audio and the picture stream
+with @code{map}:
+@example
+ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1
+-metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3
+@end example
+
@section mpegts

 MPEG transport stream muxer.
@@ -424,69 +563,21 @@ Alternatively you can write the command as:
 ffmpeg -benchmark -i INPUT -f null -
@end example

-@section matroska
+@section ogg

-Matroska container muxer.
-
-This muxer implements the matroska and webm container specs.
-
-The recognized metadata settings in this muxer are:
+Ogg container muxer.

@table @option
-
-@item title=@var{title name}
-Name provided to a single track
+@item -page_duration @var{duration}
+Preferred page duration, in microseconds. The muxer will attempt to create
+pages that are approximately @var{duration} microseconds long. This allows the
+user to compromise between seek granularity and container overhead. The default
+is 1 second. A value of 0 will fill all segments, making pages as large as
+possible. A value of 1 will effectively use 1 packet-per-page in most
+situations, giving a small seek granularity at the cost of additional container
+overhead.
@end table

-@table @option
-
-@item language=@var{language name}
-Specifies the language of the track in the Matroska languages form
-@end table
-
-@table @option
-
-@item stereo_mode=@var{mode}
-Stereo 3D video layout of two views in a single video track
-@table @option
-@item mono
-video is not stereo
-@item left_right
-Both views are arranged side by side, Left-eye view is on the left
-@item bottom_top
-Both views are arranged in top-bottom orientation, Left-eye view is at bottom
-@item top_bottom
-Both views are arranged in top-bottom orientation, Left-eye view is on top
-@item checkerboard_rl
-Each view is arranged in a checkerboard interleaved pattern, Left-eye view being first
-@item checkerboard_lr
-Each view is arranged in a checkerboard interleaved pattern, Right-eye view being first
-@item row_interleaved_rl
-Each view is constituted by a row based interleaving, Right-eye view is first row
-@item row_interleaved_lr
-Each view is constituted by a row based interleaving, Left-eye view is first row
-@item col_interleaved_rl
-Both views are arranged in a column based interleaving manner, Right-eye view is first column
-@item col_interleaved_lr
-Both views are arranged in a column based interleaving manner, Left-eye view is first column
-@item anaglyph_cyan_red
-All frames are in anaglyph format viewable through red-cyan filters
-@item right_left
-Both views are arranged side by side, Right-eye view is on the left
-@item anaglyph_green_magenta
-All frames are in anaglyph format viewable through green-magenta filters
-@item block_lr
-Both eyes laced in one Block, Left-eye view is first
-@item block_rl
-Both eyes laced in one Block, Right-eye view is first
-@end table
-@end table
-
-For example a 3D WebM clip can be created using the following command line:
-@example
-ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm
-@end example
-
@section segment, stream_segment, ssegment

 Basic stream segmenter.
@@ -523,7 +614,7 @@ Set the reference stream, as specified by the string @var{specifier}.
 If @var{specifier} is set to @code{auto}, the reference is choosen
 automatically. Otherwise it must be a stream specifier (see the ``Stream
 specifiers'' chapter in the ffmpeg manual) which specifies the
-reference stream. The default value is ``auto''.
+reference stream. The default value is @code{auto}.

@item segment_format @var{format}
 Override the inner container format, by default it is guessed by the filename
@@ -537,7 +628,7 @@ listfile is generated.
 Set flags affecting the segment list generation.

 It currently supports the following flags:
-@table @var
+@table @samp
@item cache
 Allow caching (only affects M3U8 list files).

@@ -545,18 +636,18 @@ Allow caching (only affects M3U8 list files).
 Allow live-friendly file generation.
@end table

-Default value is @code{cache}.
+Default value is @code{samp}.

@item segment_list_size @var{size}
 Update the list file so that it contains at most the last @var{size}
 segments. If 0 the list file will contain all the segments. Default
 value is 0.

-@item segment_list type @var{type}
+@item segment_list_type @var{type}
 Specify the format for the segment list file.

 The following values are recognized:
-@table @option
+@table @samp
@item flat
 Generate a flat list for the created segments, one segment per line.

@@ -577,7 +668,7 @@ the segment start and end time expressed in seconds.
 A list file with the suffix @code{".csv"} or @code{".ext"} will
 auto-select this format.

-@code{ext} is deprecated in favor or @code{csv}.
+@samp{ext} is deprecated in favor or @samp{csv}.

@item ffconcat
 Generate an ffconcat file for the created segments. The resulting file
@@ -671,9 +762,9 @@ ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_ti
@end example

@item
-As the example above, but use the @code{ffmpeg} @var{force_key_frames}
+As the example above, but use the @command{ffmpeg} @option{force_key_frames}
 option to force key frames in the input at the specified location, together
-with the segment option @var{segment_time_delta} to account for
+with the segment option @option{segment_time_delta} to account for
 possible roundings operated when setting key frame times.
@example
 ffmpeg -i in.mkv -force_key_frames 1,2,3,5,8,13,21 -codec:v mpeg4 -codec:a pcm_s16le -map 0 \
@@ -684,7 +775,7 @@ required.

@item
 Segment the input file by splitting the input file according to the
-frame numbers sequence specified with the @var{segment_frames} option:
+frame numbers sequence specified with the @option{segment_frames} option:
@example
 ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_frames 100,200,300,500,800 out%03d.nut
@end example
@@ -705,57 +796,6 @@ ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \
@end example
@end itemize

-@section mp3
-
-The MP3 muxer writes a raw MP3 stream with an ID3v2 header at the beginning and
-optionally an ID3v1 tag at the end. ID3v2.3 and ID3v2.4 are supported, the
-@code{id3v2_version} option controls which one is used. The legacy ID3v1 tag is
-not written by default, but may be enabled with the @code{write_id3v1} option.
-
-For seekable output the muxer also writes a Xing frame at the beginning, which
-contains the number of frames in the file. It is useful for computing duration
-of VBR files.
-
-The muxer supports writing ID3v2 attached pictures (APIC frames). The pictures
-are supplied to the muxer in form of a video stream with a single packet. There
-can be any number of those streams, each will correspond to a single APIC frame.
-The stream metadata tags @var{title} and @var{comment} map to APIC
-@var{description} and @var{picture type} respectively. See
-@url{http://id3.org/id3v2.4.0-frames} for allowed picture types.
-
-Note that the APIC frames must be written at the beginning, so the muxer will
-buffer the audio frames until it gets all the pictures. It is therefore advised
-to provide the pictures as soon as possible to avoid excessive buffering.
-
-Examples:
-
-Write an mp3 with an ID3v2.3 header and an ID3v1 footer:
-@example
-ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3
-@end example
-
-To attach a picture to an mp3 file select both the audio and the picture stream
-with @code{map}:
-@example
-ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1
-metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3
-@end example
-
-@section ogg
-
-Ogg container muxer.
-
-@table @option
-@item -page_duration @var{duration}
-Preferred page duration, in microseconds. The muxer will attempt to create
-pages that are approximately @var{duration} microseconds long. This allows the
-user to compromise between seek granularity and container overhead. The default
-is 1 second. A value of 0 will fill all segments, making pages as large as
-possible. A value of 1 will effectively use 1 packet-per-page in most
-situations, giving a small seek granularity at the cost of additional container
-overhead.
-@end table
-
@section tee

 The tee muxer can be used to write the same data to several files or any
--- a/doc/outdevs.texi
+++ b/doc/outdevs.texi
@@ -26,7 +26,7 @@ ALSA (Advanced Linux Sound Architecture) output device.

 CACA output device.

-This output devices allows to show a video stream in CACA window.
+This output device allows to show a video stream in CACA window.
 Only one CACA window is allowed per application, so you can
 have only one instance of this output device in an application.

@@ -112,7 +112,7 @@ OSS (Open Sound System) output device.

 SDL (Simple DirectMedia Layer) output device.

-This output devices allows to show a video stream in an SDL
+This output device allows to show a video stream in an SDL
 window. Only one SDL window is allowed per application, so you can
 have only one instance of this output device in an application.

@@ -153,4 +153,69 @@ ffmpeg -i INPUT -vcodec rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL

 sndio audio output device.

+@section xv
+
+XV (XVideo) output device.
+
+This output device allows to show a video stream in a X Window System
+window.
+
+@subsection Options
+
+@table @option
+@item display_name
+Specify the hardware display name, which determines the display and
+communications domain to be used.
+
+The display name or DISPLAY environment variable can be a string in
+the format @var{hostname}[:@var{number}[.@var{screen_number}]].
+
+@var{hostname} specifies the name of the host machine on which the
+display is physically attached. @var{number} specifies the number of
+the display server on that host machine. @var{screen_number} specifies
+the screen to be used on that server.
+
+If unspecified, it defaults to the value of the DISPLAY environment
+variable.
+
+For example, @code{dual-headed:0.1} would specify screen 1 of display
+0 on the machine named ``dual-headed''.
+
+Check the X11 specification for more detailed information about the
+display name format.
+
+@item window_size
+Set the created window size, can be a string of the form
+@var{width}x@var{height} or a video size abbreviation. If not
+specified it defaults to the size of the input video.
+
+@item window_x
+@item window_y
+Set the X and Y window offsets for the created window. They are both
+set to 0 by default. The values may be ignored by the window manager.
+
+@item window_title
+Set the window title, if not specified default to the filename
+specified for the output device.
+@end table
+
+For more information about XVideo see @url{http://www.x.org/}.
+
+@subsection Examples
+
+@itemize
+@item
+Decode, display and encode video input with @command{ffmpeg} at the
+same time:
+@example
+ffmpeg -i INPUT OUTPUT -f xv display
+@end example
+
+@item
+Decode and display the input video to multiple X11 windows:
+@example
+ffmpeg -i INPUT -f xv normal -vf negate -f xv negated
+@end example
+@end itemize
+
@c man end OUTPUT DEVICES
--- a/doc/platform.texi
+++ b/doc/platform.texi
@@ -106,10 +106,10 @@ libavformat) as DLLs.

@end itemize

-@section Microsoft Visual C++
+@section Microsoft Visual C++ or Intel C++ Compiler for Windows

-FFmpeg can be built with MSVC using a C99-to-C89 conversion utility and
-wrapper.
+FFmpeg can be built with MSVC or ICL using a C99-to-C89 conversion utility and
+wrapper. For ICL, only the wrapper is used, since ICL supports C99.

 You will need the following prerequisites:

@@ -122,28 +122,33 @@ You will need the following prerequisites:
 you want to run @uref{fate.html, FATE}.
@end itemize

-To set up a proper MSVC environment in MSYS, you simply need to run
-@code{msys.bat} from the Visual Studio command prompt.
+To set up a proper environment in MSYS, you need to run @code{msys.bat} from
+the Visual Studio or Intel Compiler command prompt.

 Place @code{makedef}, @code{c99wrap.exe}, @code{c99conv.exe}, and @code{yasm.exe}
 somewhere in your @code{PATH}.

 Next, make sure @code{inttypes.h} and any other headers and libs you want to use
-are located in a spot that MSVC can see. Do so by modifying the @code{LIB} and
-@code{INCLUDE} environment variables to include the @strong{Windows} paths to
+are located in a spot that the compiler can see. Do so by modifying the @code{LIB}
+and @code{INCLUDE} environment variables to include the @strong{Windows} paths to
 these directories. Alternatively, you can try and use the
@code{--extra-cflags}/@code{--extra-ldflags} configure options.

 Finally, run:

@example
+For MSVC:
 ./configure --toolchain=msvc
+
+For ICL:
+./configure --toolchain=icl
+
 make
 make install
@end example

 If you wish to compile shared libraries, add @code{--enable-shared} to your
-configure options. Note that due to the way MSVC handles DLL imports and
+configure options. Note that due to the way MSVC and ICL handle DLL imports and
 exports, you cannot compile static and shared libraries at the same time, and
 enabling shared libraries will automatically disable the static ones.

@@ -173,7 +178,12 @@ erroneously included when building FFmpeg.
 can see.
@end enumerate

-@item FFmpeg has been tested with Visual Studio 2010 and 2012, Pro and Express.
+@item FFmpeg has been tested with the following on i686 and x86_64:
+@itemize
+@item Visual Studio 2010 Pro and Express
+@item Visual Studio 2012 Pro and Express
+@item Intel Composer XE 2013
+@end itemize
 Anything else is not officially supported.

@end itemize
@@ -184,16 +194,7 @@ If you plan to link with MSVC-built static libraries, you will need
 to make sure you have @code{Runtime Library} set to
@code{Multi-threaded (/MT)} in your project's settings.

-FFmpeg headers do not declare global data for Windows DLLs through the usual
-dllexport/dllimport interface. Such data will be exported properly while
-building, but to use them in your MSVC code you will have to edit the
-appropriate headers and mark the data as dllimport. For example, in
-libavutil/pixdesc.h you should have:
-@example
-extern __declspec(dllimport) const AVPixFmtDescriptor av_pix_fmt_descriptors[];
-@end example
-
-You will also need to define @code{inline} to something MSVC understands:
+You will need to define @code{inline} to something MSVC understands:
@example
 #define inline __inline
@end example
--- a/doc/print_options.c
+++ b/doc/print_options.c
@@ -39,6 +39,9 @@ static void print_usage(void)

 static void print_option(const AVOption *opts, const AVOption *o, int per_stream)
 {
+    if (!(o->flags & (AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_ENCODING_PARAM)))
+        return;
+
    printf("@item -%s%s @var{", o->name, per_stream ? "[:stream_specifier]" : "");
    switch (o->type) {
    case AV_OPT_TYPE_BINARY:   printf("hexadecimal string"); break;
--- a/doc/protocols.texi
+++ b/doc/protocols.texi
@@ -49,6 +49,16 @@ Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapte
 -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
@end example

+@section cache
+
+Caching wrapper for input stream.
+
+Cache the input stream to temporary file. It brings seeking capability to live streams.
+
+@example
+cache:@var{URL}
+@end example
+
@section concat

 Physical concatenation protocol.
@@ -75,6 +85,25 @@ ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
 Note that you may need to escape the character "|" which is special for
 many shells.

+@section crypto
+
+AES-encrypted stream reading protocol.
+
+The accepted options are:
+@table @option
+@item key
+Set the AES decryption key binary block from given hexadecimal representation.
+
+@item iv
+Set the AES decryption initialization vector binary block from given hexadecimal representation.
+@end table
+
+Accepted URL formats:
+@example
+crypto:@var{URL}
+crypto+@var{URL}
+@end example
+
@section data

 Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
@@ -100,6 +129,40 @@ The ff* tools default to the file protocol, that is a resource
 specified with the name "FILE.mpeg" is interpreted as the URL
 "file:FILE.mpeg".

+@section ftp
+
+FTP (File Transfer Protocol).
+
+Allow to read from or write to remote resources using FTP protocol.
+
+Following syntax is required.
+@example
+ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
+@end example
+
+This protocol accepts the following options.
+
+@table @option
+@item timeout
+Set timeout of socket I/O operations used by the underlying low level
+operation. By default it is set to -1, which means that the timeout is
+not specified.
+
+@item ftp-anonymous-password
+Password used when login as anonymous user. Typically an e-mail address
+should be used.
+
+@item ftp-write-seekable
+Control seekability of connection during encoding. If set to 1 the
+resource is supposed to be seekable, if set to 0 it is assumed not
+to be seekable. Default value is 0.
+@end table
+
+NOTE: Protocol can be used as output, but it is recommended to not do
+it, unless special care is taken (tests, customized server configuration
+etc.). Different FTP servers behave in different way during seek
+operation. ff* tools may produce incomplete content due to server limitations.
+
@section gopher

 Gopher protocol.
@@ -165,6 +228,20 @@ not specified.
@item mime_type
 Set MIME type.

+@item icy
+If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
+supports this, the metadata has to be retrieved by the application by reading
+the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
+The default is 0.
+
+@item icy_metadata_headers
+If the server supports ICY metadata, this contains the ICY specific HTTP reply
+headers, separated with newline characters.
+
+@item icy_metadata_packet
+If the server supports ICY metadata, and @option{icy} was set to 1, this
+contains the last non-empty metadata packet sent by the server.
+
@item cookies
 Set the cookies to be sent in future requests. The format of each cookie is the
 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
@@ -538,6 +615,11 @@ To receive a stream in realtime:
 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
@end example

+@table @option
+@item stimeout
+Socket IO timeout in micro seconds.
+@end table
+
@section sap

 Session Announcement Protocol (RFC 2974). This is not technically a
@@ -630,6 +712,50 @@ To play back the first stream announced on one the default IPv6 SAP multicast ad
 ffplay sap://[ff0e::2:7ffe]
@end example

+@section sctp
+
+Stream Control Transmission Protocol.
+
+The accepted URL syntax is:
+@example
+sctp://@var{host}:@var{port}[?@var{options}]
+@end example
+
+The protocol accepts the following options:
+@table @option
+@item listen
+If set to any value, listen for an incoming connection. Outgoing connection is done by default.
+
+@item max_streams
+Set the maximum number of streams. By default no limit is set.
+@end table
+
+@section srtp
+
+Secure Real-time Transport Protocol.
+
+The accepted options are:
+@table @option
+@item srtp_in_suite
+@item srtp_out_suite
+Select input and output encoding suites.
+
+Supported values:
+@table @samp
+@item AES_CM_128_HMAC_SHA1_80
+@item SRTP_AES128_CM_HMAC_SHA1_80
+@item AES_CM_128_HMAC_SHA1_32
+@item SRTP_AES128_CM_HMAC_SHA1_32
+@end table
+
+@item srtp_in_params
+@item srtp_out_params
+Set input and output encoding parameters, which are expressed by a
+base64-encoded representation of a binary block. The first 16 bytes of
+this binary block are used as master key, the following 14 bytes are
+used as master salt.
+@end table
+
@section tcp

 Trasmission Control Protocol.
--- a/doc/resampler.texi
+++ b/doc/resampler.texi
@@ -0,0 +1,226 @@
+@chapter Resampler Options
+@c man begin RESAMPLER OPTIONS
+
+The audio resampler supports the following named options.
+
+Options may be set by specifying -@var{option} @var{value} in the
+FFmpeg tools, @var{option}=@var{value} for the aresample filter,
+by setting the value explicitly in the
+@code{SwrContext} options or using the @file{libavutil/opt.h} API for
+programmatic use.
+
+@table @option
+
+@item ich, in_channel_count
+Set the number of input channels. Default value is 0. Setting this
+value is not mandatory if the corresponding channel layout
+@option{in_channel_layout} is set.
+
+@item och, out_channel_count
+Set the number of output channels. Default value is 0. Setting this
+value is not mandatory if the corresponding channel layout
+@option{out_channel_layout} is set.
+
+@item uch, used_channel_count
+Set the number of used input channels. Default value is 0. This option is
+only used for special remapping.
+
+@item isr, in_sample_rate
+Set the input sample rate. Default value is 0.
+
+@item osr, out_sample_rate
+Set the output sample rate. Default value is 0.
+
+@item isf, in_sample_fmt
+Specify the input sample format. It is set by default to @code{none}.
+
+@item osf, out_sample_fmt
+Specify the output sample format. It is set by default to @code{none}.
+
+@item tsf, internal_sample_fmt
+Set the internal sample format. Default value is @code{none}.
+This will automatically be chosen when it is not explicitly set.
+
+@item icl, in_channel_layout
+Set the input channel layout.
+
+@item ocl, out_channel_layout
+Set the output channel layout.
+
+@item clev, center_mix_level
+Set the center mix level. It is a value expressed in deciBel, and must be
+in the interval [-32,32].
+
+@item slev, surround_mix_level
+Set the surround mix level. It is a value expressed in deciBel, and must
+be in the interval [-32,32].
+
+@item lfe_mix_level
+Set LFE mix into non LFE level. It is used when there is a LFE input but no
+LFE output. It is a value expressed in deciBel, and must
+be in the interval [-32,32].
+
+@item rmvol, rematrix_volume
+Set rematrix volume. Default value is 1.0.
+
+@item flags, swr_flags
+Set flags used by the converter. Default value is 0.
+
+It supports the following individual flags:
+@table @option
+@item res
+force resampling, this flag forces resampling to be used even when the
+input and output sample rates match.
+@end table
+
+@item dither_scale
+Set the dither scale. Default value is 1.
+
+@item dither_method
+Set dither method. Default value is 0.
+
+Supported values:
+@table @samp
+@item rectangular
+select rectangular dither
+@item triangular
+select triangular dither
+@item triangular_hp
+select triangular dither with high pass
+@item lipshitz
+select lipshitz noise shaping dither
+@item shibata
+select shibata noise shaping dither
+@item low_shibata
+select low shibata noise shaping dither
+@item high_shibata
+select high shibata noise shaping dither
+@item f_weighted
+select f-weighted noise shaping dither
+@item modified_e_weighted
+select modified-e-weighted noise shaping dither
+@item improved_e_weighted
+select improved-e-weighted noise shaping dither
+
+@end table
+
+@item resampler
+Set resampling engine. Default value is swr.
+
+Supported values:
+@table @samp
+@item swr
+select the native SW Resampler; filter options precision and cheby are not
+applicable in this case.
+@item soxr
+select the SoX Resampler (where available); compensation, and filter options
+filter_size, phase_shift, filter_type & kaiser_beta, are not applicable in this
+case.
+@end table
+
+@item filter_size
+For swr only, set resampling filter size, default value is 32.
+
+@item phase_shift
+For swr only, set resampling phase shift, default value is 10, and must be in
+the interval [0,30].
+
+@item linear_interp
+Use Linear Interpolation if set to 1, default value is 0.
+
+@item cutoff
+Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float
+value between 0 and 1.  Default value is 0.97 with swr, and 0.91 with soxr
+(which, with a sample-rate of 44100, preserves the entire audio band to 20kHz).
+
+@item precision
+For soxr only, the precision in bits to which the resampled signal will be
+calculated.  The default value of 20 (which, with suitable dithering, is
+appropriate for a destination bit-depth of 16) gives SoX's 'High Quality'; a
+value of 28 gives SoX's 'Very High Quality'.
+
+@item cheby
+For soxr only, selects passband rolloff none (Chebyshev) & higher-precision
+approximation for 'irrational' ratios. Default value is 0.
+
+@item async
+For swr only, simple 1 parameter audio sync to timestamps using stretching,
+squeezing, filling and trimming. Setting this to 1 will enable filling and
+trimming, larger values represent the maximum amount in samples that the data
+may be stretched or squeezed for each second.
+Default value is 0, thus no compensation is applied to make the samples match
+the audio timestamps.
+
+@item first_pts
+For swr only, assume the first pts should be this value. The time unit is 1 / sample rate.
+This allows for padding/trimming at the start of stream. By default, no
+assumption is made about the first frame's expected pts, so no padding or
+trimming is done. For example, this could be set to 0 to pad the beginning with
+silence if an audio stream starts after the video stream or to trim any samples
+with a negative pts due to encoder delay.
+
+@item min_comp
+For swr only, set the minimum difference between timestamps and audio data (in
+seconds) to trigger stretching/squeezing/filling or trimming of the
+data to make it match the timestamps. The default is that
+stretching/squeezing/filling and trimming is disabled
+(@option{min_comp} = @code{FLT_MAX}).
+
+@item min_hard_comp
+For swr only, set the minimum difference between timestamps and audio data (in
+seconds) to trigger adding/dropping samples to make it match the
+timestamps.  This option effectively is a threshold to select between
+hard (trim/fill) and soft (squeeze/stretch) compensation. Note that
+all compensation is by default disabled through @option{min_comp}.
+The default is 0.1.
+
+@item comp_duration
+For swr only, set duration (in seconds) over which data is stretched/squeezed
+to make it match the timestamps. Must be a non-negative double float value,
+default value is 1.0.
+
+@item max_soft_comp
+For swr only, set maximum factor by which data is stretched/squeezed to make it
+match the timestamps. Must be a non-negative double float value, default value
+is 0.
+
+@item matrix_encoding
+Select matrixed stereo encoding.
+
+It accepts the following values:
+@table @samp
+@item none
+select none
+@item dolby
+select Dolby
+@item dplii
+select Dolby Pro Logic II
+@end table
+
+Default value is @code{none}.
+
+@item filter_type
+For swr only, select resampling filter type. This only affects resampling
+operations.
+
+It accepts the following values:
+@table @samp
+@item cubic
+select cubic
+@item blackman_nuttall
+select Blackman Nuttall Windowed Sinc
+@item kaiser
+select Kaiser Windowed Sinc
+@end table
+
+@item kaiser_beta
+For swr only, set Kaiser Window Beta value. Must be an integer in the
+interval [2,16], default value is 9.
+
+@item output_sample_bits
+For swr only, set number of used output sample bits for dithering. Must be an integer in the
+interval [0,64], default value is 0, which means it's not used.
+
+@end table
+
+@c man end RESAMPLER OPTIONS
--- a/doc/scaler.texi
+++ b/doc/scaler.texi
@@ -0,0 +1,99 @@
+@chapter Scaler Options
+@c man begin SCALER OPTIONS
+
+The video scaler supports the following named options.
+
+Options may be set by specifying -@var{option} @var{value} in the
+FFmpeg tools. For programmatic use, they can be set explicitly in the
+@code{SwsContext} options or through the @file{libavutil/opt.h} API.
+
+@table @option
+
+@item sws_flags
+Set the scaler flags. This is also used to set the scaling
+algorithm. Only a single algorithm should be selected.
+
+It accepts the following values:
+@table @samp
+@item fast_bilinear
+Select fast bilinear scaling algorithm.
+
+@item bilinear
+Select bilinear scaling algorithm.
+
+@item bicubic
+Select bicubic scaling algorithm.
+
+@item experimental
+Select experimental scaling algorithm.
+
+@item neighbor
+Select nearest neighbor rescaling algorithm.
+
+@item area
+Select averaging area rescaling algorithm.
+
+@item bicubiclin
+Select bicubic scaling algorithm for the luma component, bilinear for
+chroma components.
+
+@item gauss
+Select Gaussian rescaling algorithm.
+
+@item sinc
+Select sinc rescaling algorithm.
+
+@item lanczos
+Select lanczos rescaling algorithm.
+
+@item spline
+Select natural bicubic spline rescaling algorithm.
+
+@item print_info
+Enable printing/debug logging.
+
+@item accurate_rnd
+Enable accurate rounding.
+
+@item full_chroma_int
+Enable full chroma interpolation.
+
+@item full_chroma_inp
+Select full chroma input.
+
+@item bitexact
+Enable bitexact output.
+@end table
+
+@item srcw
+Set source width.
+
+@item srch
+Set source height.
+
+@item dstw
+Set destination width.
+
+@item dsth
+Set destination height.
+
+@item src_format
+Set source pixel format (must be expressed as an integer).
+
+@item dst_format
+Set destination pixel format (must be expressed as an integer).
+
+@item src_range
+Select source range.
+
+@item dst_range
+Select destination range.
+
+@item param0, param1
+Set scaling algorithm parameters. The specified values are specific of
+some scaling algorithms and ignored by others. The specified values
+are floating point number values.
+
+@end table
+
+@c man end SCALER OPTIONS
--- a/doc/syntax.texi
+++ b/doc/syntax.texi
@@ -1,258 +0,0 @@
-@chapter Syntax
-@c man begin SYNTAX
-
-This section documents the syntax and formats employed by the FFmpeg
-libraries and tools.
-
-@anchor{quoting_and_escaping}
-@section Quoting and escaping
-
-FFmpeg adopts the following quoting and escaping mechanism, unless
-explicitly specified. The following rules are applied:
-
-@itemize
-@item
-@code{'} and @code{\} are special characters (respectively used for
-quoting and escaping). In addition to them, there might be other
-special characters depending on the specific syntax where the escaping
-and quoting are employed.
-
-@item
-A special character is escaped by prefixing it with a '\'.
-
-@item
-All characters enclosed between '' are included literally in the
-parsed string. The quote character @code{'} itself cannot be quoted,
-so you may need to close the quote and escape it.
-
-@item
-Leading and trailing whitespaces, unless escaped or quoted, are
-removed from the parsed string.
-@end itemize
-
-Note that you may need to add a second level of escaping when using
-the command line or a script, which depends on the syntax of the
-adopted shell language.
-
-The function @code{av_get_token} defined in
-@file{libavutil/avstring.h} can be used to parse a token quoted or
-escaped according to the rules defined above.
-
-The tool @file{tools/ffescape} in the FFmpeg source tree can be used
-to automatically quote or escape a string in a script.
-
-@subsection Examples
-
-@itemize
-@item
-Escape the string @code{Crime d'Amour} containing the @code{'} special
-character:
-@example
-Crime d\'Amour
-@end example
-
-@item
-The string above contains a quote, so the @code{'} needs to be escaped
-when quoting it:
-@example
-'Crime d'\''Amour'
-@end example
-
-@item
-Include leading or trailing whitespaces using quoting:
-@example
-'  this string starts and ends with whitespaces  '
-@end example
-
-@item
-Escaping and quoting can be mixed together:
-@example
-' The string '\'string\'' is a string '
-@end example
-
-@item
-To include a literal @code{\} you can use either escaping or quoting:
-@example
-'c:\foo' can be written as c:\\foo
-@end example
-@end itemize
-
-@anchor{date syntax}
-@section Date
-
-The accepted syntax is:
-@example
-[(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z]
-now
-@end example
-
-If the value is "now" it takes the current time.
-
-Time is local time unless Z is appended, in which case it is
-interpreted as UTC.
-If the year-month-day part is not specified it takes the current
-year-month-day.
-
-@anchor{time duration syntax}
-@section Time duration
-
-The accepted syntax is:
-@example
-[-][HH:]MM:SS[.m...]
-[-]S+[.m...]
-@end example
-
-@var{HH} expresses the number of hours, @var{MM} the number a of minutes
-and @var{SS} the number of seconds.
-
-@anchor{video size syntax}
-@section Video size
-Specify the size of the sourced video, it may be a string of the form
-@var{width}x@var{height}, or the name of a size abbreviation.
-
-The following abbreviations are recognized:
-@table @samp
-@item ntsc
-720x480
-@item pal
-720x576
-@item qntsc
-352x240
-@item qpal
-352x288
-@item sntsc
-640x480
-@item spal
-768x576
-@item film
-352x240
-@item ntsc-film
-352x240
-@item sqcif
-128x96
-@item qcif
-176x144
-@item cif
-352x288
-@item 4cif
-704x576
-@item 16cif
-1408x1152
-@item qqvga
-160x120
-@item qvga
-320x240
-@item vga
-640x480
-@item svga
-800x600
-@item xga
-1024x768
-@item uxga
-1600x1200
-@item qxga
-2048x1536
-@item sxga
-1280x1024
-@item qsxga
-2560x2048
-@item hsxga
-5120x4096
-@item wvga
-852x480
-@item wxga
-1366x768
-@item wsxga
-1600x1024
-@item wuxga
-1920x1200
-@item woxga
-2560x1600
-@item wqsxga
-3200x2048
-@item wquxga
-3840x2400
-@item whsxga
-6400x4096
-@item whuxga
-7680x4800
-@item cga
-320x200
-@item ega
-640x350
-@item hd480
-852x480
-@item hd720
-1280x720
-@item hd1080
-1920x1080
-@item 2k
-2048x1080
-@item 2kflat
-1998x1080
-@item 2kscope
-2048x858
-@item 4k
-4096x2160
-@item 4kflat
-3996x2160
-@item 4kscope
-4096x1716
-@end table
-
-@anchor{video rate syntax}
-@section Video rate
-
-Specify the frame rate of a video, expressed as the number of frames
-generated per second. It has to be a string in the format
-@var{frame_rate_num}/@var{frame_rate_den}, an integer number, a float
-number or a valid video frame rate abbreviation.
-
-The following abbreviations are recognized:
-@table @samp
-@item ntsc
-30000/1001
-@item pal
-25/1
-@item qntsc
-30000/1001
-@item qpal
-25/1
-@item sntsc
-30000/1001
-@item spal
-25/1
-@item film
-24/1
-@item ntsc-film
-24000/1001
-@end table
-
-@anchor{ratio syntax}
-@section Ratio
-
-A ratio can be expressed as an expression, or in the form
-@var{numerator}:@var{denominator}.
-
-Note that a ratio with infinite (1/0) or negative value is
-considered valid, so you should check on the returned value if you
-want to exclude those values.
-
-The undefined value can be expressed using the "0:0" string.
-
-@anchor{color syntax}
-@section Color
-
-It can be the name of a color (case insensitive match) or a
-[0x|#]RRGGBB[AA] sequence, possibly followed by "@@" and a string
-representing the alpha component.
-
-The alpha component may be a string composed by "0x" followed by an
-hexadecimal number or a decimal number between 0.0 and 1.0, which
-represents the opacity value (0x00/0.0 means completely transparent,
-0xff/1.0 completely opaque).
-If the alpha component is not specified then 0xff is assumed.
-
-The string "random" will result in a random color.
-
-@c man end SYNTAX
--- a/doc/texi2pod.pl
+++ b/doc/texi2pod.pl
@@ -121,7 +121,7 @@ INF: while(<$inf>) {
        $chapters{$chapter_name} .= postprocess($chapter) if ($chapter_name);

        # start new chapter
-        $chapter_name = $1, push (@chapters_sequence, $chapter_name);
+        $chapter_name = $1, push (@chapters_sequence, $chapter_name) unless $skipping;
        $chapters{$chapter_name} = "" unless exists $chapters{$chapter_name};
        $chapter = "";
        $output = 1;
--- a/doc/utils.texi
+++ b/doc/utils.texi
@@ -1,3 +1,262 @@
+@chapter Syntax
+@c man begin SYNTAX
+
+This section documents the syntax and formats employed by the FFmpeg
+libraries and tools.
+
+@anchor{quoting_and_escaping}
+@section Quoting and escaping
+
+FFmpeg adopts the following quoting and escaping mechanism, unless
+explicitly specified. The following rules are applied:
+
+@itemize
+@item
+@code{'} and @code{\} are special characters (respectively used for
+quoting and escaping). In addition to them, there might be other
+special characters depending on the specific syntax where the escaping
+and quoting are employed.
+
+@item
+A special character is escaped by prefixing it with a '\'.
+
+@item
+All characters enclosed between '' are included literally in the
+parsed string. The quote character @code{'} itself cannot be quoted,
+so you may need to close the quote and escape it.
+
+@item
+Leading and trailing whitespaces, unless escaped or quoted, are
+removed from the parsed string.
+@end itemize
+
+Note that you may need to add a second level of escaping when using
+the command line or a script, which depends on the syntax of the
+adopted shell language.
+
+The function @code{av_get_token} defined in
+@file{libavutil/avstring.h} can be used to parse a token quoted or
+escaped according to the rules defined above.
+
+The tool @file{tools/ffescape} in the FFmpeg source tree can be used
+to automatically quote or escape a string in a script.
+
+@subsection Examples
+
+@itemize
+@item
+Escape the string @code{Crime d'Amour} containing the @code{'} special
+character:
+@example
+Crime d\'Amour
+@end example
+
+@item
+The string above contains a quote, so the @code{'} needs to be escaped
+when quoting it:
+@example
+'Crime d'\''Amour'
+@end example
+
+@item
+Include leading or trailing whitespaces using quoting:
+@example
+'  this string starts and ends with whitespaces  '
+@end example
+
+@item
+Escaping and quoting can be mixed together:
+@example
+' The string '\'string\'' is a string '
+@end example
+
+@item
+To include a literal @code{\} you can use either escaping or quoting:
+@example
+'c:\foo' can be written as c:\\foo
+@end example
+@end itemize
+
+@anchor{date syntax}
+@section Date
+
+The accepted syntax is:
+@example
+[(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z]
+now
+@end example
+
+If the value is "now" it takes the current time.
+
+Time is local time unless Z is appended, in which case it is
+interpreted as UTC.
+If the year-month-day part is not specified it takes the current
+year-month-day.
+
+@anchor{time duration syntax}
+@section Time duration
+
+The accepted syntax is:
+@example
+[-][HH:]MM:SS[.m...]
+[-]S+[.m...]
+@end example
+
+@var{HH} expresses the number of hours, @var{MM} the number a of minutes
+and @var{SS} the number of seconds.
+
+@anchor{video size syntax}
+@section Video size
+Specify the size of the sourced video, it may be a string of the form
+@var{width}x@var{height}, or the name of a size abbreviation.
+
+The following abbreviations are recognized:
+@table @samp
+@item ntsc
+720x480
+@item pal
+720x576
+@item qntsc
+352x240
+@item qpal
+352x288
+@item sntsc
+640x480
+@item spal
+768x576
+@item film
+352x240
+@item ntsc-film
+352x240
+@item sqcif
+128x96
+@item qcif
+176x144
+@item cif
+352x288
+@item 4cif
+704x576
+@item 16cif
+1408x1152
+@item qqvga
+160x120
+@item qvga
+320x240
+@item vga
+640x480
+@item svga
+800x600
+@item xga
+1024x768
+@item uxga
+1600x1200
+@item qxga
+2048x1536
+@item sxga
+1280x1024
+@item qsxga
+2560x2048
+@item hsxga
+5120x4096
+@item wvga
+852x480
+@item wxga
+1366x768
+@item wsxga
+1600x1024
+@item wuxga
+1920x1200
+@item woxga
+2560x1600
+@item wqsxga
+3200x2048
+@item wquxga
+3840x2400
+@item whsxga
+6400x4096
+@item whuxga
+7680x4800
+@item cga
+320x200
+@item ega
+640x350
+@item hd480
+852x480
+@item hd720
+1280x720
+@item hd1080
+1920x1080
+@item 2k
+2048x1080
+@item 2kflat
+1998x1080
+@item 2kscope
+2048x858
+@item 4k
+4096x2160
+@item 4kflat
+3996x2160
+@item 4kscope
+4096x1716
+@end table
+
+@anchor{video rate syntax}
+@section Video rate
+
+Specify the frame rate of a video, expressed as the number of frames
+generated per second. It has to be a string in the format
+@var{frame_rate_num}/@var{frame_rate_den}, an integer number, a float
+number or a valid video frame rate abbreviation.
+
+The following abbreviations are recognized:
+@table @samp
+@item ntsc
+30000/1001
+@item pal
+25/1
+@item qntsc
+30000/1001
+@item qpal
+25/1
+@item sntsc
+30000/1001
+@item spal
+25/1
+@item film
+24/1
+@item ntsc-film
+24000/1001
+@end table
+
+@anchor{ratio syntax}
+@section Ratio
+
+A ratio can be expressed as an expression, or in the form
+@var{numerator}:@var{denominator}.
+
+Note that a ratio with infinite (1/0) or negative value is
+considered valid, so you should check on the returned value if you
+want to exclude those values.
+
+The undefined value can be expressed using the "0:0" string.
+
+@anchor{color syntax}
+@section Color
+
+It can be the name of a color (case insensitive match) or a
+[0x|#]RRGGBB[AA] sequence, possibly followed by "@@" and a string
+representing the alpha component.
+
+The alpha component may be a string composed by "0x" followed by an
+hexadecimal number or a decimal number between 0.0 and 1.0, which
+represents the opacity value (0x00/0.0 means completely transparent,
+0xff/1.0 completely opaque).
+If the alpha component is not specified then 0xff is assumed.
+
+The string "random" will result in a random color.
+
+@c man end SYNTAX
+
@chapter Expression Evaluation
@c man begin EXPRESSION EVALUATION

@@ -32,6 +291,21 @@ Compute arcsine of @var{x}.
@item atan(x)
 Compute arctangent of @var{x}.

+@item between(x, min, max)
+Return 1 if @var{x} is greater than or equal to @var{min} and lesser than or
+equal to @var{max}, 0 otherwise.
+
+@item bitand(x, y)
+@item bitor(x, y)
+Compute bitwise and/or operation on @var{x} and @var{y}.
+
+The results of the evaluation of @var{x} and @var{y} are converted to
+integers before executing the bitwise operation.
+
+Note that both the conversion to integer and the conversion back to
+floating point can lose precision. Beware of unexpected results for
+large numbers (usually 2^53 and larger).
+
@item ceil(expr)
 Round the value of expression @var{expr} upwards to the nearest
 integer. For example, "ceil(1.5)" is "2.0".
@@ -286,3 +560,33 @@ indication of the corresponding powers of 10 and of 2.
@end table

@c man end
+
+@chapter OpenCL Options
+@c man begin OPENCL OPTIONS
+
+When FFmpeg is configured with @code{--enable-opencl}, it is possible
+to set the options for the global OpenCL context.
+
+The list of supported options follows:
+
+@table @option
+@item build_options
+Set build options used to compile the registered kernels.
+
+See reference "OpenCL Specification Version: 1.2 chapter 5.6.4".
+
+@item platform_idx
+Select the index of the platform to run OpenCL code.
+
+The specified index must be one of the indexes in the device list
+which can be obtained with @code{av_opencl_get_device_list()}.
+
+@item device_idx
+Select the index of the device used to run OpenCL code.
+
+The specifed index must be one of the indexes in the device list which
+can be obtained with @code{av_opencl_get_device_list()}.
+
+@end table
+
+@c man end OPENCL OPTIONS
--- a/ffmpeg.c
+++ b/ffmpeg.c
--- a/ffmpeg.h
+++ b/ffmpeg.h
@@ -37,7 +37,6 @@
 #include "libavcodec/avcodec.h"

 #include "libavfilter/avfilter.h"
-#include "libavfilter/avfiltergraph.h"

 #include "libavutil/avutil.h"
 #include "libavutil/dict.h"
@@ -165,16 +164,22 @@ typedef struct OptionsContext {
    int        nb_copy_prior_start;
    SpecifierOpt *filters;
    int        nb_filters;
+    SpecifierOpt *filter_scripts;
+    int        nb_filter_scripts;
    SpecifierOpt *reinit_filters;
    int        nb_reinit_filters;
    SpecifierOpt *fix_sub_duration;
    int        nb_fix_sub_duration;
+    SpecifierOpt *canvas_sizes;
+    int        nb_canvas_sizes;
    SpecifierOpt *pass;
    int        nb_pass;
    SpecifierOpt *passlogfiles;
    int        nb_passlogfiles;
    SpecifierOpt *guess_layout_max;
    int        nb_guess_layout_max;
+    SpecifierOpt *apad;
+    int        nb_apad;
 } OptionsContext;

 typedef struct InputFilter {
@@ -214,6 +219,7 @@ typedef struct InputStream {
    int decoding_needed;     /* true if the packets must be decoded in 'raw_fifo' */
    AVCodec *dec;
    AVFrame *decoded_frame;
+    AVFrame *filter_frame; /* a ref of decoded_frame, to be sent to filters */

    int64_t       start;     /* time when read started */
    /* predicted dts of the next packet read for this stream or (when there are
@@ -255,12 +261,10 @@ typedef struct InputStream {
    struct sub2video {
        int64_t last_pts;
        int64_t end_pts;
-        AVFilterBufferRef *ref;
+        AVFrame *frame;
        int w, h;
    } sub2video;

-    /* a pool of free buffers for decoded data */
-    FrameBuffer *buffer_pool;
    int dr1;

    /* decoded data from this stream goes into all those filters
@@ -277,6 +281,7 @@ typedef struct InputFile {
    int eagain;           /* true if last read attempt returned EAGAIN */
    int ist_index;        /* index of first stream in input_streams */
    int64_t ts_offset;
+    int64_t last_ts;
    int nb_streams;       /* number of stream that ffmpeg is aware of; may be different
                             from ctx.nb_streams if new streams appear during av_read_frame() */
    int nb_streams_warn;  /* number of streams that the user was warned of */
@@ -317,6 +322,8 @@ typedef struct OutputStream {
    /* pts of the first frame encoded for this stream, used for limiting
     * recording time */
    int64_t first_pts;
+    /* dts of the last packet sent to the muxer */
+    int64_t last_mux_dts;
    AVBitStreamFilterContext *bitstream_filters;
    AVCodec *enc;
    int64_t max_frames;
@@ -327,7 +334,7 @@ typedef struct OutputStream {
    int force_fps;
    int top_field_first;

-    float frame_aspect_ratio;
+    AVRational frame_aspect_ratio;

    /* forced key frames */
    int64_t *forced_kf_pts;
@@ -351,6 +358,7 @@ typedef struct OutputStream {
    AVDictionary *opts;
    AVDictionary *swr_opts;
    AVDictionary *resample_opts;
+    char *apad;
    int finished;        /* no more packets should be written for this stream */
    int unavailable;                     /* true if the steram is unavailable (possibly temporarily) */
    int stream_copy;
--- a/ffmpeg_filter.c
+++ b/ffmpeg_filter.c
@@ -21,7 +21,6 @@
 #include "ffmpeg.h"

 #include "libavfilter/avfilter.h"
-#include "libavfilter/avfiltergraph.h"
 #include "libavfilter/buffersink.h"

 #include "libavresample/avresample.h"
@@ -93,6 +92,11 @@ void choose_sample_fmt(AVStream *st, AVCodec *codec)

 static char *choose_pix_fmts(OutputStream *ost)
 {
+    AVDictionaryEntry *strict_dict = av_dict_get(ost->opts, "strict", NULL, 0);
+    if (strict_dict)
+        // used by choose_pixel_fmt() and below
+        av_opt_set(ost->st->codec, "strict", strict_dict->value, 0);
+
     if (ost->keep_pix_fmt) {
        if (ost->filter)
            avfilter_graph_set_auto_convert(ost->filter->graph->graph,
@@ -110,7 +114,7 @@ static char *choose_pix_fmts(OutputStream *ost)
        int len;

        if (avio_open_dyn_buf(&s) < 0)
-            exit(1);
+            exit_program(1);

        p = ost->enc->pix_fmts;
        if (ost->st->codec->strict_std_compliance <= FF_COMPLIANCE_UNOFFICIAL) {
@@ -124,7 +128,7 @@ static char *choose_pix_fmts(OutputStream *ost)

        for (; *p != AV_PIX_FMT_NONE; p++) {
            const char *name = av_get_pix_fmt_name(*p);
-            avio_printf(s, "%s:", name);
+            avio_printf(s, "%s|", name);
        }
        len = avio_close_dyn_buf(s, &ret);
        ret[len - 1] = 0;
@@ -135,7 +139,7 @@ static char *choose_pix_fmts(OutputStream *ost)

 /* Define a function for building a string containing a list of
 * allowed formats. */
-#define DEF_CHOOSE_FORMAT(type, var, supported_list, none, get_name, separator)\
+#define DEF_CHOOSE_FORMAT(type, var, supported_list, none, get_name)           \
 static char *choose_ ## var ## s(OutputStream *ost)                            \
 {                                                                              \
    if (ost->st->codec->var != none) {                                         \
@@ -148,11 +152,11 @@ static char *choose_ ## var ## s(OutputStream *ost)                            \
        int len;                                                               \
                                                                               \
        if (avio_open_dyn_buf(&s) < 0)                                         \
-            exit(1);                                                           \
+            exit_program(1);                                                           \
                                                                               \
        for (p = ost->enc->supported_list; *p != none; p++) {                  \
            get_name(*p);                                                      \
-            avio_printf(s, "%s" separator, name);                              \
+            avio_printf(s, "%s|", name);                                       \
        }                                                                      \
        len = avio_close_dyn_buf(s, &ret);                                     \
        ret[len - 1] = 0;                                                      \
@@ -162,28 +166,28 @@ static char *choose_ ## var ## s(OutputStream *ost)                            \
 }

 // DEF_CHOOSE_FORMAT(enum AVPixelFormat, pix_fmt, pix_fmts, AV_PIX_FMT_NONE,
-//                   GET_PIX_FMT_NAME, ":")
+//                   GET_PIX_FMT_NAME)

 DEF_CHOOSE_FORMAT(enum AVSampleFormat, sample_fmt, sample_fmts,
-                  AV_SAMPLE_FMT_NONE, GET_SAMPLE_FMT_NAME, ",")
+                  AV_SAMPLE_FMT_NONE, GET_SAMPLE_FMT_NAME)

 DEF_CHOOSE_FORMAT(int, sample_rate, supported_samplerates, 0,
-                  GET_SAMPLE_RATE_NAME, ",")
+                  GET_SAMPLE_RATE_NAME)

 DEF_CHOOSE_FORMAT(uint64_t, channel_layout, channel_layouts, 0,
-                  GET_CH_LAYOUT_NAME, ",")
+                  GET_CH_LAYOUT_NAME)

 FilterGraph *init_simple_filtergraph(InputStream *ist, OutputStream *ost)
 {
    FilterGraph *fg = av_mallocz(sizeof(*fg));

    if (!fg)
-        exit(1);
+        exit_program(1);
    fg->index = nb_filtergraphs;

    GROW_ARRAY(fg->outputs, fg->nb_outputs);
    if (!(fg->outputs[0] = av_mallocz(sizeof(*fg->outputs[0]))))
-        exit(1);
+        exit_program(1);
    fg->outputs[0]->ost   = ost;
    fg->outputs[0]->graph = fg;

@@ -191,7 +195,7 @@ FilterGraph *init_simple_filtergraph(InputStream *ist, OutputStream *ost)

    GROW_ARRAY(fg->inputs, fg->nb_inputs);
    if (!(fg->inputs[0] = av_mallocz(sizeof(*fg->inputs[0]))))
-        exit(1);
+        exit_program(1);
    fg->inputs[0]->ist   = ist;
    fg->inputs[0]->graph = fg;

@@ -214,7 +218,7 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
    if (type != AVMEDIA_TYPE_VIDEO && type != AVMEDIA_TYPE_AUDIO) {
        av_log(NULL, AV_LOG_FATAL, "Only video and audio filters supported "
               "currently.\n");
-        exit(1);
+        exit_program(1);
    }

    if (in->name) {
@@ -226,7 +230,7 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
        if (file_idx < 0 || file_idx >= nb_input_files) {
            av_log(NULL, AV_LOG_FATAL, "Invalid file index %d in filtergraph description %s.\n",
                   file_idx, fg->graph_desc);
-            exit(1);
+            exit_program(1);
        }
        s = input_files[file_idx]->ctx;

@@ -244,7 +248,7 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
        if (!st) {
            av_log(NULL, AV_LOG_FATAL, "Stream specifier '%s' in filtergraph description %s "
                   "matches no streams.\n", p, fg->graph_desc);
-            exit(1);
+            exit_program(1);
        }
        ist = input_streams[input_files[file_idx]->ist_index + st->index];
    } else {
@@ -258,7 +262,7 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
            av_log(NULL, AV_LOG_FATAL, "Cannot find a matching stream for "
                   "unlabeled input pad %d on filter %s\n", in->pad_idx,
                   in->filter_ctx->name);
-            exit(1);
+            exit_program(1);
        }
    }
    av_assert0(ist);
@@ -269,7 +273,7 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)

    GROW_ARRAY(fg->inputs, fg->nb_inputs);
    if (!(fg->inputs[fg->nb_inputs - 1] = av_mallocz(sizeof(*fg->inputs[0]))))
-        exit(1);
+        exit_program(1);
    fg->inputs[fg->nb_inputs - 1]->ist   = ist;
    fg->inputs[fg->nb_inputs - 1]->graph = fg;

@@ -277,6 +281,62 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
    ist->filters[ist->nb_filters - 1] = fg->inputs[fg->nb_inputs - 1];
 }

+static int insert_trim(OutputStream *ost, AVFilterContext **last_filter, int *pad_idx)
+{
+    OutputFile *of = output_files[ost->file_index];
+    AVFilterGraph *graph = (*last_filter)->graph;
+    AVFilterContext *ctx;
+    const AVFilter *trim;
+    const char *name = ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO ? "trim" : "atrim";
+    char filter_name[128];
+    int ret = 0;
+
+    if (of->recording_time == INT64_MAX && !of->start_time)
+        return 0;
+
+    // Use with duration and without output starttime is buggy with trim filters
+    if (!of->start_time)
+        return 0;
+
+    trim = avfilter_get_by_name(name);
+    if (!trim) {
+        av_log(NULL, AV_LOG_ERROR, "%s filter not present, cannot limit "
+               "recording time.\n", name);
+        return AVERROR_FILTER_NOT_FOUND;
+    }
+
+    snprintf(filter_name, sizeof(filter_name), "%s for output stream %d:%d",
+             name, ost->file_index, ost->index);
+    ctx = avfilter_graph_alloc_filter(graph, trim, filter_name);
+    if (!ctx)
+        return AVERROR(ENOMEM);
+
+    if (of->recording_time != INT64_MAX) {
+        ret = av_opt_set_double(ctx, "duration", (double)of->recording_time / 1e6,
+                                AV_OPT_SEARCH_CHILDREN);
+    }
+    if (ret >= 0 && of->start_time) {
+        ret = av_opt_set_double(ctx, "start", (double)of->start_time / 1e6,
+                                AV_OPT_SEARCH_CHILDREN);
+    }
+    if (ret < 0) {
+        av_log(ctx, AV_LOG_ERROR, "Error configuring the %s filter", name);
+        return ret;
+    }
+
+    ret = avfilter_init_str(ctx, NULL);
+    if (ret < 0)
+        return ret;
+
+    ret = avfilter_link(*last_filter, *pad_idx, ctx, 0);
+    if (ret < 0)
+        return ret;
+
+    *last_filter = ctx;
+    *pad_idx     = 0;
+    return 0;
+}
+
 static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out)
 {
    char *pix_fmts;
@@ -286,13 +346,11 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter,
    int pad_idx = out->pad_idx;
    int ret;
    char name[255];
-    AVBufferSinkParams *buffersink_params = av_buffersink_params_alloc();

    snprintf(name, sizeof(name), "output stream %d:%d", ost->file_index, ost->index);
    ret = avfilter_graph_create_filter(&ofilter->filter,
-                                       avfilter_get_by_name("ffbuffersink"),
+                                       avfilter_get_by_name("buffersink"),
                                       name, NULL, NULL, fg->graph);
-    av_freep(&buffersink_params);

    if (ret < 0)
        return ret;
@@ -301,7 +359,7 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter,
        char args[255];
        AVFilterContext *filter;

-        snprintf(args, sizeof(args), "%d:%d:flags=0x%X",
+        snprintf(args, sizeof(args), "%d:%d:0x%X",
                 codec->width,
                 codec->height,
                 (unsigned)ost->sws_flags);
@@ -321,17 +379,18 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter,
        AVFilterContext *filter;
        snprintf(name, sizeof(name), "pixel format for output stream %d:%d",
                 ost->file_index, ost->index);
-        if ((ret = avfilter_graph_create_filter(&filter,
+        ret = avfilter_graph_create_filter(&filter,
                                                avfilter_get_by_name("format"),
                                                "format", pix_fmts, NULL,
-                                                fg->graph)) < 0)
+                                                fg->graph);
+        av_freep(&pix_fmts);
+        if (ret < 0)
            return ret;
        if ((ret = avfilter_link(last_filter, pad_idx, filter, 0)) < 0)
            return ret;

        last_filter = filter;
        pad_idx     = 0;
-        av_freep(&pix_fmts);
    }

    if (ost->frame_rate.num && 0) {
@@ -354,6 +413,11 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter,
        pad_idx = 0;
    }

+    ret = insert_trim(ost, &last_filter, &pad_idx);
+    if (ret < 0)
+        return ret;
+
+
    if ((ret = avfilter_link(last_filter, pad_idx, ofilter->filter, 0)) < 0)
        return ret;

@@ -365,22 +429,20 @@ static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter,
    OutputStream *ost = ofilter->ost;
    AVCodecContext *codec  = ost->st->codec;
    AVFilterContext *last_filter = out->filter_ctx;
+    OutputFile *of = output_files[ost->file_index];
    int pad_idx = out->pad_idx;
    char *sample_fmts, *sample_rates, *channel_layouts;
    char name[255];
    int ret;
-    AVABufferSinkParams *params = av_abuffersink_params_alloc();

-    if (!params)
-        return AVERROR(ENOMEM);
-    params->all_channel_counts = 1;
    snprintf(name, sizeof(name), "output stream %d:%d", ost->file_index, ost->index);
    ret = avfilter_graph_create_filter(&ofilter->filter,
-                                       avfilter_get_by_name("ffabuffersink"),
-                                       name, NULL, params, fg->graph);
-    av_freep(&params);
+                                       avfilter_get_by_name("abuffersink"),
+                                       name, NULL, NULL, fg->graph);
    if (ret < 0)
        return ret;
+    if ((ret = av_opt_set_int(ofilter->filter, "all_channel_counts", 1, AV_OPT_SEARCH_CHILDREN)) < 0)
+        return ret;

 #define AUTO_INSERT_FILTER(opt_name, filter_name, arg) do {                 \
    AVFilterContext *filt_ctx;                                              \
@@ -463,6 +525,24 @@ static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter,
        AUTO_INSERT_FILTER("-vol", "volume", args);
    }

+    if (ost->apad && of->shortest) {
+        char args[256];
+        int i;
+
+        for (i=0; i<of->ctx->nb_streams; i++)
+            if (of->ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
+                break;
+
+        if (i<of->ctx->nb_streams) {
+            snprintf(args, sizeof(args), "%s", ost->apad);
+            AUTO_INSERT_FILTER("-apad", "apad", args);
+        }
+    }
+
+    ret = insert_trim(ost, &last_filter, &pad_idx);
+    if (ret < 0)
+        return ret;
+
    if ((ret = avfilter_link(last_filter, pad_idx, ofilter->filter, 0)) < 0)
        return ret;

@@ -477,7 +557,7 @@ static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter,
    AVIOContext *pb;                                               \
                                                                   \
    if (avio_open_dyn_buf(&pb) < 0)                                \
-        exit(1);                                                   \
+        exit_program(1);                                           \
                                                                   \
    avio_printf(pb, "%s", ctx->filter->name);                      \
    if (nb_pads > 1)                                               \
@@ -501,9 +581,7 @@ int configure_output_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOu
 static int sub2video_prepare(InputStream *ist)
 {
    AVFormatContext *avf = input_files[ist->file_index]->ctx;
-    int i, ret, w, h;
-    uint8_t *image[4];
-    int linesize[4];
+    int i, w, h;

    /* Compute the size of the canvas for the subtitles stream.
       If the subtitles codec has set a size, use it. Otherwise use the
@@ -530,17 +608,9 @@ static int sub2video_prepare(InputStream *ist)
       palettes for all rectangles are identical or compatible */
    ist->resample_pix_fmt = ist->st->codec->pix_fmt = AV_PIX_FMT_RGB32;

-    ret = av_image_alloc(image, linesize, w, h, AV_PIX_FMT_RGB32, 32);
-    if (ret < 0)
-        return ret;
-    memset(image[0], 0, h * linesize[0]);
-    ist->sub2video.ref = avfilter_get_video_buffer_ref_from_arrays(
-            image, linesize, AV_PERM_READ | AV_PERM_PRESERVE,
-            w, h, AV_PIX_FMT_RGB32);
-    if (!ist->sub2video.ref) {
-        av_free(image[0]);
+    ist->sub2video.frame = av_frame_alloc();
+    if (!ist->sub2video.frame)
        return AVERROR(ENOMEM);
-    }
    return 0;
 }

@@ -552,22 +622,15 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
    InputStream *ist = ifilter->ist;
    AVRational tb = ist->framerate.num ? av_inv_q(ist->framerate) :
                                         ist->st->time_base;
-    AVRational fr = ist->framerate.num ? ist->framerate :
-                                         ist->st->r_frame_rate;
+    AVRational fr = ist->framerate;
    AVRational sar;
    AVBPrint args;
    char name[255];
    int pad_idx = in->pad_idx;
    int ret;

-    if (!ist->framerate.num && ist->st->codec->ticks_per_frame>1) {
-        AVRational codec_fr = av_inv_q(ist->st->codec->time_base);
-        AVRational   avg_fr = ist->st->avg_frame_rate;
-        codec_fr.den *= ist->st->codec->ticks_per_frame;
-        if (   codec_fr.num>0 && codec_fr.den>0 && av_q2d(codec_fr) < av_q2d(fr)*0.7
-            && fabs(1.0 - av_q2d(av_div_q(avg_fr, fr)))>0.1)
-            fr = codec_fr;
-    }
+    if (!fr.num)
+        fr = av_guess_frame_rate(input_files[ist->file_index]->ctx, ist->st, NULL);

    if (ist->st->codec->codec_type == AVMEDIA_TYPE_SUBTITLE) {
        ret = sub2video_prepare(ist);
@@ -614,6 +677,24 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
        pad_idx = 0;
    }

+    if (do_deinterlace) {
+        AVFilterContext *yadif;
+
+        snprintf(name, sizeof(name), "deinterlace input from stream %d:%d",
+                 ist->file_index, ist->st->index);
+        if ((ret = avfilter_graph_create_filter(&yadif,
+                                                avfilter_get_by_name("yadif"),
+                                                name, "", NULL,
+                                                fg->graph)) < 0)
+            return ret;
+
+        if ((ret = avfilter_link(yadif, 0, first_filter, pad_idx)) < 0)
+            return ret;
+
+        first_filter = yadif;
+        pad_idx = 0;
+    }
+
    if ((ret = avfilter_link(ifilter->filter, 0, first_filter, pad_idx)) < 0)
        return ret;
    return 0;
@@ -790,7 +871,7 @@ int configure_filtergraph(FilterGraph *fg)
        for (cur = outputs; cur;) {
            GROW_ARRAY(fg->outputs, fg->nb_outputs);
            if (!(fg->outputs[fg->nb_outputs - 1] = av_mallocz(sizeof(*fg->outputs[0]))))
-                exit(1);
+                exit_program(1);
            fg->outputs[fg->nb_outputs - 1]->graph   = fg;
            fg->outputs[fg->nb_outputs - 1]->out_tmp = cur;
            cur = cur->next;
--- a/ffmpeg_opt.c
+++ b/ffmpeg_opt.c
--- a/ffplay.c
+++ b/ffplay.c
--- a/ffprobe.c
+++ b/ffprobe.c
@@ -52,6 +52,7 @@ static int do_count_frames = 0;
 static int do_count_packets = 0;
 static int do_read_frames  = 0;
 static int do_read_packets = 0;
+static int do_show_chapters = 0;
 static int do_show_error   = 0;
 static int do_show_format  = 0;
 static int do_show_frames  = 0;
@@ -93,6 +94,9 @@ struct section {

 typedef enum {
    SECTION_ID_NONE = -1,
+    SECTION_ID_CHAPTER,
+    SECTION_ID_CHAPTER_TAGS,
+    SECTION_ID_CHAPTERS,
    SECTION_ID_ERROR,
    SECTION_ID_FORMAT,
    SECTION_ID_FORMAT_TAGS,
@@ -113,6 +117,9 @@ typedef enum {
 } SectionID;

 static struct section sections[] = {
+    [SECTION_ID_CHAPTERS] =           { SECTION_ID_CHAPTERS, "chapters", SECTION_FLAG_IS_ARRAY, { SECTION_ID_CHAPTER, -1 } },
+    [SECTION_ID_CHAPTER] =            { SECTION_ID_CHAPTER, "chapter", 0, { SECTION_ID_CHAPTER_TAGS, -1 } },
+    [SECTION_ID_CHAPTER_TAGS] =       { SECTION_ID_CHAPTER_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "chapter_tags" },
    [SECTION_ID_ERROR] =              { SECTION_ID_ERROR, "error", 0, { -1 } },
    [SECTION_ID_FORMAT] =             { SECTION_ID_FORMAT, "format", 0, { SECTION_ID_FORMAT_TAGS, -1 } },
    [SECTION_ID_FORMAT_TAGS] =        { SECTION_ID_FORMAT_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "format_tags" },
@@ -126,7 +133,7 @@ static struct section sections[] = {
    [SECTION_ID_PACKET] =             { SECTION_ID_PACKET, "packet", 0, { -1 } },
    [SECTION_ID_PROGRAM_VERSION] =    { SECTION_ID_PROGRAM_VERSION, "program_version", 0, { -1 } },
    [SECTION_ID_ROOT] =               { SECTION_ID_ROOT, "root", SECTION_FLAG_IS_WRAPPER,
-                                        { SECTION_ID_FORMAT, SECTION_ID_FRAMES, SECTION_ID_STREAMS, SECTION_ID_PACKETS,
+                                        { SECTION_ID_CHAPTERS, SECTION_ID_FORMAT, SECTION_ID_FRAMES, SECTION_ID_STREAMS, SECTION_ID_PACKETS,
                                          SECTION_ID_ERROR, SECTION_ID_PROGRAM_VERSION, SECTION_ID_LIBRARY_VERSIONS, -1} },
    [SECTION_ID_STREAMS] =            { SECTION_ID_STREAMS, "streams", SECTION_FLAG_IS_ARRAY, { SECTION_ID_STREAM, -1 } },
    [SECTION_ID_STREAM] =             { SECTION_ID_STREAM, "stream", 0, { SECTION_ID_STREAM_DISPOSITION, SECTION_ID_STREAM_TAGS, -1 } },
@@ -152,7 +159,7 @@ static uint64_t *nb_streams_packets;
 static uint64_t *nb_streams_frames;
 static int *selected_streams;

-static void exit_program(void)
+static void ffprobe_cleanup(int ret)
 {
    int i;
    for (i = 0; i < FF_ARRAY_ELEMS(sections); i++)
@@ -300,7 +307,7 @@ static int writer_open(WriterContext **wctx, const Writer *writer, const char *a
 {
    int i, ret = 0;

-    if (!(*wctx = av_malloc(sizeof(WriterContext)))) {
+    if (!(*wctx = av_mallocz(sizeof(WriterContext)))) {
        ret = AVERROR(ENOMEM);
        goto fail;
    }
@@ -1500,7 +1507,6 @@ static void show_frame(WriterContext *w, AVFrame *frame, AVStream *stream,
        print_int("interlaced_frame",       frame->interlaced_frame);
        print_int("top_field_first",        frame->top_field_first);
        print_int("repeat_pict",            frame->repeat_pict);
-        print_int("reference",              frame->reference);
        break;

    case AVMEDIA_TYPE_AUDIO:
@@ -1752,6 +1758,27 @@ static void show_streams(WriterContext *w, AVFormatContext *fmt_ctx)
    writer_print_section_footer(w);
 }

+static void show_chapters(WriterContext *w, AVFormatContext *fmt_ctx)
+{
+    int i;
+
+    writer_print_section_header(w, SECTION_ID_CHAPTERS);
+    for (i = 0; i < fmt_ctx->nb_chapters; i++) {
+        AVChapter *chapter = fmt_ctx->chapters[i];
+
+        writer_print_section_header(w, SECTION_ID_CHAPTER);
+        print_int("id", chapter->id);
+        print_q  ("time_base", chapter->time_base, '/');
+        print_int("start", chapter->start);
+        print_time("start_time", chapter->start, &chapter->time_base);
+        print_int("end", chapter->end);
+        print_time("end_time", chapter->end, &chapter->time_base);
+        show_tags(w, chapter->metadata, SECTION_ID_CHAPTER_TAGS);
+        writer_print_section_footer(w);
+    }
+    writer_print_section_footer(w);
+}
+
 static void show_format(WriterContext *w, AVFormatContext *fmt_ctx)
 {
    char val_str[128];
@@ -1793,9 +1820,10 @@ static void show_error(WriterContext *w, int err)

 static int open_input_file(AVFormatContext **fmt_ctx_ptr, const char *filename)
 {
-    int err, i;
+    int err, i, orig_nb_streams;
    AVFormatContext *fmt_ctx = NULL;
    AVDictionaryEntry *t;
+    AVDictionary **opts;

    if ((err = avformat_open_input(&fmt_ctx, filename,
                                   iformat, &format_opts)) < 0) {
@@ -1807,12 +1835,17 @@ static int open_input_file(AVFormatContext **fmt_ctx_ptr, const char *filename)
        return AVERROR_OPTION_NOT_FOUND;
    }

-
    /* fill the streams in the format context */
-    if ((err = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
+    opts = setup_find_stream_info_opts(fmt_ctx, codec_opts);
+    orig_nb_streams = fmt_ctx->nb_streams;
+
+    if ((err = avformat_find_stream_info(fmt_ctx, opts)) < 0) {
        print_error(filename, err);
        return err;
    }
+    for (i = 0; i < orig_nb_streams; i++)
+        av_dict_free(&opts[i]);
+    av_freep(&opts);

    av_dump_format(fmt_ctx, 0, filename, 0);

@@ -1829,9 +1862,18 @@ static int open_input_file(AVFormatContext **fmt_ctx_ptr, const char *filename)
            av_log(NULL, AV_LOG_ERROR,
                    "Unsupported codec with id %d for input stream %d\n",
                    stream->codec->codec_id, stream->index);
-        } else if (avcodec_open2(stream->codec, codec, NULL) < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Error while opening codec for input stream %d\n",
-                   stream->index);
+        } else {
+            AVDictionary *opts = filter_codec_opts(codec_opts, stream->codec->codec_id,
+                                                   fmt_ctx, stream, codec);
+            if (avcodec_open2(stream->codec, codec, &opts) < 0) {
+                av_log(NULL, AV_LOG_ERROR, "Error while opening codec for input stream %d\n",
+                       stream->index);
+            }
+            if ((t = av_dict_get(opts, "", NULL, AV_DICT_IGNORE_SUFFIX))) {
+                av_log(NULL, AV_LOG_ERROR, "Option %s for input stream %d not found\n",
+                       t->key, stream->index);
+                return AVERROR_OPTION_NOT_FOUND;
+            }
        }
    }

@@ -1862,50 +1904,55 @@ static int probe_file(WriterContext *wctx, const char *filename)
    do_read_packets = do_show_packets || do_count_packets;

    ret = open_input_file(&fmt_ctx, filename);
-    if (ret >= 0) {
-        nb_streams_frames  = av_calloc(fmt_ctx->nb_streams, sizeof(*nb_streams_frames));
-        nb_streams_packets = av_calloc(fmt_ctx->nb_streams, sizeof(*nb_streams_packets));
-        selected_streams   = av_calloc(fmt_ctx->nb_streams, sizeof(*selected_streams));
+    if (ret < 0)
+        return ret;

-        for (i = 0; i < fmt_ctx->nb_streams; i++) {
-            if (stream_specifier) {
-                ret = avformat_match_stream_specifier(fmt_ctx,
-                                                      fmt_ctx->streams[i],
-                                                      stream_specifier);
-                if (ret < 0)
-                    goto end;
-                else
-                    selected_streams[i] = ret;
-            } else {
-                selected_streams[i] = 1;
-            }
+    nb_streams_frames  = av_calloc(fmt_ctx->nb_streams, sizeof(*nb_streams_frames));
+    nb_streams_packets = av_calloc(fmt_ctx->nb_streams, sizeof(*nb_streams_packets));
+    selected_streams   = av_calloc(fmt_ctx->nb_streams, sizeof(*selected_streams));
+
+    for (i = 0; i < fmt_ctx->nb_streams; i++) {
+        if (stream_specifier) {
+            ret = avformat_match_stream_specifier(fmt_ctx,
+                                                  fmt_ctx->streams[i],
+                                                  stream_specifier);
+            if (ret < 0)
+                goto end;
+            else
+                selected_streams[i] = ret;
+            ret = 0;
+        } else {
+            selected_streams[i] = 1;
        }
-
-        if (do_read_frames || do_read_packets) {
-            if (do_show_frames && do_show_packets &&
-                wctx->writer->flags & WRITER_FLAG_PUT_PACKETS_AND_FRAMES_IN_SAME_CHAPTER)
-                section_id = SECTION_ID_PACKETS_AND_FRAMES;
-            else if (do_show_packets && !do_show_frames)
-                section_id = SECTION_ID_PACKETS;
-            else // (!do_show_packets && do_show_frames)
-                section_id = SECTION_ID_FRAMES;
-            if (do_show_frames || do_show_packets)
-                writer_print_section_header(wctx, section_id);
-            read_packets(wctx, fmt_ctx);
-            if (do_show_frames || do_show_packets)
-                writer_print_section_footer(wctx);
-        }
-        if (do_show_streams)
-            show_streams(wctx, fmt_ctx);
-        if (do_show_format)
-            show_format(wctx, fmt_ctx);
-
-    end:
-        close_input_file(&fmt_ctx);
-        av_freep(&nb_streams_frames);
-        av_freep(&nb_streams_packets);
-        av_freep(&selected_streams);
    }
+
+    if (do_read_frames || do_read_packets) {
+        if (do_show_frames && do_show_packets &&
+            wctx->writer->flags & WRITER_FLAG_PUT_PACKETS_AND_FRAMES_IN_SAME_CHAPTER)
+            section_id = SECTION_ID_PACKETS_AND_FRAMES;
+        else if (do_show_packets && !do_show_frames)
+            section_id = SECTION_ID_PACKETS;
+        else // (!do_show_packets && do_show_frames)
+            section_id = SECTION_ID_FRAMES;
+        if (do_show_frames || do_show_packets)
+            writer_print_section_header(wctx, section_id);
+        read_packets(wctx, fmt_ctx);
+        if (do_show_frames || do_show_packets)
+            writer_print_section_footer(wctx);
+    }
+    if (do_show_streams)
+        show_streams(wctx, fmt_ctx);
+    if (do_show_chapters)
+        show_chapters(wctx, fmt_ctx);
+    if (do_show_format)
+        show_format(wctx, fmt_ctx);
+
+end:
+    close_input_file(&fmt_ctx);
+    av_freep(&nb_streams_frames);
+    av_freep(&nb_streams_packets);
+    av_freep(&selected_streams);
+
    return ret;
 }

@@ -2076,7 +2123,7 @@ static void opt_input_file(void *optctx, const char *arg)
        av_log(NULL, AV_LOG_ERROR,
                "Argument '%s' provided as input filename, but '%s' was already specified.\n",
                arg, input_filename);
-        exit(1);
+        exit_program(1);
    }
    if (!strcmp(arg, "-"))
        arg = "pipe:";
@@ -2151,6 +2198,7 @@ static int opt_show_versions(const char *opt, const char *arg)
        return 0;                                                       \
    }

+DEFINE_OPT_SHOW_SECTION(chapters,         CHAPTERS);
 DEFINE_OPT_SHOW_SECTION(error,            ERROR);
 DEFINE_OPT_SHOW_SECTION(format,           FORMAT);
 DEFINE_OPT_SHOW_SECTION(frames,           FRAMES);
@@ -2185,6 +2233,7 @@ static const OptionDef real_options[] = {
      "show a set of specified entries", "entry_list" },
    { "show_packets", 0, {(void*)&opt_show_packets}, "show packets info" },
    { "show_streams", 0, {(void*)&opt_show_streams}, "show streams info" },
+    { "show_chapters", 0, {(void*)&opt_show_chapters}, "show chapters info" },
    { "count_frames", OPT_BOOL, {(void*)&do_count_frames}, "count the number of frames per stream" },
    { "count_packets", OPT_BOOL, {(void*)&do_count_packets}, "count the number of packets per stream" },
    { "show_program_version",  0, {(void*)&opt_show_program_version},  "show ffprobe version" },
@@ -2224,7 +2273,7 @@ int main(int argc, char **argv)
    int ret, i;

    av_log_set_flags(AV_LOG_SKIP_REPEATED);
-    atexit(exit_program);
+    register_exit(ffprobe_cleanup);

    options = real_options;
    parse_loglevel(argc, argv, options);
@@ -2239,6 +2288,7 @@ int main(int argc, char **argv)
    parse_options(NULL, argc, argv, options, opt_input_file);

    /* mark things to show, based on -show_entries */
+    SET_DO_SHOW(CHAPTERS, chapters);
    SET_DO_SHOW(ERROR, error);
    SET_DO_SHOW(FORMAT, format);
    SET_DO_SHOW(FRAMES, frames);
@@ -2284,7 +2334,7 @@ int main(int argc, char **argv)
            ffprobe_show_library_versions(wctx);

        if (!input_filename &&
-            ((do_show_format || do_show_streams || do_show_packets || do_show_error) ||
+            ((do_show_format || do_show_streams || do_show_chapters || do_show_packets || do_show_error) ||
             (!do_show_program_version && !do_show_library_versions))) {
            show_usage();
            av_log(NULL, AV_LOG_ERROR, "You have to specify one input file.\n");
@@ -2309,5 +2359,5 @@ end:

    avformat_network_deinit();

-    return ret;
+    return ret < 0;
 }
--- a/ffserver.c
+++ b/ffserver.c
@@ -308,7 +308,7 @@ static int rtp_new_av_stream(HTTPContext *c,

 static const char *my_program_name;

-static const char *config_filename = "/etc/ffserver.conf";
+static const char *config_filename;

 static int ffserver_debug;
 static int no_launch;
@@ -328,6 +328,14 @@ static AVLFG random_state;

 static FILE *logfile = NULL;

+static void htmlstrip(char *s) {
+    while (s && *s) {
+        s += strspn(s, "0123456789abcdefghijklmnopqrstuvwxyzABCDEFGHIJKLMNOPQRSTUVWXYZ,. ");
+        if (*s)
+            *s++ = '?';
+    }
+}
+
 static int64_t ffm_read_write_index(int fd)
 {
    uint8_t buf[8];
@@ -396,14 +404,14 @@ static int resolve_host(struct in_addr *sin_addr, const char *hostname)
    return 0;
 }

-static char *ctime1(char *buf2)
+static char *ctime1(char *buf2, int buf_size)
 {
    time_t ti;
    char *p;

    ti = time(NULL);
    p = ctime(&ti);
-    strcpy(buf2, p);
+    av_strlcpy(buf2, p, buf_size);
    p = buf2 + strlen(p) - 1;
    if (*p == '\n')
        *p = '\0';
@@ -416,7 +424,7 @@ static void http_vlog(const char *fmt, va_list vargs)
    if (logfile) {
        if (print_prefix) {
            char buf[32];
-            ctime1(buf);
+            ctime1(buf, sizeof(buf));
            fprintf(logfile, "%s ", buf);
        }
        print_prefix = strstr(fmt, "\n") != NULL;
@@ -1887,6 +1895,7 @@ static int http_parse_request(HTTPContext *c)
 send_error:
    c->http_error = 404;
    q = c->buffer;
+    htmlstrip(msg);
    snprintf(q, c->buffer_size,
                  "HTTP/1.0 404 Not Found\r\n"
                  "Content-type: text/html\r\n"
@@ -3974,6 +3983,7 @@ static void load_module(const char *filename)
                "%s: init function 'ffserver_module_init()' not found\n",
                filename);
        dlclose(dll);
+        return;
    }

    init_func();
@@ -4151,7 +4161,7 @@ static int parse_ffconfig(const char *filename)
        } else if (!av_strcasecmp(cmd, "MaxBandwidth")) {
            int64_t llval;
            get_arg(arg, sizeof(arg), &p);
-            llval = atoll(arg);
+            llval = strtoll(arg, NULL, 10);
            if (llval < 10 || llval > 10000000) {
                ERROR("Invalid MaxBandwidth: %s\n", arg);
            } else
@@ -4545,14 +4555,6 @@ static int parse_ffconfig(const char *filename)
                    ERROR("VideoQMin out of range\n");
                }
            }
-        } else if (!av_strcasecmp(cmd, "LumaElim")) {
-            get_arg(arg, sizeof(arg), &p);
-            if (stream)
-                video_enc.luma_elim_threshold = atoi(arg);
-        } else if (!av_strcasecmp(cmd, "ChromaElim")) {
-            get_arg(arg, sizeof(arg), &p);
-            if (stream)
-                video_enc.chroma_elim_threshold = atoi(arg);
        } else if (!av_strcasecmp(cmd, "LumiMask")) {
            get_arg(arg, sizeof(arg), &p);
            if (stream)
@@ -4724,6 +4726,9 @@ int main(int argc, char **argv)

    parse_options(NULL, argc, argv, options, NULL);

+    if (!config_filename)
+        config_filename = av_strdup("/etc/ffserver.conf");
+
    unsetenv("http_proxy");             /* Kill the http_proxy */

    av_lfg_init(&random_state, av_get_random_seed());
@@ -4736,6 +4741,7 @@ int main(int argc, char **argv)
        fprintf(stderr, "Incorrect config file - exiting.\n");
        exit(1);
    }
+    av_freep(&config_filename);

    /* open log file if needed */
    if (logfilename[0] != '\0') {
--- a/libavcodec/012v.c
+++ b/libavcodec/012v.c
@@ -26,18 +26,12 @@

 static av_cold int zero12v_decode_init(AVCodecContext *avctx)
 {
-    avctx->pix_fmt             = PIX_FMT_YUV422P16;
+    avctx->pix_fmt             = AV_PIX_FMT_YUV422P16;
    avctx->bits_per_raw_sample = 10;

-    avctx->coded_frame = avcodec_alloc_frame();
-    if (!avctx->coded_frame)
-        return AVERROR(ENOMEM);
-
    if (avctx->codec_tag == MKTAG('a', '1', '2', 'v'))
-        av_log_ask_for_sample(avctx, "Samples with actual transparency needed\n");
+        avpriv_request_sample(avctx, "transparency");

-    avctx->coded_frame->pict_type = AV_PICTURE_TYPE_I;
-    avctx->coded_frame->key_frame = 1;
    return 0;
 }

@@ -46,14 +40,11 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
 {
    int line = 0, ret;
    const int width = avctx->width;
-    AVFrame *pic = avctx->coded_frame;
+    AVFrame *pic = data;
    uint16_t *y, *u, *v;
    const uint8_t *line_end, *src = avpkt->data;
    int stride = avctx->width * 8 / 3;

-    if (pic->data[0])
-        avctx->release_buffer(avctx, pic);
-
    if (width == 1) {
        av_log(avctx, AV_LOG_ERROR, "Width 1 not supported.\n");
        return AVERROR_INVALIDDATA;
@@ -64,10 +55,12 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
        return AVERROR_INVALIDDATA;
    }

-    pic->reference = 0;
-    if ((ret = ff_get_buffer(avctx, pic)) < 0)
+    if ((ret = ff_get_buffer(avctx, pic, 0)) < 0)
        return ret;

+    pic->pict_type = AV_PICTURE_TYPE_I;
+    pic->key_frame = 1;
+
    y = (uint16_t *)pic->data[0];
    u = (uint16_t *)pic->data[1];
    v = (uint16_t *)pic->data[2];
@@ -145,27 +138,15 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
    }

    *got_frame = 1;
-    *(AVFrame*)data= *avctx->coded_frame;

    return avpkt->size;
 }

-static av_cold int zero12v_decode_close(AVCodecContext *avctx)
-{
-    AVFrame *pic = avctx->coded_frame;
-    if (pic->data[0])
-        avctx->release_buffer(avctx, pic);
-    av_freep(&avctx->coded_frame);
-
-    return 0;
-}
-
 AVCodec ff_zero12v_decoder = {
    .name           = "012v",
    .type           = AVMEDIA_TYPE_VIDEO,
    .id             = AV_CODEC_ID_012V,
    .init           = zero12v_decode_init,
-    .close          = zero12v_decode_close,
    .decode         = zero12v_decode_frame,
    .capabilities   = CODEC_CAP_DR1,
    .long_name      = NULL_IF_CONFIG_SMALL("Uncompressed 4:2:2 10-bit"),
--- a/libavcodec/4xm.c
+++ b/libavcodec/4xm.c
@@ -24,6 +24,8 @@
 * 4XM codec.
 */

+#include "libavutil/avassert.h"
+#include "libavutil/frame.h"
 #include "libavutil/intreadwrite.h"
 #include "avcodec.h"
 #include "bytestream.h"
@@ -31,7 +33,6 @@
 #include "get_bits.h"
 #include "internal.h"

-#include "libavutil/avassert.h"

 #define BLOCK_TYPE_VLC_BITS 5
 #define ACDC_VLC_BITS 9
@@ -255,15 +256,15 @@ static av_cold void init_vlcs(FourXContext *f)
    }
 }

-static void init_mv(FourXContext *f)
+static void init_mv(FourXContext *f, int linesize)
 {
    int i;

    for (i = 0; i < 256; i++) {
        if (f->version > 1)
-            f->mv[i] = mv[i][0] + mv[i][1] * f->current_picture->linesize[0] / 2;
+            f->mv[i] = mv[i][0] + mv[i][1] * linesize / 2;
        else
-            f->mv[i] = (i & 15) - 8 + ((i >> 4) - 8) * f->current_picture->linesize[0] / 2;
+            f->mv[i] = (i & 15) - 8 + ((i >> 4) - 8) * linesize / 2;
    }
 }

@@ -328,12 +329,12 @@ static inline void mcdc(uint16_t *dst, const uint16_t *src, int log2w,
        }
        break;
    default:
-        av_assert2(0);
+        av_assert0(0);
    }
 }

-static void decode_p_block(FourXContext *f, uint16_t *dst, uint16_t *src,
-                           int log2w, int log2h, int stride)
+static int decode_p_block(FourXContext *f, uint16_t *dst, uint16_t *src,
+                          int log2w, int log2h, int stride)
 {
    const int index = size2index[log2h][log2w];
    const int h     = 1 << log2h;
@@ -342,57 +343,30 @@ static void decode_p_block(FourXContext *f, uint16_t *dst, uint16_t *src,
                               BLOCK_TYPE_VLC_BITS, 1);
    uint16_t *start = (uint16_t *)f->last_picture->data[0];
    uint16_t *end   = start + stride * (f->avctx->height - h + 1) - (1 << log2w);
+    int ret;
+    int scale   = 1;
+    unsigned dc = 0;

-    av_assert2(code >= 0 && code <= 6);
+    av_assert0(code >= 0 && code <= 6 && log2w >= 0);

-    if (code == 0) {
-        if (bytestream2_get_bytes_left(&f->g) < 1) {
-            av_log(f->avctx, AV_LOG_ERROR, "bytestream overread\n");
-            return;
-        }
-        src += f->mv[bytestream2_get_byteu(&f->g)];
-        if (start > src || src > end) {
-            av_log(f->avctx, AV_LOG_ERROR, "mv out of pic\n");
-            return;
-        }
-        mcdc(dst, src, log2w, h, stride, 1, 0);
-    } else if (code == 1) {
+    if (code == 1) {
        log2h--;
-        decode_p_block(f, dst, src, log2w, log2h, stride);
-        decode_p_block(f, dst + (stride << log2h),
-                          src + (stride << log2h), log2w, log2h, stride);
+        if ((ret = decode_p_block(f, dst, src, log2w, log2h, stride)) < 0)
+            return ret;
+        return decode_p_block(f, dst + (stride << log2h),
+                              src + (stride << log2h),
+                              log2w, log2h, stride);
    } else if (code == 2) {
        log2w--;
-        decode_p_block(f, dst , src, log2w, log2h, stride);
-        decode_p_block(f, dst + (1 << log2w),
-                          src + (1 << log2w), log2w, log2h, stride);
-    } else if (code == 3 && f->version < 2) {
-        mcdc(dst, src, log2w, h, stride, 1, 0);
-    } else if (code == 4) {
-        if (bytestream2_get_bytes_left(&f->g) < 1) {
-            av_log(f->avctx, AV_LOG_ERROR, "bytestream overread\n");
-            return;
-        }
-        src += f->mv[bytestream2_get_byteu(&f->g)];
-        if (start > src || src > end) {
-            av_log(f->avctx, AV_LOG_ERROR, "mv out of pic\n");
-            return;
-        }
-        if (bytestream2_get_bytes_left(&f->g2) < 2){
-            av_log(f->avctx, AV_LOG_ERROR, "wordstream overread\n");
-            return;
-        }
-        mcdc(dst, src, log2w, h, stride, 1, bytestream2_get_le16u(&f->g2));
-    } else if (code == 5) {
-        if (bytestream2_get_bytes_left(&f->g2) < 2) {
-            av_log(f->avctx, AV_LOG_ERROR, "wordstream overread\n");
-            return;
-        }
-        mcdc(dst, src, log2w, h, stride, 0, bytestream2_get_le16u(&f->g2));
+        if ((ret = decode_p_block(f, dst , src, log2w, log2h, stride)) < 0)
+            return ret;
+        return decode_p_block(f, dst + (1 << log2w),
+                              src + (1 << log2w),
+                              log2w, log2h, stride);
    } else if (code == 6) {
        if (bytestream2_get_bytes_left(&f->g2) < 4) {
            av_log(f->avctx, AV_LOG_ERROR, "wordstream overread\n");
-            return;
+            return AVERROR_INVALIDDATA;
        }
        if (log2w) {
            dst[0]      = bytestream2_get_le16u(&f->g2);
@@ -401,24 +375,73 @@ static void decode_p_block(FourXContext *f, uint16_t *dst, uint16_t *src,
            dst[0]      = bytestream2_get_le16u(&f->g2);
            dst[stride] = bytestream2_get_le16u(&f->g2);
        }
+        return 0;
    }
+
+    if ((code&3)==0 && bytestream2_get_bytes_left(&f->g) < 1) {
+        av_log(f->avctx, AV_LOG_ERROR, "bytestream overread\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (code == 0) {
+        src  += f->mv[bytestream2_get_byte(&f->g)];
+    } else if (code == 3 && f->version >= 2) {
+        return 0;
+    } else if (code == 4) {
+        src  += f->mv[bytestream2_get_byte(&f->g)];
+        if (bytestream2_get_bytes_left(&f->g2) < 2){
+            av_log(f->avctx, AV_LOG_ERROR, "wordstream overread\n");
+            return AVERROR_INVALIDDATA;
+        }
+        dc    = bytestream2_get_le16(&f->g2);
+    } else if (code == 5) {
+        if (bytestream2_get_bytes_left(&f->g2) < 2){
+            av_log(f->avctx, AV_LOG_ERROR, "wordstream overread\n");
+            return AVERROR_INVALIDDATA;
+        }
+        av_assert0(start <= src && src <= end);
+        scale = 0;
+        dc    = bytestream2_get_le16(&f->g2);
+    }
+
+    if (start > src || src > end) {
+        av_log(f->avctx, AV_LOG_ERROR, "mv out of pic\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    mcdc(dst, src, log2w, h, stride, scale, dc);
+
+    return 0;
 }

-static int decode_p_frame(FourXContext *f, const uint8_t *buf, int length)
+static int decode_p_frame(FourXContext *f, AVFrame *frame,
+                          const uint8_t *buf, int length)
 {
    int x, y;
    const int width  = f->avctx->width;
    const int height = f->avctx->height;
-    uint16_t *src    = (uint16_t *)f->last_picture->data[0];
-    uint16_t *dst    = (uint16_t *)f->current_picture->data[0];
-    const int stride =             f->current_picture->linesize[0] >> 1;
+    uint16_t *dst    = (uint16_t *)frame->data[0];
+    const int stride =             frame->linesize[0] >> 1;
+    uint16_t *src;
    unsigned int bitstream_size, bytestream_size, wordstream_size, extra,
                 bytestream_offset, wordstream_offset;
+    int ret;
+
+    if (!f->last_picture->data[0]) {
+        if ((ret = ff_get_buffer(f->avctx, f->last_picture,
+                                 AV_GET_BUFFER_FLAG_REF)) < 0) {
+            return ret;
+        }
+        for (y=0; y<f->avctx->height; y++)
+            memset(f->last_picture->data[0] + y*f->last_picture->linesize[0], 0, 2*f->avctx->width);
+    }
+
+    src = (uint16_t *)f->last_picture->data[0];

    if (f->version > 1) {
        extra           = 20;
        if (length < extra)
-            return -1;
+            return AVERROR_INVALIDDATA;
        bitstream_size  = AV_RL32(buf + 8);
        wordstream_size = AV_RL32(buf + 12);
        bytestream_size = AV_RL32(buf + 16);
@@ -455,11 +478,12 @@ static int decode_p_frame(FourXContext *f, const uint8_t *buf, int length)
    bytestream2_init(&f->g, buf + bytestream_offset,
                     length - bytestream_offset);

-    init_mv(f);
+    init_mv(f, frame->linesize[0]);

    for (y = 0; y < height; y += 8) {
        for (x = 0; x < width; x += 8)
-            decode_p_block(f, dst + x, src + x, 3, 3, stride);
+            if ((ret = decode_p_block(f, dst + x, src + x, 3, 3, stride)) < 0)
+                return ret;
        src += 8 * stride;
        dst += 8 * stride;
    }
@@ -482,8 +506,10 @@ static int decode_i_block(FourXContext *f, int16_t *block)

    /* DC coef */
    val = get_vlc2(&f->pre_gb, f->pre_vlc.table, ACDC_VLC_BITS, 3);
-    if (val >> 4)
+    if (val >> 4) {
        av_log(f->avctx, AV_LOG_ERROR, "error dc run != 0\n");
+        return AVERROR_INVALIDDATA;
+    }

    if (val)
        val = get_xbits(&f->gb, val);
@@ -501,7 +527,12 @@ static int decode_i_block(FourXContext *f, int16_t *block)
        if (code == 0xf0) {
            i += 16;
        } else {
-            level = get_xbits(&f->gb, code & 0xf);
+            if (code & 0xf) {
+                level = get_xbits(&f->gb, code & 0xf);
+            } else {
+                av_log(f->avctx, AV_LOG_ERROR, "0 coeff\n");
+                return AVERROR_INVALIDDATA;
+            }
            i    += code >> 4;
            if (i >= 64) {
                av_log(f->avctx, AV_LOG_ERROR, "run %d oveflow\n", i);
@@ -519,12 +550,12 @@ static int decode_i_block(FourXContext *f, int16_t *block)
    return 0;
 }

-static inline void idct_put(FourXContext *f, int x, int y)
+static inline void idct_put(FourXContext *f, AVFrame *frame, int x, int y)
 {
    int16_t (*block)[64] = f->block;
-    int stride           = f->current_picture->linesize[0] >> 1;
+    int stride           = frame->linesize[0] >> 1;
    int i;
-    uint16_t *dst = ((uint16_t*)f->current_picture->data[0]) + y * stride + x;
+    uint16_t *dst = ((uint16_t*)frame->data[0]) + y * stride + x;

    for (i = 0; i < 4; i++) {
        block[i][0] += 0x80 * 8 * 8;
@@ -580,7 +611,8 @@ static int decode_i_mb(FourXContext *f)
 }

 static const uint8_t *read_huffman_tables(FourXContext *f,
-                                          const uint8_t * const buf, int buf_size)
+                                          const uint8_t * const buf,
+                                          int buf_size)
 {
    int frequency[512] = { 0 };
    uint8_t flag[512];
@@ -603,6 +635,7 @@ static const uint8_t *read_huffman_tables(FourXContext *f,
            av_log(f->avctx, AV_LOG_ERROR, "invalid data in read_huffman_tables\n");
            return NULL;
        }
+
        for (i = start; i <= end; i++)
            frequency[i] = *ptr++;
        start = *ptr++;
@@ -682,14 +715,14 @@ static int mix(int c0, int c1)
    return red / 3 * 1024 + green / 3 * 32 + blue / 3;
 }

-static int decode_i2_frame(FourXContext *f, const uint8_t *buf, int length)
+static int decode_i2_frame(FourXContext *f, AVFrame *frame, const uint8_t *buf, int length)
 {
    int x, y, x2, y2;
    const int width  = f->avctx->width;
    const int height = f->avctx->height;
    const int mbs    = (FFALIGN(width, 16) >> 4) * (FFALIGN(height, 16) >> 4);
-    uint16_t *dst    = (uint16_t*)f->current_picture->data[0];
-    const int stride =            f->current_picture->linesize[0]>>1;
+    uint16_t *dst    = (uint16_t*)frame->data[0];
+    const int stride =            frame->linesize[0]>>1;
    const uint8_t *buf_end = buf + length;
    GetByteContext g3;

@@ -709,9 +742,9 @@ static int decode_i2_frame(FourXContext *f, const uint8_t *buf, int length)
            color[1] = bytestream2_get_le16u(&g3);

            if (color[0] & 0x8000)
-                av_log(NULL, AV_LOG_ERROR, "unk bit 1\n");
+                av_log(f->avctx, AV_LOG_ERROR, "unk bit 1\n");
            if (color[1] & 0x8000)
-                av_log(NULL, AV_LOG_ERROR, "unk bit 2\n");
+                av_log(f->avctx, AV_LOG_ERROR, "unk bit 2\n");

            color[2] = mix(color[0], color[1]);
            color[3] = mix(color[1], color[0]);
@@ -731,7 +764,7 @@ static int decode_i2_frame(FourXContext *f, const uint8_t *buf, int length)
    return 0;
 }

-static int decode_i_frame(FourXContext *f, const uint8_t *buf, int length)
+static int decode_i_frame(FourXContext *f, AVFrame *frame, const uint8_t *buf, int length)
 {
    int x, y, ret;
    const int width  = f->avctx->width;
@@ -740,7 +773,10 @@ static int decode_i_frame(FourXContext *f, const uint8_t *buf, int length)
    unsigned int prestream_size;
    const uint8_t *prestream;

-    if (bitstream_size > (1<<26) || length < bitstream_size + 12) {
+    if (bitstream_size > (1 << 26))
+        return AVERROR_INVALIDDATA;
+
+    if (length < bitstream_size + 12) {
        av_log(f->avctx, AV_LOG_ERROR, "packet size too small\n");
        return AVERROR_INVALIDDATA;
    }
@@ -749,14 +785,13 @@ static int decode_i_frame(FourXContext *f, const uint8_t *buf, int length)
    prestream      =             buf + bitstream_size + 12;

    if (prestream_size + bitstream_size + 12 != length
-        || bitstream_size > (1 << 26)
        || prestream_size > (1 << 26)) {
        av_log(f->avctx, AV_LOG_ERROR, "size mismatch %d %d %d\n",
               prestream_size, bitstream_size, length);
        return AVERROR_INVALIDDATA;
    }

-    prestream = read_huffman_tables(f, prestream, buf + length - prestream);
+    prestream = read_huffman_tables(f, prestream, prestream_size);
    if (!prestream) {
        av_log(f->avctx, AV_LOG_ERROR, "Error reading Huffman tables.\n");
        return AVERROR_INVALIDDATA;
@@ -785,7 +820,7 @@ static int decode_i_frame(FourXContext *f, const uint8_t *buf, int length)
            if ((ret = decode_i_mb(f)) < 0)
                return ret;

-            idct_put(f, x, y);
+            idct_put(f, frame, x, y);
        }
    }

@@ -802,33 +837,38 @@ static int decode_frame(AVCodecContext *avctx, void *data,
    int buf_size          = avpkt->size;
    FourXContext *const f = avctx->priv_data;
    AVFrame *picture      = data;
-    AVFrame *p;
    int i, frame_4cc, frame_size, ret;

-    if (buf_size < 12)
+    if (buf_size < 20)
        return AVERROR_INVALIDDATA;
-    frame_4cc = AV_RL32(buf);
-    if (buf_size != AV_RL32(buf + 4) + 8 || buf_size < 20)
+
+    if (buf_size < AV_RL32(buf + 4) + 8) {
        av_log(f->avctx, AV_LOG_ERROR, "size mismatch %d %d\n",
               buf_size, AV_RL32(buf + 4));
+        return AVERROR_INVALIDDATA;
+    }
+
+    frame_4cc = AV_RL32(buf);

    if (frame_4cc == AV_RL32("cfrm")) {
        int free_index       = -1;
+        int id, whole_size;
        const int data_size  = buf_size - 20;
-        const int id         = AV_RL32(buf + 12);
-        const int whole_size = AV_RL32(buf + 16);
        CFrameBuffer *cfrm;

-        if (data_size < 0 || whole_size < 0) {
-            av_log(f->avctx, AV_LOG_ERROR, "sizes invalid\n");
-            return AVERROR_INVALIDDATA;
-        }
-
        if (f->version <= 1) {
            av_log(f->avctx, AV_LOG_ERROR, "cfrm in version %d\n", f->version);
            return AVERROR_INVALIDDATA;
        }

+        id         = AV_RL32(buf + 12);
+        whole_size = AV_RL32(buf + 16);
+
+        if (data_size < 0 || whole_size < 0) {
+            av_log(f->avctx, AV_LOG_ERROR, "sizes invalid\n");
+            return AVERROR_INVALIDDATA;
+        }
+
        for (i = 0; i < CFRAME_BUFFER_COUNT; i++)
            if (f->cfrm[i].id && f->cfrm[i].id < avctx->frame_number)
                av_log(f->avctx, AV_LOG_ERROR, "lost c frame %d\n",
@@ -869,6 +909,9 @@ static int decode_frame(AVCodecContext *avctx, void *data,
                av_log(f->avctx, AV_LOG_ERROR, "cframe id mismatch %d %d\n",
                       id, avctx->frame_number);

+            if (f->version <= 1)
+                return AVERROR_INVALIDDATA;
+
            cfrm->size = cfrm->id = 0;
            frame_4cc  = AV_RL32("pfrm");
        } else
@@ -880,43 +923,27 @@ static int decode_frame(AVCodecContext *avctx, void *data,

    FFSWAP(AVFrame*, f->current_picture, f->last_picture);

-    p                  = f->current_picture;
-    avctx->coded_frame = p;
-
    // alternatively we would have to use our own buffer management
    avctx->flags |= CODEC_FLAG_EMU_EDGE;

-    p->reference= 3;
-    if ((ret = avctx->reget_buffer(avctx, p)) < 0) {
-        av_log(avctx, AV_LOG_ERROR, "reget_buffer() failed\n");
+    if ((ret = ff_reget_buffer(avctx, f->current_picture)) < 0)
        return ret;
-    }

    if (frame_4cc == AV_RL32("ifr2")) {
-        p->pict_type= AV_PICTURE_TYPE_I;
-        if ((ret = decode_i2_frame(f, buf - 4, frame_size + 4)) < 0) {
+        f->current_picture->pict_type = AV_PICTURE_TYPE_I;
+        if ((ret = decode_i2_frame(f, f->current_picture, buf - 4, frame_size + 4)) < 0) {
            av_log(f->avctx, AV_LOG_ERROR, "decode i2 frame failed\n");
            return ret;
        }
    } else if (frame_4cc == AV_RL32("ifrm")) {
-        p->pict_type= AV_PICTURE_TYPE_I;
-        if ((ret = decode_i_frame(f, buf, frame_size)) < 0) {
+        f->current_picture->pict_type = AV_PICTURE_TYPE_I;
+        if ((ret = decode_i_frame(f, f->current_picture, buf, frame_size)) < 0) {
            av_log(f->avctx, AV_LOG_ERROR, "decode i frame failed\n");
            return ret;
        }
    } else if (frame_4cc == AV_RL32("pfrm") || frame_4cc == AV_RL32("pfr2")) {
-        if (!f->last_picture->data[0]) {
-            f->last_picture->reference = 3;
-            if ((ret = ff_get_buffer(avctx, f->last_picture)) < 0) {
-                av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
-                return ret;
-            }
-            for (i=0; i<avctx->height; i++)
-                memset(f->last_picture->data[0] + i*f->last_picture->linesize[0], 0, 2*avctx->width);
-        }
-
-        p->pict_type = AV_PICTURE_TYPE_P;
-        if ((ret = decode_p_frame(f, buf, frame_size)) < 0) {
+        f->current_picture->pict_type = AV_PICTURE_TYPE_P;
+        if ((ret = decode_p_frame(f, f->current_picture, buf, frame_size)) < 0) {
            av_log(f->avctx, AV_LOG_ERROR, "decode p frame failed\n");
            return ret;
        }
@@ -928,9 +955,10 @@ static int decode_frame(AVCodecContext *avctx, void *data,
               buf_size);
    }

-    p->key_frame = p->pict_type == AV_PICTURE_TYPE_I;
+    f->current_picture->key_frame = f->current_picture->pict_type == AV_PICTURE_TYPE_I;

-    *picture   = *p;
+    if ((ret = av_frame_ref(picture, f->current_picture)) < 0)
+        return ret;
    *got_frame = 1;

    emms_c();
@@ -961,13 +989,10 @@ static av_cold int decode_init(AVCodecContext *avctx)
    else
        avctx->pix_fmt = AV_PIX_FMT_BGR555;

-    f->current_picture = avcodec_alloc_frame();
-    f->last_picture    = avcodec_alloc_frame();
-    if (!f->current_picture || !f->last_picture) {
-        avcodec_free_frame(&f->current_picture);
-        avcodec_free_frame(&f->last_picture);
+    f->current_picture = av_frame_alloc();
+    f->last_picture    = av_frame_alloc();
+    if (!f->current_picture  || !f->last_picture)
        return AVERROR(ENOMEM);
-    }

    return 0;
 }
@@ -985,12 +1010,8 @@ static av_cold int decode_end(AVCodecContext *avctx)
        f->cfrm[i].allocated_size = 0;
    }
    ff_free_vlc(&f->pre_vlc);
-    if (f->current_picture->data[0])
-        avctx->release_buffer(avctx, f->current_picture);
-    if (f->last_picture->data[0])
-        avctx->release_buffer(avctx, f->last_picture);
-    avcodec_free_frame(&f->current_picture);
-    avcodec_free_frame(&f->last_picture);
+    av_frame_free(&f->current_picture);
+    av_frame_free(&f->last_picture);

    return 0;
 }
--- a/libavcodec/8bps.c
+++ b/libavcodec/8bps.c
@@ -46,7 +46,6 @@ static const enum AVPixelFormat pixfmt_rgb24[] = {

 typedef struct EightBpsContext {
    AVCodecContext *avctx;
-    AVFrame pic;

    unsigned char planes;
    unsigned char planemap[4];
@@ -57,6 +56,7 @@ typedef struct EightBpsContext {
 static int decode_frame(AVCodecContext *avctx, void *data,
                        int *got_frame, AVPacket *avpkt)
 {
+    AVFrame *frame = data;
    const uint8_t *buf = avpkt->data;
    int buf_size       = avpkt->size;
    EightBpsContext * const c = avctx->priv_data;
@@ -70,15 +70,8 @@ static int decode_frame(AVCodecContext *avctx, void *data,
    unsigned char *planemap = c->planemap;
    int ret;

-    if (c->pic.data[0])
-        avctx->release_buffer(avctx, &c->pic);
-
-    c->pic.reference    = 0;
-    c->pic.buffer_hints = FF_BUFFER_HINTS_VALID;
-    if ((ret = ff_get_buffer(avctx, &c->pic)) < 0) {
-        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
        return ret;
-    }

    /* Set data pointer after line lengths */
    dp = encoded + planes * (height << 1);
@@ -89,8 +82,8 @@ static int decode_frame(AVCodecContext *avctx, void *data,

        /* Decode a plane */
        for (row = 0; row < height; row++) {
-            pixptr = c->pic.data[0] + row * c->pic.linesize[0] + planemap[p];
-            pixptr_end = pixptr + c->pic.linesize[0];
+            pixptr = frame->data[0] + row * frame->linesize[0] + planemap[p];
+            pixptr_end = pixptr + frame->linesize[0];
            if(lp - encoded + row*2 + 1 >= buf_size)
                return -1;
            dlen = av_be2ne16(*(const unsigned short *)(lp + row * 2));
@@ -129,15 +122,14 @@ static int decode_frame(AVCodecContext *avctx, void *data,
                                                     AV_PKT_DATA_PALETTE,
                                                     NULL);
        if (pal) {
-            c->pic.palette_has_changed = 1;
+            frame->palette_has_changed = 1;
            memcpy(c->pal, pal, AVPALETTE_SIZE);
        }

-        memcpy (c->pic.data[1], c->pal, AVPALETTE_SIZE);
+        memcpy (frame->data[1], c->pal, AVPALETTE_SIZE);
    }

    *got_frame = 1;
-    *(AVFrame*)data = c->pic;

    /* always report that the buffer was completely consumed */
    return buf_size;
@@ -148,9 +140,7 @@ static av_cold int decode_init(AVCodecContext *avctx)
    EightBpsContext * const c = avctx->priv_data;

    c->avctx       = avctx;
-    c->pic.data[0] = NULL;

-    avcodec_get_frame_defaults(&c->pic);
    switch (avctx->bits_per_coded_sample) {
    case 8:
        avctx->pix_fmt = AV_PIX_FMT_PAL8;
@@ -188,23 +178,12 @@ static av_cold int decode_init(AVCodecContext *avctx)
    return 0;
 }

-static av_cold int decode_end(AVCodecContext *avctx)
-{
-    EightBpsContext * const c = avctx->priv_data;
-
-    if (c->pic.data[0])
-        avctx->release_buffer(avctx, &c->pic);
-
-    return 0;
-}
-
 AVCodec ff_eightbps_decoder = {
    .name           = "8bps",
    .type           = AVMEDIA_TYPE_VIDEO,
    .id             = AV_CODEC_ID_8BPS,
    .priv_data_size = sizeof(EightBpsContext),
    .init           = decode_init,
-    .close          = decode_end,
    .decode         = decode_frame,
    .capabilities   = CODEC_CAP_DR1,
    .long_name      = NULL_IF_CONFIG_SMALL("QuickTime 8BPS video"),
--- a/libavcodec/8svx.c
+++ b/libavcodec/8svx.c
@@ -136,10 +136,8 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,

    /* get output buffer */
    frame->nb_samples = buf_size * 2;
-    if ((ret = ff_get_buffer(avctx, frame)) < 0) {
-        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
        return ret;
-    }

    for (ch = 0; ch < avctx->channels; ch++) {
        delta_decode(frame->data[ch], &esc->data[ch][esc->data_idx],
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -20,11 +20,8 @@ OBJS = allcodecs.o                                                      \
       bitstream.o                                                      \
       bitstream_filter.o                                               \
       codec_desc.o                                                     \
-       dsputil.o                                                        \
-       faanidct.o                                                       \
       fmtconvert.o                                                     \
       imgconvert.o                                                     \
-       jrevdct.o                                                        \
       mathtables.o                                                     \
       options.o                                                        \
       parser.o                                                         \
@@ -32,7 +29,6 @@ OBJS = allcodecs.o                                                      \
       rawdec.o                                                         \
       resample.o                                                       \
       resample2.o                                                      \
-       simple_idct.o                                                    \
       utils.o                                                          \

 # parts needed for many different codecs
@@ -42,6 +38,8 @@ OBJS-$(CONFIG_AUDIO_FRAME_QUEUE)       += audio_frame_queue.o
 OBJS-$(CONFIG_CRYSTALHD)               += crystalhd.o
 OBJS-$(CONFIG_DCT)                     += dct.o dct32_fixed.o dct32_float.o
 OBJS-$(CONFIG_DXVA2)                   += dxva2.o
+OBJS-$(CONFIG_DSPUTIL)                 += dsputil.o faanidct.o          \
+                                          simple_idct.o jrevdct.o
 OBJS-$(CONFIG_ENCODERS)                += faandct.o jfdctfst.o jfdctint.o
 OBJS-$(CONFIG_ERROR_RESILIENCE)        += error_resilience.o
 FFT-OBJS-$(CONFIG_HARDCODED_TABLES)    += cos_tables.o cos_fixed_tables.o
@@ -52,6 +50,7 @@ OBJS-$(CONFIG_H264CHROMA)              += h264chroma.o
 OBJS-$(CONFIG_H264DSP)                 += h264dsp.o h264idct.o
 OBJS-$(CONFIG_H264PRED)                += h264pred.o
 OBJS-$(CONFIG_H264QPEL)                += h264qpel.o
+OBJS-$(CONFIG_HPELDSP)                 += hpeldsp.o
 OBJS-$(CONFIG_HUFFMAN)                 += huffman.o
 OBJS-$(CONFIG_LIBXVID)                 += libxvid_rc.o
 OBJS-$(CONFIG_LPC)                     += lpc.o
@@ -76,7 +75,7 @@ OBJS-$(CONFIG_VDPAU)                   += vdpau.o
 OBJS-$(CONFIG_VIDEODSP)                += videodsp.o
 OBJS-$(CONFIG_VP3DSP)                  += vp3dsp.o

-# decoders/encoders/hardware accelerators
+# decoders/encoders
 OBJS-$(CONFIG_ZERO12V_DECODER)         += 012v.o
 OBJS-$(CONFIG_A64MULTI_ENCODER)        += a64multienc.o elbg.o
 OBJS-$(CONFIG_A64MULTI5_ENCODER)       += a64multienc.o elbg.o
@@ -92,6 +91,7 @@ OBJS-$(CONFIG_AC3_DECODER)             += ac3dec.o ac3dec_data.o ac3.o kbdwin.o
 OBJS-$(CONFIG_AC3_ENCODER)             += ac3enc_float.o ac3enc.o ac3tab.o \
                                          ac3.o kbdwin.o
 OBJS-$(CONFIG_AC3_FIXED_ENCODER)       += ac3enc_fixed.o ac3enc.o ac3tab.o ac3.o
+OBJS-$(CONFIG_AIC_DECODER)             += aic.o
 OBJS-$(CONFIG_ALAC_DECODER)            += alac.o alac_data.o
 OBJS-$(CONFIG_ALAC_ENCODER)            += alacenc.o alac_data.o
 OBJS-$(CONFIG_ALS_DECODER)             += alsdec.o bgmc.o mpeg4audio.o
@@ -111,6 +111,8 @@ OBJS-$(CONFIG_AMV_ENCODER)             += mjpegenc.o mjpeg.o           \
 OBJS-$(CONFIG_ANM_DECODER)             += anm.o
 OBJS-$(CONFIG_ANSI_DECODER)            += ansi.o cga_data.o
 OBJS-$(CONFIG_APE_DECODER)             += apedec.o
+OBJS-$(CONFIG_SSA_DECODER)             += assdec.o ass.o ass_split.o
+OBJS-$(CONFIG_SSA_ENCODER)             += assenc.o ass.o
 OBJS-$(CONFIG_ASS_DECODER)             += assdec.o ass.o ass_split.o
 OBJS-$(CONFIG_ASS_ENCODER)             += assenc.o ass.o
 OBJS-$(CONFIG_ASV1_DECODER)            += asvdec.o asv.o mpeg12data.o
@@ -156,7 +158,7 @@ OBJS-$(CONFIG_CPIA_DECODER)            += cpia.o
 OBJS-$(CONFIG_CSCD_DECODER)            += cscd.o
 OBJS-$(CONFIG_CYUV_DECODER)            += cyuv.o
 OBJS-$(CONFIG_DCA_DECODER)             += dcadec.o dca.o dcadsp.o      \
-                                          dca_parser.o synth_filter.o
+                                          synth_filter.o
 OBJS-$(CONFIG_DCA_ENCODER)             += dcaenc.o dca.o
 OBJS-$(CONFIG_DIRAC_DECODER)           += diracdec.o dirac.o diracdsp.o \
                                          dirac_arith.o mpeg12data.o dirac_dwt.o
@@ -182,8 +184,8 @@ OBJS-$(CONFIG_EAMAD_DECODER)           += eamad.o eaidct.o mpeg12.o \
                                          mpeg12data.o
 OBJS-$(CONFIG_EATGQ_DECODER)           += eatgq.o eaidct.o
 OBJS-$(CONFIG_EATGV_DECODER)           += eatgv.o
-OBJS-$(CONFIG_EATQI_DECODER)           += eatqi.o eaidct.o mpeg12.o \
-                                          mpeg12data.o
+OBJS-$(CONFIG_EATQI_DECODER)           += eatqi.o eaidct.o mpeg12dec.o  \
+                                          mpeg12.o mpeg12data.o
 OBJS-$(CONFIG_EIGHTBPS_DECODER)        += 8bps.o
 OBJS-$(CONFIG_EIGHTSVX_EXP_DECODER)    += 8svx.o
 OBJS-$(CONFIG_EIGHTSVX_FIB_DECODER)    += 8svx.o
@@ -206,6 +208,7 @@ OBJS-$(CONFIG_FLIC_DECODER)            += flicvideo.o
 OBJS-$(CONFIG_FOURXM_DECODER)          += 4xm.o
 OBJS-$(CONFIG_FRAPS_DECODER)           += fraps.o
 OBJS-$(CONFIG_FRWU_DECODER)            += frwu.o
+OBJS-$(CONFIG_G2M_DECODER)             += g2meet.o mjpeg.o
 OBJS-$(CONFIG_G723_1_DECODER)          += g723_1.o acelp_vectors.o \
                                          celp_filters.o celp_math.o
 OBJS-$(CONFIG_G723_1_ENCODER)          += g723_1.o acelp_vectors.o celp_math.o
@@ -214,24 +217,18 @@ OBJS-$(CONFIG_GIF_DECODER)             += gifdec.o lzw.o
 OBJS-$(CONFIG_GIF_ENCODER)             += gif.o lzwenc.o
 OBJS-$(CONFIG_GSM_DECODER)             += gsmdec.o gsmdec_data.o msgsmdec.o
 OBJS-$(CONFIG_GSM_MS_DECODER)          += gsmdec.o gsmdec_data.o msgsmdec.o
-OBJS-$(CONFIG_H261_DECODER)            += h261dec.o h261.o h261data.o
-OBJS-$(CONFIG_H261_ENCODER)            += h261enc.o h261.o h261data.o
+OBJS-$(CONFIG_H261_DECODER)            += h261dec.o h261data.o h261.o
+OBJS-$(CONFIG_H261_ENCODER)            += h261enc.o h261data.o h261.o
 OBJS-$(CONFIG_H263_DECODER)            += h263dec.o h263.o ituh263dec.o        \
                                          mpeg4video.o mpeg4videodec.o flvdec.o\
                                          intelh263dec.o
-OBJS-$(CONFIG_H263_VAAPI_HWACCEL)      += vaapi_mpeg4.o
-OBJS-$(CONFIG_H263_VDPAU_HWACCEL)      += vdpau_mpeg4.o
 OBJS-$(CONFIG_H263_ENCODER)            += mpeg4videoenc.o mpeg4video.o  \
                                          h263.o ituh263enc.o flvenc.o
 OBJS-$(CONFIG_H264_DECODER)            += h264.o                               \
                                          h264_loopfilter.o h264_direct.o      \
                                          cabac.o h264_sei.o h264_ps.o         \
                                          h264_refs.o h264_cavlc.o h264_cabac.o
-OBJS-$(CONFIG_H264_DXVA2_HWACCEL)      += dxva2_h264.o
-OBJS-$(CONFIG_H264_VAAPI_HWACCEL)      += vaapi_h264.o
-OBJS-$(CONFIG_H264_VDA_HWACCEL)        += vda_h264.o
 OBJS-$(CONFIG_H264_VDA_DECODER)        += vda_h264_dec.o
-OBJS-$(CONFIG_H264_VDPAU_HWACCEL)      += vdpau_h264.o
 OBJS-$(CONFIG_HUFFYUV_DECODER)         += huffyuv.o huffyuvdec.o
 OBJS-$(CONFIG_HUFFYUV_ENCODER)         += huffyuv.o huffyuvenc.o
 OBJS-$(CONFIG_IAC_DECODER)             += imc.o
@@ -247,8 +244,10 @@ OBJS-$(CONFIG_INDEO5_DECODER)          += indeo5.o ivi_common.o ivi_dsp.o
 OBJS-$(CONFIG_INTERPLAY_DPCM_DECODER)  += dpcm.o
 OBJS-$(CONFIG_INTERPLAY_VIDEO_DECODER) += interplayvideo.o
 OBJS-$(CONFIG_JACOSUB_DECODER)         += jacosubdec.o ass.o
-OBJS-$(CONFIG_JPEG2000_DECODER)        += j2kdec.o mqcdec.o mqc.o j2k.o j2k_dwt.o
-OBJS-$(CONFIG_JPEG2000_ENCODER)        += j2kenc.o mqcenc.o mqc.o j2k.o j2k_dwt.o
+OBJS-$(CONFIG_JPEG2000_ENCODER)        += j2kenc.o mqcenc.o mqc.o jpeg2000.o \
+                                          jpeg2000dwt.o
+OBJS-$(CONFIG_JPEG2000_DECODER)        += jpeg2000dec.o jpeg2000.o      \
+                                          jpeg2000dwt.o mqcdec.o mqc.o
 OBJS-$(CONFIG_JPEGLS_DECODER)          += jpeglsdec.o jpegls.o \
                                          mjpegdec.o mjpeg.o
 OBJS-$(CONFIG_JPEGLS_ENCODER)          += jpeglsenc.o jpegls.o
@@ -288,30 +287,23 @@ OBJS-$(CONFIG_MPC8_DECODER)            += mpc8.o mpc.o
 OBJS-$(CONFIG_MPEGVIDEO_DECODER)       += mpeg12.o mpeg12data.o \
                                          mpegvideo.o error_resilience.o
 OBJS-$(CONFIG_MPEG_XVMC_DECODER)       += mpegvideo_xvmc.o
-OBJS-$(CONFIG_MPEG1_VDPAU_HWACCEL)     += vdpau_mpeg12.o
-OBJS-$(CONFIG_MPEG1VIDEO_DECODER)      += mpeg12.o mpeg12data.o
-OBJS-$(CONFIG_MPEG1VIDEO_ENCODER)      += mpeg12enc.o mpeg12.o          \
-                                          timecode.o
-OBJS-$(CONFIG_MPEG2_DXVA2_HWACCEL)     += dxva2_mpeg2.o
-OBJS-$(CONFIG_MPEG2_VAAPI_HWACCEL)     += vaapi_mpeg2.o
-OBJS-$(CONFIG_MPEG2_VDPAU_HWACCEL)     += vdpau_mpeg12.o
-OBJS-$(CONFIG_MPEG2VIDEO_DECODER)      += mpeg12.o mpeg12data.o
+OBJS-$(CONFIG_MPEG1VIDEO_DECODER)      += mpeg12dec.o mpeg12.o mpeg12data.o
+OBJS-$(CONFIG_MPEG1VIDEO_ENCODER)      += mpeg12enc.o mpeg12.o
+OBJS-$(CONFIG_MPEG2VIDEO_DECODER)      += mpeg12dec.o mpeg12.o mpeg12data.o
 OBJS-$(CONFIG_MPEG2VIDEO_ENCODER)      += mpeg12enc.o mpeg12.o          \
                                          timecode.o
-OBJS-$(CONFIG_MPEG4_VAAPI_HWACCEL)     += vaapi_mpeg4.o
-OBJS-$(CONFIG_MPEG4_VDPAU_HWACCEL)     += vdpau_mpeg4.o
 OBJS-$(CONFIG_MPL2_DECODER)            += mpl2dec.o ass.o
-OBJS-$(CONFIG_MSMPEG4V1_DECODER)       += msmpeg4.o msmpeg4data.o
-OBJS-$(CONFIG_MSMPEG4V2_DECODER)       += msmpeg4.o msmpeg4data.o h263dec.o \
-                                          h263.o ituh263dec.o mpeg4videodec.o
+OBJS-$(CONFIG_MSMPEG4V1_DECODER)       += msmpeg4dec.o msmpeg4.o msmpeg4data.o
+OBJS-$(CONFIG_MSMPEG4V2_DECODER)       += msmpeg4dec.o msmpeg4.o msmpeg4data.o \
+                                          h263dec.o h263.o ituh263dec.o \
+                                          mpeg4videodec.o
 OBJS-$(CONFIG_MSMPEG4V2_ENCODER)       += msmpeg4.o msmpeg4enc.o msmpeg4data.o \
-                                          h263dec.o h263.o ituh263dec.o        \
+                                          h263.o
+OBJS-$(CONFIG_MSMPEG4V3_DECODER)       += msmpeg4dec.o msmpeg4.o msmpeg4data.o \
+                                          h263dec.o h263.o ituh263dec.o \
                                          mpeg4videodec.o
-OBJS-$(CONFIG_MSMPEG4V3_DECODER)       += msmpeg4.o msmpeg4data.o h263dec.o \
-                                          h263.o ituh263dec.o mpeg4videodec.o
 OBJS-$(CONFIG_MSMPEG4V3_ENCODER)       += msmpeg4.o msmpeg4enc.o msmpeg4data.o \
-                                          h263dec.o h263.o ituh263dec.o        \
-                                          mpeg4videodec.o
+                                          h263.o
 OBJS-$(CONFIG_MSRLE_DECODER)           += msrle.o msrledec.o
 OBJS-$(CONFIG_MSA1_DECODER)            += mss3.o mss34dsp.o
 OBJS-$(CONFIG_MSS1_DECODER)            += mss1.o mss12.o
@@ -348,8 +340,8 @@ OBJS-$(CONFIG_PPM_ENCODER)             += pnmenc.o pnm.o
 OBJS-$(CONFIG_PRORES_DECODER)          += proresdec2.o proresdsp.o
 OBJS-$(CONFIG_PRORES_LGPL_DECODER)     += proresdec_lgpl.o proresdsp.o proresdata.o
 OBJS-$(CONFIG_PRORES_ENCODER)          += proresenc_anatoliy.o
-OBJS-$(CONFIG_PRORES_ANATOLIY_ENCODER) += proresenc_anatoliy.o
-OBJS-$(CONFIG_PRORES_KOSTYA_ENCODER)   += proresenc_kostya.o proresdata.o proresdsp.o
+OBJS-$(CONFIG_PRORES_AW_ENCODER)       += proresenc_anatoliy.o
+OBJS-$(CONFIG_PRORES_KS_ENCODER)       += proresenc_kostya.o proresdata.o proresdsp.o
 OBJS-$(CONFIG_PTX_DECODER)             += ptx.o
 OBJS-$(CONFIG_QCELP_DECODER)           += qcelpdec.o                     \
                                          celp_filters.o acelp_vectors.o \
@@ -384,6 +376,7 @@ OBJS-$(CONFIG_RV30_DECODER)            += rv30.o rv34.o rv30dsp.o rv34dsp.o
 OBJS-$(CONFIG_RV40_DECODER)            += rv40.o rv34.o rv34dsp.o rv40dsp.o
 OBJS-$(CONFIG_SAMI_DECODER)            += samidec.o ass.o
 OBJS-$(CONFIG_S302M_DECODER)           += s302m.o
+OBJS-$(CONFIG_S302M_ENCODER)           += s302menc.o
 OBJS-$(CONFIG_SANM_DECODER)            += sanm.o
 OBJS-$(CONFIG_SGI_DECODER)             += sgidec.o
 OBJS-$(CONFIG_SGI_ENCODER)             += sgienc.o rle.o
@@ -396,6 +389,7 @@ OBJS-$(CONFIG_SIPR_DECODER)            += sipr.o acelp_pitch_delay.o \
 OBJS-$(CONFIG_SMACKAUD_DECODER)        += smacker.o
 OBJS-$(CONFIG_SMACKER_DECODER)         += smacker.o
 OBJS-$(CONFIG_SMC_DECODER)             += smc.o
+OBJS-$(CONFIG_SMVJPEG_DECODER)         += smvjpegdec.o
 OBJS-$(CONFIG_SNOW_DECODER)            += snowdec.o snow.o snow_dwt.o
 OBJS-$(CONFIG_SNOW_ENCODER)            += snowenc.o snow.o snow_dwt.o             \
                                          h263.o ituh263enc.o
@@ -436,7 +430,8 @@ OBJS-$(CONFIG_TRUEMOTION2_DECODER)     += truemotion2.o
 OBJS-$(CONFIG_TRUESPEECH_DECODER)      += truespeech.o
 OBJS-$(CONFIG_TSCC_DECODER)            += tscc.o msrledec.o
 OBJS-$(CONFIG_TSCC2_DECODER)           += tscc2.o
-OBJS-$(CONFIG_TTA_DECODER)             += tta.o
+OBJS-$(CONFIG_TTA_DECODER)             += tta.o ttadata.o
+OBJS-$(CONFIG_TTA_ENCODER)             += ttaenc.o ttadata.o
 OBJS-$(CONFIG_TWINVQ_DECODER)          += twinvq.o
 OBJS-$(CONFIG_TXD_DECODER)             += txd.o s3tc.o
 OBJS-$(CONFIG_ULTI_DECODER)            += ulti.o
@@ -454,11 +449,8 @@ OBJS-$(CONFIG_V210X_DECODER)           += v210x.o
 OBJS-$(CONFIG_VB_DECODER)              += vb.o
 OBJS-$(CONFIG_VBLE_DECODER)            += vble.o
 OBJS-$(CONFIG_VC1_DECODER)             += vc1dec.o vc1.o vc1data.o vc1dsp.o \
-                                          msmpeg4.o msmpeg4data.o           \
-                                          intrax8.o intrax8dsp.o
-OBJS-$(CONFIG_VC1_DXVA2_HWACCEL)       += dxva2_vc1.o
-OBJS-$(CONFIG_VC1_VAAPI_HWACCEL)       += vaapi_vc1.o
-OBJS-$(CONFIG_VC1_VDPAU_HWACCEL)       += vdpau_vc1.o
+                                          msmpeg4dec.o msmpeg4.o msmpeg4data.o \
+                                          intrax8.o intrax8dsp.o wmv2dsp.o
 OBJS-$(CONFIG_VCR1_DECODER)            += vcr1.o
 OBJS-$(CONFIG_VMDAUDIO_DECODER)        += vmdav.o
 OBJS-$(CONFIG_VMDVIDEO_DECODER)        += vmdav.o
@@ -476,6 +468,7 @@ OBJS-$(CONFIG_VP8_DECODER)             += vp8.o vp8dsp.o vp56rac.o
 OBJS-$(CONFIG_VPLAYER_DECODER)         += textdec.o ass.o
 OBJS-$(CONFIG_VQA_DECODER)             += vqavideo.o
 OBJS-$(CONFIG_WAVPACK_DECODER)         += wavpack.o
+OBJS-$(CONFIG_WEBP_DECODER)            += vp8.o vp8dsp.o vp56rac.o
 OBJS-$(CONFIG_WEBVTT_DECODER)          += webvttdec.o
 OBJS-$(CONFIG_WMALOSSLESS_DECODER)     += wmalosslessdec.o wma_common.o
 OBJS-$(CONFIG_WMAPRO_DECODER)          += wmaprodec.o wma.o wma_common.o
@@ -486,13 +479,12 @@ OBJS-$(CONFIG_WMAV2_ENCODER)           += wmaenc.o wma.o wma_common.o aactab.o
 OBJS-$(CONFIG_WMAVOICE_DECODER)        += wmavoice.o \
                                          celp_filters.o \
                                          acelp_vectors.o acelp_filters.o
-OBJS-$(CONFIG_WMV1_DECODER)            += msmpeg4.o msmpeg4data.o
+OBJS-$(CONFIG_WMV1_DECODER)            += msmpeg4dec.o msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_WMV2_DECODER)            += wmv2dec.o wmv2.o wmv2dsp.o \
-                                          msmpeg4.o msmpeg4data.o \
+                                          msmpeg4dec.o msmpeg4.o msmpeg4data.o \
                                          intrax8.o intrax8dsp.o
 OBJS-$(CONFIG_WMV2_ENCODER)            += wmv2enc.o wmv2.o wmv2dsp.o \
-                                          msmpeg4.o msmpeg4enc.o msmpeg4data.o \
-                                          mpeg4videodec.o ituh263dec.o h263dec.o
+                                          msmpeg4.o msmpeg4enc.o msmpeg4data.o
 OBJS-$(CONFIG_WNV1_DECODER)            += wnv1.o
 OBJS-$(CONFIG_WS_SND1_DECODER)         += ws-snd1.o
 OBJS-$(CONFIG_XAN_DPCM_DECODER)        += dpcm.o
@@ -582,6 +574,7 @@ OBJS-$(CONFIG_ADPCM_ADX_DECODER)          += adxdec.o adx.o
 OBJS-$(CONFIG_ADPCM_ADX_ENCODER)          += adxenc.o adx.o
 OBJS-$(CONFIG_ADPCM_AFC_DECODER)          += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_CT_DECODER)           += adpcm.o adpcm_data.o
+OBJS-$(CONFIG_ADPCM_DTK_DECODER)          += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_EA_DECODER)           += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_EA_MAXIS_XA_DECODER)  += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_EA_R1_DECODER)        += adpcm.o adpcm_data.o
@@ -602,6 +595,7 @@ OBJS-$(CONFIG_ADPCM_IMA_ISS_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_OKI_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_QT_DECODER)       += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_QT_ENCODER)       += adpcmenc.o adpcm_data.o
+OBJS-$(CONFIG_ADPCM_IMA_RAD_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_SMJPEG_DECODER)   += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_WAV_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_WAV_ENCODER)      += adpcmenc.o adpcm_data.o
@@ -619,6 +613,23 @@ OBJS-$(CONFIG_ADPCM_YAMAHA_DECODER)       += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER)       += adpcmenc.o adpcm_data.o
 OBJS-$(CONFIG_VIMA_DECODER)               += vima.o adpcm_data.o

+# hardware accelerators
+OBJS-$(CONFIG_H263_VAAPI_HWACCEL)         += vaapi_mpeg4.o
+OBJS-$(CONFIG_H263_VDPAU_HWACCEL)         += vdpau_mpeg4.o
+OBJS-$(CONFIG_H264_DXVA2_HWACCEL)         += dxva2_h264.o
+OBJS-$(CONFIG_H264_VAAPI_HWACCEL)         += vaapi_h264.o
+OBJS-$(CONFIG_H264_VDA_HWACCEL)           += vda_h264.o
+OBJS-$(CONFIG_H264_VDPAU_HWACCEL)         += vdpau_h264.o
+OBJS-$(CONFIG_MPEG1_VDPAU_HWACCEL)        += vdpau_mpeg12.o
+OBJS-$(CONFIG_MPEG2_DXVA2_HWACCEL)        += dxva2_mpeg2.o
+OBJS-$(CONFIG_MPEG2_VAAPI_HWACCEL)        += vaapi_mpeg2.o
+OBJS-$(CONFIG_MPEG2_VDPAU_HWACCEL)        += vdpau_mpeg12.o
+OBJS-$(CONFIG_MPEG4_VAAPI_HWACCEL)        += vaapi_mpeg4.o
+OBJS-$(CONFIG_MPEG4_VDPAU_HWACCEL)        += vdpau_mpeg4.o
+OBJS-$(CONFIG_VC1_DXVA2_HWACCEL)          += dxva2_vc1.o
+OBJS-$(CONFIG_VC1_VAAPI_HWACCEL)          += vaapi_vc1.o
+OBJS-$(CONFIG_VC1_VDPAU_HWACCEL)          += vdpau_vc1.o
+
 # libavformat dependencies
 OBJS-$(CONFIG_ADTS_MUXER)              += mpeg4audio.o
 OBJS-$(CONFIG_ADX_DEMUXER)             += adx.o
@@ -687,6 +698,7 @@ OBJS-$(CONFIG_LIBSCHROEDINGER_DECODER)    += libschroedingerdec.o \
                                             libschroedinger.o
 OBJS-$(CONFIG_LIBSCHROEDINGER_ENCODER)    += libschroedingerenc.o \
                                             libschroedinger.o
+OBJS-$(CONFIG_LIBSHINE_ENCODER)           += libshine.o
 OBJS-$(CONFIG_LIBSPEEX_DECODER)           += libspeexdec.o
 OBJS-$(CONFIG_LIBSPEEX_ENCODER)           += libspeexenc.o
 OBJS-$(CONFIG_LIBSTAGEFRIGHT_H264_DECODER)+= libstagefright.o
@@ -703,6 +715,7 @@ OBJS-$(CONFIG_LIBVPX_VP8_DECODER)         += libvpxdec.o
 OBJS-$(CONFIG_LIBVPX_VP8_ENCODER)         += libvpxenc.o
 OBJS-$(CONFIG_LIBVPX_VP9_DECODER)         += libvpxdec.o
 OBJS-$(CONFIG_LIBVPX_VP9_ENCODER)         += libvpxenc.o
+OBJS-$(CONFIG_LIBWAVPACK_ENCODER)         += libwavpackenc.o
 OBJS-$(CONFIG_LIBX264_ENCODER)            += libx264.o
 OBJS-$(CONFIG_LIBXAVS_ENCODER)            += libxavs.o
 OBJS-$(CONFIG_LIBXVID_ENCODER)            += libxvid.o
@@ -721,6 +734,7 @@ OBJS-$(CONFIG_DCA_PARSER)              += dca_parser.o dca.o
 OBJS-$(CONFIG_DIRAC_PARSER)            += dirac_parser.o
 OBJS-$(CONFIG_DNXHD_PARSER)            += dnxhd_parser.o
 OBJS-$(CONFIG_DVBSUB_PARSER)           += dvbsub_parser.o
+OBJS-$(CONFIG_DVD_NAV_PARSER)          += dvd_nav_parser.o
 OBJS-$(CONFIG_DVDSUB_PARSER)           += dvdsub_parser.o
 OBJS-$(CONFIG_FLAC_PARSER)             += flac_parser.o flacdata.o flac.o \
                                          vorbis_data.o
@@ -771,9 +785,11 @@ OBJS-$(CONFIG_REMOVE_EXTRADATA_BSF)       += remove_extradata_bsf.o
 OBJS-$(CONFIG_TEXT2MOVSUB_BSF)            += movsub_bsf.o

 # thread libraries
-OBJS-$(HAVE_PTHREADS)                  += pthread.o frame_thread_encoder.o
-OBJS-$(HAVE_W32THREADS)                += pthread.o frame_thread_encoder.o
-OBJS-$(HAVE_OS2THREADS)                += pthread.o frame_thread_encoder.o
+OBJS-$(HAVE_PTHREADS)                  += pthread.o
+OBJS-$(HAVE_W32THREADS)                += pthread.o
+OBJS-$(HAVE_OS2THREADS)                += pthread.o
+
+OBJS-$(CONFIG_FRAME_THREAD_ENCODER)    += frame_thread_encoder.o

 SKIPHEADERS                            += %_tablegen.h                  \
                                          %_tables.h                    \
@@ -790,8 +806,6 @@ SKIPHEADERS-$(CONFIG_MPEG_XVMC_DECODER) += xvmc.h
 SKIPHEADERS-$(CONFIG_VAAPI)            += vaapi_internal.h
 SKIPHEADERS-$(CONFIG_VDA)              += vda.h
 SKIPHEADERS-$(CONFIG_VDPAU)            += vdpau.h
-SKIPHEADERS-$(HAVE_OS2THREADS)         += os2threads.h
-SKIPHEADERS-$(HAVE_W32THREADS)         += w32pthreads.h

 TESTPROGS = cabac                                                       \
            dct                                                         \
--- a/libavcodec/a64enc.h
+++ b/libavcodec/a64enc.h
@@ -1,58 +0,0 @@
-/*
- * a64 video encoder - basic headers
- * Copyright (c) 2009 Tobias Bindhammer
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * a64 video encoder - basic headers
- */
-
-#ifndef AVCODEC_A64ENC_H
-#define AVCODEC_A64ENC_H
-
-#include "libavutil/lfg.h"
-#include "avcodec.h"
-
-#define C64XRES 320
-#define C64YRES 200
-
-typedef struct A64Context {
-    /* general variables */
-    AVFrame picture;
-
-    /* variables for multicolor modes */
-    AVLFG randctx;
-    int mc_lifetime;
-    int mc_use_5col;
-    unsigned mc_frame_counter;
-    int *mc_meta_charset;
-    int *mc_charmap;
-    int *mc_best_cb;
-    int mc_luma_vals[5];
-    uint8_t *mc_charset;
-    uint8_t *mc_colram;
-    uint8_t *mc_palette;
-    int mc_pal_size;
-
-    /* pts of the next packet that will be output */
-    int64_t next_pts;
-} A64Context;
-
-#endif /* AVCODEC_A64ENC_H */
--- a/libavcodec/a64multienc.c
+++ b/libavcodec/a64multienc.c
@@ -24,7 +24,6 @@
 * a64 video encoder - multicolor modes
 */

-#include "a64enc.h"
 #include "a64colors.h"
 #include "a64tables.h"
 #include "elbg.h"
@@ -37,6 +36,31 @@
 #define INTERLACED    1
 #define CROP_SCREENS  1

+#define C64XRES 320
+#define C64YRES 200
+
+typedef struct A64Context {
+    /* general variables */
+    AVFrame picture;
+
+    /* variables for multicolor modes */
+    AVLFG randctx;
+    int mc_lifetime;
+    int mc_use_5col;
+    unsigned mc_frame_counter;
+    int *mc_meta_charset;
+    int *mc_charmap;
+    int *mc_best_cb;
+    int mc_luma_vals[5];
+    uint8_t *mc_charset;
+    uint8_t *mc_colram;
+    uint8_t *mc_palette;
+    int mc_pal_size;
+
+    /* pts of the next packet that will be output */
+    int64_t next_pts;
+} A64Context;
+
 /* gray gradient */
 static const int mc_colors[5]={0x0,0xb,0xc,0xf,0x1};

--- a/libavcodec/aac_adtstoasc_bsf.c
+++ b/libavcodec/aac_adtstoasc_bsf.c
@@ -61,7 +61,8 @@ static int aac_adtstoasc_filter(AVBitStreamFilterContext *bsfc,
    }

    if (!hdr.crc_absent && hdr.num_aac_frames > 1) {
-        av_log_missing_feature(avctx, "Multiple RDBs per frame with CRC", 0);
+        avpriv_report_missing_feature(avctx,
+                                      "Multiple RDBs per frame with CRC");
        return AVERROR_PATCHWELCOME;
    }

@@ -74,7 +75,10 @@ static int aac_adtstoasc_filter(AVBitStreamFilterContext *bsfc,
        if (!hdr.chan_config) {
            init_get_bits(&gb, buf, buf_size * 8);
            if (get_bits(&gb, 3) != 5) {
-                av_log_missing_feature(avctx, "PCE based channel configuration, where the PCE is not the first syntax element", 0);
+                avpriv_report_missing_feature(avctx,
+                                              "PCE-based channel configuration "
+                                              "without PCE as first syntax "
+                                              "element");
                return AVERROR_PATCHWELCOME;
            }
            init_put_bits(&pb, pce_data, MAX_PCE_SIZE);
--- a/libavcodec/aaccoder.c
+++ b/libavcodec/aaccoder.c
@@ -710,7 +710,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
                                          const float lambda)
 {
    int start = 0, i, w, w2, g;
-    int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->channels;
+    int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->channels * (lambda / 120.f);
    float dists[128] = { 0 }, uplims[128];
    float maxvals[128];
    int fflag, minscaler;
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -148,6 +148,8 @@ static av_cold int che_configure(AACContext *ac,
                                 enum ChannelPosition che_pos,
                                 int type, int id, int *channels)
 {
+    if (*channels >= MAX_CHANNELS)
+        return AVERROR_INVALIDDATA;
    if (che_pos) {
        if (!ac->che[type][id]) {
            if (!(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
@@ -190,11 +192,10 @@ static int frame_configure_elements(AVCodecContext *avctx)
    }

    /* get output buffer */
+    av_frame_unref(ac->frame);
    ac->frame->nb_samples = 2048;
-    if ((ret = ff_get_buffer(avctx, ac->frame)) < 0) {
-        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+    if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0)
        return ret;
-    }

    /* map output channel pointers to AVFrame data */
    for (ch = 0; ch < avctx->channels; ch++) {
@@ -214,28 +215,39 @@ struct elem_to_channel {

 static int assign_pair(struct elem_to_channel e2c_vec[MAX_ELEM_ID],
                       uint8_t (*layout_map)[3], int offset, uint64_t left,
-    uint64_t right, int pos)
+                       uint64_t right, int pos)
 {
    if (layout_map[offset][0] == TYPE_CPE) {
        e2c_vec[offset] = (struct elem_to_channel) {
-            .av_position = left | right, .syn_ele = TYPE_CPE,
-            .elem_id = layout_map[offset    ][1], .aac_position = pos };
+            .av_position  = left | right,
+            .syn_ele      = TYPE_CPE,
+            .elem_id      = layout_map[offset][1],
+            .aac_position = pos
+        };
        return 1;
    } else {
-        e2c_vec[offset]   = (struct elem_to_channel) {
-            .av_position = left, .syn_ele = TYPE_SCE,
-            .elem_id = layout_map[offset    ][1], .aac_position = pos };
+        e2c_vec[offset] = (struct elem_to_channel) {
+            .av_position  = left,
+            .syn_ele      = TYPE_SCE,
+            .elem_id      = layout_map[offset][1],
+            .aac_position = pos
+        };
        e2c_vec[offset + 1] = (struct elem_to_channel) {
-            .av_position = right, .syn_ele = TYPE_SCE,
-            .elem_id = layout_map[offset + 1][1], .aac_position = pos };
+            .av_position  = right,
+            .syn_ele      = TYPE_SCE,
+            .elem_id      = layout_map[offset + 1][1],
+            .aac_position = pos
+        };
        return 2;
    }
 }

-static int count_paired_channels(uint8_t (*layout_map)[3], int tags, int pos, int *current) {
+static int count_paired_channels(uint8_t (*layout_map)[3], int tags, int pos,
+                                 int *current)
+{
    int num_pos_channels = 0;
-    int first_cpe = 0;
-    int sce_parity = 0;
+    int first_cpe        = 0;
+    int sce_parity       = 0;
    int i;
    for (i = *current; i < tags; i++) {
        if (layout_map[i][2] != pos)
@@ -249,7 +261,7 @@ static int count_paired_channels(uint8_t (*layout_map)[3], int tags, int pos, in
                }
            }
            num_pos_channels += 2;
-            first_cpe = 1;
+            first_cpe         = 1;
        } else {
            num_pos_channels++;
            sce_parity ^= 1;
@@ -257,7 +269,7 @@ static int count_paired_channels(uint8_t (*layout_map)[3], int tags, int pos, in
    }
    if (sce_parity &&
        ((pos == AAC_CHANNEL_FRONT && first_cpe) || pos == AAC_CHANNEL_SIDE))
-            return -1;
+        return -1;
    *current = i;
    return num_pos_channels;
 }
@@ -265,7 +277,7 @@ static int count_paired_channels(uint8_t (*layout_map)[3], int tags, int pos, in
 static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags)
 {
    int i, n, total_non_cc_elements;
-    struct elem_to_channel e2c_vec[4*MAX_ELEM_ID] = {{ 0 }};
+    struct elem_to_channel e2c_vec[4 * MAX_ELEM_ID] = { { 0 } };
    int num_front_channels, num_side_channels, num_back_channels;
    uint64_t layout;

@@ -289,8 +301,11 @@ static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags)
    i = 0;
    if (num_front_channels & 1) {
        e2c_vec[i] = (struct elem_to_channel) {
-            .av_position = AV_CH_FRONT_CENTER, .syn_ele = TYPE_SCE,
-            .elem_id = layout_map[i][1], .aac_position = AAC_CHANNEL_FRONT };
+            .av_position  = AV_CH_FRONT_CENTER,
+            .syn_ele      = TYPE_SCE,
+            .elem_id      = layout_map[i][1],
+            .aac_position = AAC_CHANNEL_FRONT
+        };
        i++;
        num_front_channels--;
    }
@@ -347,22 +362,31 @@ static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags)
    }
    if (num_back_channels) {
        e2c_vec[i] = (struct elem_to_channel) {
-          .av_position = AV_CH_BACK_CENTER, .syn_ele = TYPE_SCE,
-          .elem_id = layout_map[i][1], .aac_position = AAC_CHANNEL_BACK };
+            .av_position  = AV_CH_BACK_CENTER,
+            .syn_ele      = TYPE_SCE,
+            .elem_id      = layout_map[i][1],
+            .aac_position = AAC_CHANNEL_BACK
+        };
        i++;
        num_back_channels--;
    }

    if (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
        e2c_vec[i] = (struct elem_to_channel) {
-          .av_position = AV_CH_LOW_FREQUENCY, .syn_ele = TYPE_LFE,
-          .elem_id = layout_map[i][1], .aac_position = AAC_CHANNEL_LFE };
+            .av_position  = AV_CH_LOW_FREQUENCY,
+            .syn_ele      = TYPE_LFE,
+            .elem_id      = layout_map[i][1],
+            .aac_position = AAC_CHANNEL_LFE
+        };
        i++;
    }
    while (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
        e2c_vec[i] = (struct elem_to_channel) {
-          .av_position = UINT64_MAX, .syn_ele = TYPE_LFE,
-          .elem_id = layout_map[i][1], .aac_position = AAC_CHANNEL_LFE };
+            .av_position  = UINT64_MAX,
+            .syn_ele      = TYPE_LFE,
+            .elem_id      = layout_map[i][1],
+            .aac_position = AAC_CHANNEL_LFE
+        };
        i++;
    }

@@ -370,12 +394,11 @@ static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags)
    total_non_cc_elements = n = i;
    do {
        int next_n = 0;
-        for (i = 1; i < n; i++) {
-            if (e2c_vec[i-1].av_position > e2c_vec[i].av_position) {
-                FFSWAP(struct elem_to_channel, e2c_vec[i-1], e2c_vec[i]);
+        for (i = 1; i < n; i++)
+            if (e2c_vec[i - 1].av_position > e2c_vec[i].av_position) {
+                FFSWAP(struct elem_to_channel, e2c_vec[i - 1], e2c_vec[i]);
                next_n = i;
            }
-        }
        n = next_n;
    } while (n > 0);

@@ -417,12 +440,13 @@ static void pop_output_configuration(AACContext *ac) {
 }

 /**
- * Configure output channel order based on the current program configuration element.
+ * Configure output channel order based on the current program
+ * configuration element.
 *
 * @return  Returns error status. 0 - OK, !0 - error
 */
 static int output_configure(AACContext *ac,
-                            uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
+                            uint8_t layout_map[MAX_ELEM_ID * 4][3], int tags,
                            enum OCStatus oc_type, int get_new_frame)
 {
    AVCodecContext *avctx = ac->avctx;
@@ -458,8 +482,8 @@ static int output_configure(AACContext *ac,

    memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
    if (layout) avctx->channel_layout = layout;
-    ac->oc[1].channel_layout = layout;
-    avctx->channels = ac->oc[1].channels = channels;
+                            ac->oc[1].channel_layout = layout;
+    avctx->channels       = ac->oc[1].channels       = channels;
    ac->oc[1].status = oc_type;

    if (get_new_frame) {
@@ -494,36 +518,40 @@ static void flush(AVCodecContext *avctx)
 * @return  Returns error status. 0 - OK, !0 - error
 */
 static int set_default_channel_config(AVCodecContext *avctx,
-                                              uint8_t (*layout_map)[3],
-                                              int *tags,
-                                              int channel_config)
+                                      uint8_t (*layout_map)[3],
+                                      int *tags,
+                                      int channel_config)
 {
    if (channel_config < 1 || channel_config > 7) {
-        av_log(avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
+        av_log(avctx, AV_LOG_ERROR,
+               "invalid default channel configuration (%d)\n",
               channel_config);
-        return -1;
+        return AVERROR_INVALIDDATA;
    }
    *tags = tags_per_config[channel_config];
-    memcpy(layout_map, aac_channel_layout_map[channel_config-1], *tags * sizeof(*layout_map));
+    memcpy(layout_map, aac_channel_layout_map[channel_config - 1],
+           *tags * sizeof(*layout_map));
    return 0;
 }

 static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
 {
-    // For PCE based channel configurations map the channels solely based on tags.
+    /* For PCE based channel configurations map the channels solely based
+     * on tags. */
    if (!ac->oc[1].m4ac.chan_config) {
        return ac->tag_che_map[type][elem_id];
    }
    // Allow single CPE stereo files to be signalled with mono configuration.
-    if (!ac->tags_mapped && type == TYPE_CPE && ac->oc[1].m4ac.chan_config == 1) {
+    if (!ac->tags_mapped && type == TYPE_CPE &&
+        ac->oc[1].m4ac.chan_config == 1) {
        uint8_t layout_map[MAX_ELEM_ID*4][3];
        int layout_map_tags;
        push_output_configuration(ac);

        av_log(ac->avctx, AV_LOG_DEBUG, "mono with CPE\n");

-        if (set_default_channel_config(ac->avctx, layout_map, &layout_map_tags,
-                                       2) < 0)
+        if (set_default_channel_config(ac->avctx, layout_map,
+                                       &layout_map_tags, 2) < 0)
            return NULL;
        if (output_configure(ac, layout_map, layout_map_tags,
                             OC_TRIAL_FRAME, 1) < 0)
@@ -533,15 +561,16 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
        ac->oc[1].m4ac.ps = 0;
    }
    // And vice-versa
-    if (!ac->tags_mapped && type == TYPE_SCE && ac->oc[1].m4ac.chan_config == 2) {
-        uint8_t layout_map[MAX_ELEM_ID*4][3];
+    if (!ac->tags_mapped && type == TYPE_SCE &&
+        ac->oc[1].m4ac.chan_config == 2) {
+        uint8_t layout_map[MAX_ELEM_ID * 4][3];
        int layout_map_tags;
        push_output_configuration(ac);

        av_log(ac->avctx, AV_LOG_DEBUG, "stereo with SCE\n");

-        if (set_default_channel_config(ac->avctx, layout_map, &layout_map_tags,
-                                       1) < 0)
+        if (set_default_channel_config(ac->avctx, layout_map,
+                                       &layout_map_tags, 1) < 0)
            return NULL;
        if (output_configure(ac, layout_map, layout_map_tags,
                             OC_TRIAL_FRAME, 1) < 0)
@@ -551,7 +580,8 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
        if (ac->oc[1].m4ac.sbr)
            ac->oc[1].m4ac.ps = -1;
    }
-    // For indexed channel configurations map the channels solely based on position.
+    /* For indexed channel configurations map the channels solely based
+     * on position. */
    switch (ac->oc[1].m4ac.chan_config) {
    case 7:
        if (ac->tags_mapped == 3 && type == TYPE_CPE) {
@@ -559,9 +589,12 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
        }
    case 6:
-        /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1]
-           instead of SCE[0] CPE[0] CPE[1] LFE[0]. If we seem to have
-           encountered such a stream, transfer the LFE[0] element to the SCE[1]'s mapping */
+        /* Some streams incorrectly code 5.1 audio as
+         * SCE[0] CPE[0] CPE[1] SCE[1]
+         * instead of
+         * SCE[0] CPE[0] CPE[1] LFE[0].
+         * If we seem to have encountered such a stream, transfer
+         * the LFE[0] element to the SCE[1]'s mapping */
        if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
            ac->tags_mapped++;
            return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
@@ -572,13 +605,16 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
        }
    case 4:
-        if (ac->tags_mapped == 2 && ac->oc[1].m4ac.chan_config == 4 && type == TYPE_SCE) {
+        if (ac->tags_mapped == 2 &&
+            ac->oc[1].m4ac.chan_config == 4 &&
+            type == TYPE_SCE) {
            ac->tags_mapped++;
            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
        }
    case 3:
    case 2:
-        if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) && type == TYPE_CPE) {
+        if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) &&
+            type == TYPE_CPE) {
            ac->tags_mapped++;
            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
        } else if (ac->oc[1].m4ac.chan_config == 2) {
@@ -595,7 +631,8 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
 }

 /**
- * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
+ * Decode an array of 4 bit element IDs, optionally interleaved with a
+ * stereo/mono switching bit.
 *
 * @param type speaker type/position for these channels
 */
@@ -637,7 +674,8 @@ static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
                      uint8_t (*layout_map)[3],
                      GetBitContext *gb)
 {
-    int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
+    int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
+    int sampling_index;
    int comment_len;
    int tags;

@@ -645,7 +683,9 @@ static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,

    sampling_index = get_bits(gb, 4);
    if (m4ac->sampling_index != sampling_index)
-        av_log(avctx, AV_LOG_WARNING, "Sample rate index in program config element does not match the sample rate index configured by the container.\n");
+        av_log(avctx, AV_LOG_WARNING,
+               "Sample rate index in program config element does not "
+               "match the sample rate index configured by the container.\n");

    num_front       = get_bits(gb, 4);
    num_side        = get_bits(gb, 4);
@@ -686,7 +726,7 @@ static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
    comment_len = get_bits(gb, 8) * 8;
    if (get_bits_left(gb) < comment_len) {
        av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
-        return -1;
+        return AVERROR_INVALIDDATA;
    }
    skip_bits_long(gb, comment_len);
    return tags;
@@ -710,7 +750,7 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
    int tags = 0;

    if (get_bits1(gb)) { // frameLengthFlag
-        av_log_missing_feature(avctx, "960/120 MDCT window", 1);
+        avpriv_request_sample(avctx, "960/120 MDCT window");
        return AVERROR_PATCHWELCOME;
    }

@@ -728,7 +768,8 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
        if (tags < 0)
            return tags;
    } else {
-        if ((ret = set_default_channel_config(avctx, layout_map, &tags, channel_config)))
+        if ((ret = set_default_channel_config(avctx, layout_map,
+                                              &tags, channel_config)))
            return ret;
    }

@@ -750,7 +791,7 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
        case AOT_ER_AAC_LTP:
        case AOT_ER_AAC_SCALABLE:
        case AOT_ER_AAC_LD:
-            skip_bits(gb, 3);  /* aacSectionDataResilienceFlag
+            skip_bits(gb, 3);      /* aacSectionDataResilienceFlag
                                    * aacScalefactorDataResilienceFlag
                                    * aacSpectralDataResilienceFlag
                                    */
@@ -780,22 +821,24 @@ static int decode_audio_specific_config(AACContext *ac,
                                        int sync_extension)
 {
    GetBitContext gb;
-    int i;
-    int ret;
+    int i, ret;

    av_dlog(avctx, "audio specific config size %d\n", bit_size >> 3);
    for (i = 0; i < bit_size >> 3; i++)
-         av_dlog(avctx, "%02x ", data[i]);
+        av_dlog(avctx, "%02x ", data[i]);
    av_dlog(avctx, "\n");

    if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
        return ret;

-    if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size, sync_extension)) < 0)
-        return -1;
+    if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size,
+                                          sync_extension)) < 0)
+        return AVERROR_INVALIDDATA;
    if (m4ac->sampling_index > 12) {
-        av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index);
-        return -1;
+        av_log(avctx, AV_LOG_ERROR,
+               "invalid sampling rate index %d\n",
+               m4ac->sampling_index);
+        return AVERROR_INVALIDDATA;
    }

    skip_bits_long(&gb, i);
@@ -804,18 +847,23 @@ static int decode_audio_specific_config(AACContext *ac,
    case AOT_AAC_MAIN:
    case AOT_AAC_LC:
    case AOT_AAC_LTP:
-        if (decode_ga_specific_config(ac, avctx, &gb, m4ac, m4ac->chan_config))
-            return -1;
+        if ((ret = decode_ga_specific_config(ac, avctx, &gb,
+                                            m4ac, m4ac->chan_config)) < 0)
+            return ret;
        break;
    default:
-        av_log(avctx, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
-               m4ac->sbr == 1? "SBR+" : "", m4ac->object_type);
-        return -1;
+        av_log(avctx, AV_LOG_ERROR,
+               "Audio object type %s%d is not supported.\n",
+               m4ac->sbr == 1 ? "SBR+" : "",
+               m4ac->object_type);
+        return AVERROR(ENOSYS);
    }

-    av_dlog(avctx, "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
+    av_dlog(avctx,
+            "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
            m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
-            m4ac->sample_rate, m4ac->sbr, m4ac->ps);
+            m4ac->sample_rate, m4ac->sbr,
+            m4ac->ps);

    return get_bits_count(&gb);
 }
@@ -873,10 +921,12 @@ static void reset_predictor_group(PredictorState *ps, int group_num)
        reset_predict_state(&ps[i]);
 }

-#define AAC_INIT_VLC_STATIC(num, size) \
-    INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
-         ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
-        ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
+#define AAC_INIT_VLC_STATIC(num, size)                                     \
+    INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num],     \
+         ff_aac_spectral_bits[num], sizeof(ff_aac_spectral_bits[num][0]),  \
+                                    sizeof(ff_aac_spectral_bits[num][0]),  \
+        ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), \
+                                    sizeof(ff_aac_spectral_codes[num][0]), \
        size);

 static void aacdec_init(AACContext *ac);
@@ -884,6 +934,7 @@ static void aacdec_init(AACContext *ac);
 static av_cold int aac_decode_init(AVCodecContext *avctx)
 {
    AACContext *ac = avctx->priv_data;
+    int ret;

    ac->avctx = avctx;
    ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
@@ -893,10 +944,11 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
    avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

    if (avctx->extradata_size > 0) {
-        if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
-                                         avctx->extradata,
-                                         avctx->extradata_size*8, 1) < 0)
-            return -1;
+        if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
+                                                avctx->extradata,
+                                                avctx->extradata_size * 8,
+                                                1)) < 0)
+            return ret;
    } else {
        int sr, i;
        uint8_t layout_map[MAX_ELEM_ID*4][3];
@@ -953,9 +1005,14 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)

    ff_aac_tableinit();

-    INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
-                    ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
-                    ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
+    INIT_VLC_STATIC(&vlc_scalefactors, 7,
+                    FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
+                    ff_aac_scalefactor_bits,
+                    sizeof(ff_aac_scalefactor_bits[0]),
+                    sizeof(ff_aac_scalefactor_bits[0]),
+                    ff_aac_scalefactor_code,
+                    sizeof(ff_aac_scalefactor_code[0]),
+                    sizeof(ff_aac_scalefactor_code[0]),
                    352);

    ff_mdct_init(&ac->mdct,       11, 1, 1.0 / (32768.0 * 1024.0));
@@ -986,7 +1043,7 @@ static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)

    if (get_bits_left(gb) < 8 * count) {
        av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err);
-        return -1;
+        return AVERROR_INVALIDDATA;
    }
    skip_bits_long(gb, 8 * count);
    return 0;
@@ -998,9 +1055,11 @@ static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
    int sfb;
    if (get_bits1(gb)) {
        ics->predictor_reset_group = get_bits(gb, 5);
-        if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
-            av_log(ac->avctx, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
-            return -1;
+        if (ics->predictor_reset_group == 0 ||
+            ics->predictor_reset_group > 30) {
+            av_log(ac->avctx, AV_LOG_ERROR,
+                   "Invalid Predictor Reset Group.\n");
+            return AVERROR_INVALIDDATA;
        }
    }
    for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) {
@@ -1069,7 +1128,8 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
                    goto fail;
                }
            } else if (ac->oc[1].m4ac.object_type == AOT_AAC_LC) {
-                av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
+                av_log(ac->avctx, AV_LOG_ERROR,
+                       "Prediction is not allowed in AAC-LC.\n");
                goto fail;
            } else {
                if ((ics->ltp.present = get_bits(gb, 1)))
@@ -1080,7 +1140,8 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,

    if (ics->max_sfb > ics->num_swb) {
        av_log(ac->avctx, AV_LOG_ERROR,
-               "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
+               "Number of scalefactor bands in group (%d) "
+               "exceeds limit (%d).\n",
               ics->max_sfb, ics->num_swb);
        goto fail;
    }
@@ -1113,20 +1174,20 @@ static int decode_band_types(AACContext *ac, enum BandType band_type[120],
            int sect_band_type = get_bits(gb, 4);
            if (sect_band_type == 12) {
                av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
-                return -1;
+                return AVERROR_INVALIDDATA;
            }
            do {
                sect_len_incr = get_bits(gb, bits);
                sect_end += sect_len_incr;
                if (get_bits_left(gb) < 0) {
                    av_log(ac->avctx, AV_LOG_ERROR, "decode_band_types: "overread_err);
-                    return -1;
+                    return AVERROR_INVALIDDATA;
                }
                if (sect_end > ics->max_sfb) {
                    av_log(ac->avctx, AV_LOG_ERROR,
                           "Number of bands (%d) exceeds limit (%d).\n",
                           sect_end, ics->max_sfb);
-                    return -1;
+                    return AVERROR_INVALIDDATA;
                }
            } while (sect_len_incr == (1 << bits) - 1);
            for (; k < sect_end; k++) {
@@ -1164,15 +1225,16 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
            if (band_type[idx] == ZERO_BT) {
                for (; i < run_end; i++, idx++)
                    sf[idx] = 0.;
-            } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
+            } else if ((band_type[idx] == INTENSITY_BT) ||
+                       (band_type[idx] == INTENSITY_BT2)) {
                for (; i < run_end; i++, idx++) {
                    offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
                    clipped_offset = av_clip(offset[2], -155, 100);
                    if (offset[2] != clipped_offset) {
-                        av_log_ask_for_sample(ac->avctx, "Intensity stereo "
-                                "position clipped (%d -> %d).\nIf you heard an "
-                                "audible artifact, there may be a bug in the "
-                                "decoder. ", offset[2], clipped_offset);
+                        avpriv_request_sample(ac->avctx,
+                                              "If you heard an audible artifact, there may be a bug in the decoder. "
+                                              "Clipped intensity stereo position (%d -> %d)",
+                                              offset[2], clipped_offset);
                    }
                    sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
                }
@@ -1184,10 +1246,10 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
                        offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
                    clipped_offset = av_clip(offset[1], -100, 155);
                    if (offset[1] != clipped_offset) {
-                        av_log_ask_for_sample(ac->avctx, "Noise gain clipped "
-                                "(%d -> %d).\nIf you heard an audible "
-                                "artifact, there may be a bug in the decoder. ",
-                                offset[1], clipped_offset);
+                        avpriv_request_sample(ac->avctx,
+                                              "If you heard an audible artifact, there may be a bug in the decoder. "
+                                              "Clipped noise gain (%d -> %d)",
+                                              offset[1], clipped_offset);
                    }
                    sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
                }
@@ -1197,7 +1259,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
                    if (offset[0] > 255U) {
                        av_log(ac->avctx, AV_LOG_ERROR,
                               "Scalefactor (%d) out of range.\n", offset[0]);
-                        return -1;
+                        return AVERROR_INVALIDDATA;
                    }
                    sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
                }
@@ -1252,10 +1314,11 @@ static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
                tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);

                if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
-                    av_log(ac->avctx, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n",
+                    av_log(ac->avctx, AV_LOG_ERROR,
+                           "TNS filter order %d is greater than maximum %d.\n",
                           tns->order[w][filt], tns_max_order);
                    tns->order[w][filt] = 0;
-                    return -1;
+                    return AVERROR_INVALIDDATA;
                }
                if (tns->order[w][filt]) {
                    tns->direction[w][filt] = get_bits1(gb);
@@ -1284,7 +1347,9 @@ static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
 {
    int idx;
    if (ms_present == 1) {
-        for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
+        for (idx = 0;
+             idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
+             idx++)
            cpe->ms_mask[idx] = get_bits1(gb);
    } else if (ms_present == 2) {
        memset(cpe->ms_mask, 1,  sizeof(cpe->ms_mask[0]) * cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb);
@@ -1383,7 +1448,8 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
    float *coef_base = coef;

    for (g = 0; g < ics->num_windows; g++)
-        memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb]));
+        memset(coef + g * 128 + offsets[ics->max_sfb], 0,
+               sizeof(float) * (c - offsets[ics->max_sfb]));

    for (g = 0; g < ics->num_window_groups; g++) {
        unsigned g_len = ics->group_len[g];
@@ -1538,7 +1604,7 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],

                                    if (b > 8) {
                                        av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
-                                        return -1;
+                                        return AVERROR_INVALIDDATA;
                                    }

                                    SKIP_BITS(re, gb, b + 1);
@@ -1653,14 +1719,20 @@ static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
    }

    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
-        for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]; sfb++) {
-            for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
+        for (sfb = 0;
+             sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index];
+             sfb++) {
+            for (k = sce->ics.swb_offset[sfb];
+                 k < sce->ics.swb_offset[sfb + 1];
+                 k++) {
                predict(&sce->predictor_state[k], &sce->coeffs[k],
-                        sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
+                        sce->ics.predictor_present &&
+                        sce->ics.prediction_used[sfb]);
            }
        }
        if (sce->ics.predictor_reset_group)
-            reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
+            reset_predictor_group(sce->predictor_state,
+                                  sce->ics.predictor_reset_group);
    } else
        reset_all_predictors(sce->predictor_state);
 }
@@ -1681,6 +1753,7 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
    IndividualChannelStream *ics = &sce->ics;
    float *out = sce->coeffs;
    int global_gain, pulse_present = 0;
+    int ret;

    /* This assignment is to silence a GCC warning about the variable being used
     * uninitialized when in fact it always is.
@@ -1694,33 +1767,38 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
            return AVERROR_INVALIDDATA;
    }

-    if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
-        return -1;
-    if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
-        return -1;
+    if ((ret = decode_band_types(ac, sce->band_type,
+                                 sce->band_type_run_end, gb, ics)) < 0)
+        return ret;
+    if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics,
+                                  sce->band_type, sce->band_type_run_end)) < 0)
+        return ret;

    pulse_present = 0;
    if (!scale_flag) {
        if ((pulse_present = get_bits1(gb))) {
            if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
-                av_log(ac->avctx, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
-                return -1;
+                av_log(ac->avctx, AV_LOG_ERROR,
+                       "Pulse tool not allowed in eight short sequence.\n");
+                return AVERROR_INVALIDDATA;
            }
            if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
-                av_log(ac->avctx, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
-                return -1;
+                av_log(ac->avctx, AV_LOG_ERROR,
+                       "Pulse data corrupt or invalid.\n");
+                return AVERROR_INVALIDDATA;
            }
        }
        if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
-            return -1;
+            return AVERROR_INVALIDDATA;
        if (get_bits1(gb)) {
-            av_log_missing_feature(ac->avctx, "SSR", 1);
+            avpriv_request_sample(ac->avctx, "SSR");
            return AVERROR_PATCHWELCOME;
        }
    }

-    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
-        return -1;
+    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
+                                    &pulse, ics, sce->band_type) < 0)
+        return AVERROR_INVALIDDATA;

    if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
        apply_prediction(ac, sce);
@@ -1741,7 +1819,8 @@ static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
    for (g = 0; g < ics->num_window_groups; g++) {
        for (i = 0; i < ics->max_sfb; i++, idx++) {
            if (cpe->ms_mask[idx] &&
-                    cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
+                cpe->ch[0].band_type[idx] < NOISE_BT &&
+                cpe->ch[1].band_type[idx] < NOISE_BT) {
                for (group = 0; group < ics->group_len[g]; group++) {
                    ac->fdsp.butterflies_float(ch0 + group * 128 + offsets[i],
                                               ch1 + group * 128 + offsets[i],
@@ -1761,7 +1840,8 @@ static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
 *                      [3] reserved for scalable AAC
 */
-static void apply_intensity_stereo(AACContext *ac, ChannelElement *cpe, int ms_present)
+static void apply_intensity_stereo(AACContext *ac,
+                                   ChannelElement *cpe, int ms_present)
 {
    const IndividualChannelStream *ics = &cpe->ch[1].ics;
    SingleChannelElement         *sce1 = &cpe->ch[1];
@@ -1772,7 +1852,8 @@ static void apply_intensity_stereo(AACContext *ac, ChannelElement *cpe, int ms_p
    float scale;
    for (g = 0; g < ics->num_window_groups; g++) {
        for (i = 0; i < ics->max_sfb;) {
-            if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) {
+            if (sce1->band_type[idx] == INTENSITY_BT ||
+                sce1->band_type[idx] == INTENSITY_BT2) {
                const int bt_run_end = sce1->band_type_run_end[idx];
                for (; i < bt_run_end; i++, idx++) {
                    c = -1 + 2 * (sce1->band_type[idx] - 14);
@@ -1812,13 +1893,14 @@ static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
        i = cpe->ch[1].ics.use_kb_window[0];
        cpe->ch[1].ics = cpe->ch[0].ics;
        cpe->ch[1].ics.use_kb_window[1] = i;
-        if (cpe->ch[1].ics.predictor_present && (ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
+        if (cpe->ch[1].ics.predictor_present &&
+            (ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
            if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
                decode_ltp(&cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
        ms_present = get_bits(gb, 2);
        if (ms_present == 3) {
            av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
-            return -1;
+            return AVERROR_INVALIDDATA;
        } else if (ms_present)
            decode_mid_side_stereo(cpe, gb, ms_present);
    }
@@ -2431,7 +2513,8 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
        if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
            // This is 2 for "VLB " audio in NSV files.
            // See samples/nsv/vlb_audio.
-            av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame", 0);
+            avpriv_report_missing_feature(ac->avctx,
+                                          "More than one AAC RDB per ADTS frame");
            ac->warned_num_aac_frames = 1;
        }
        push_output_configuration(ac);
@@ -2609,6 +2692,8 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,

    if (samples)
        ac->frame->nb_samples = samples;
+    else
+        av_frame_unref(ac->frame);
    *got_frame_ptr = !!samples;

    if (is_dmono) {
@@ -2717,13 +2802,13 @@ static av_cold int aac_decode_close(AVCodecContext *avctx)
 #define LOAS_SYNC_WORD   0x2b7       ///< 11 bits LOAS sync word

 struct LATMContext {
-    AACContext      aac_ctx;             ///< containing AACContext
-    int             initialized;         ///< initialized after a valid extradata was seen
+    AACContext aac_ctx;     ///< containing AACContext
+    int initialized;        ///< initialized after a valid extradata was seen

    // parser data
-    int             audio_mux_version_A; ///< LATM syntax version
-    int             frame_length_type;   ///< 0/1 variable/fixed frame length
-    int             frame_length;        ///< frame length for fixed frame length
+    int audio_mux_version_A; ///< LATM syntax version
+    int frame_length_type;   ///< 0/1 variable/fixed frame length
+    int frame_length;        ///< frame length for fixed frame length
 };

 static inline uint32_t latm_get_value(GetBitContext *b)
@@ -2750,8 +2835,8 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
        asclen         = get_bits_left(gb);

    if (config_start_bit % 8) {
-        av_log_missing_feature(latmctx->aac_ctx.avctx,
-                               "Non-byte-aligned audio-specific config", 1);
+        avpriv_request_sample(latmctx->aac_ctx.avctx,
+                              "Non-byte-aligned audio-specific config");
        return AVERROR_PATCHWELCOME;
    }
    if (asclen <= 0)
@@ -2810,8 +2895,7 @@ static int read_stream_mux_config(struct LATMContext *latmctx,
        skip_bits(gb, 6);                       // numSubFrames
        // numPrograms
        if (get_bits(gb, 4)) {                  // numPrograms
-            av_log_missing_feature(latmctx->aac_ctx.avctx,
-                                   "Multiple programs", 1);
+            avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple programs");
            return AVERROR_PATCHWELCOME;
        }

@@ -2819,8 +2903,7 @@ static int read_stream_mux_config(struct LATMContext *latmctx,

        // for each layer (which there is only one in DVB)
        if (get_bits(gb, 3)) {                   // numLayer
-            av_log_missing_feature(latmctx->aac_ctx.avctx,
-                                   "Multiple layers", 1);
+            avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers");
            return AVERROR_PATCHWELCOME;
        }

--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -593,7 +593,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                coeffs[ch] = cpe->ch[ch].coeffs;
            s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
            for (ch = 0; ch < chans; ch++) {
-                s->cur_channel = start_ch * 2 + ch;
+                s->cur_channel = start_ch + ch;
                s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
            }
            cpe->common_window = 0;
@@ -609,7 +609,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                    }
                }
            }
-            s->cur_channel = start_ch * 2;
+            s->cur_channel = start_ch;
            if (s->options.stereo_mode && cpe->common_window) {
                if (s->options.stereo_mode > 0) {
                    IndividualChannelStream *ics = &cpe->ch[0].ics;
@@ -679,9 +679,6 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)
    av_freep(&s->buffer.samples);
    av_freep(&s->cpe);
    ff_af_queue_close(&s->afq);
-#if FF_API_OLD_ENCODE_AUDIO
-    av_freep(&avctx->coded_frame);
-#endif
    return 0;
 }

@@ -715,11 +712,6 @@ static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
    for(ch = 0; ch < s->channels; ch++)
        s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;

-#if FF_API_OLD_ENCODE_AUDIO
-    if (!(avctx->coded_frame = avcodec_alloc_frame()))
-        goto alloc_fail;
-#endif
-
    return 0;
 alloc_fail:
    return AVERROR(ENOMEM);
@@ -774,6 +766,9 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
    s->psypp = ff_psy_preprocess_init(avctx);
    s->coder = &ff_aac_coders[s->options.aac_coder];

+    if (HAVE_MIPSDSPR1)
+        ff_aac_coder_init_mips(s);
+
    s->lambda = avctx->global_quality ? avctx->global_quality : 120;

    ff_aac_tableinit();
--- a/libavcodec/aacenc.h
+++ b/libavcodec/aacenc.h
@@ -85,4 +85,6 @@ typedef struct AACEncContext {

 extern float ff_aac_pow34sf_tab[428];

+void ff_aac_coder_init_mips(AACEncContext *c);
+
 #endif /* AVCODEC_AACENC_H */
--- a/libavcodec/aacps.c
+++ b/libavcodec/aacps.c
@@ -823,7 +823,8 @@ static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2
            h12 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][1];
            h21 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][2];
            h22 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][3];
-            if (!PS_BASELINE && ps->enable_ipdopd && b < ps->nr_ipdopd_par) {
+
+            if (!PS_BASELINE && ps->enable_ipdopd && 2*b <= NR_PAR_BANDS[is34]) {
                //The spec say says to only run this smoother when enable_ipdopd
                //is set but the reference decoder appears to run it constantly
                float h11i, h12i, h21i, h22i;
--- a/libavcodec/aacpsy.c
+++ b/libavcodec/aacpsy.c
@@ -24,6 +24,7 @@
 * AAC encoder psychoacoustic model
 */

+#include "libavutil/attributes.h"
 #include "libavutil/libm.h"

 #include "avcodec.h"
@@ -217,6 +218,10 @@ static const float psy_fir_coeffs[] = {
    -5.52212e-17 * 2, -0.313819 * 2
 };

+#if ARCH_MIPS
+#   include "mips/aacpsy_mips.h"
+#endif /* ARCH_MIPS */
+
 /**
 * Calculate the ABR attack threshold from the above LAME psymodel table.
 */
@@ -250,7 +255,8 @@ static float lame_calc_attack_threshold(int bitrate)
 /**
 * LAME psy model specific initialization
 */
-static void lame_window_init(AacPsyContext *ctx, AVCodecContext *avctx) {
+static av_cold void lame_window_init(AacPsyContext *ctx, AVCodecContext *avctx)
+{
    int i, j;

    for (i = 0; i < avctx->channels; i++) {
@@ -312,7 +318,7 @@ static av_cold int psy_3gpp_init(FFPsyContext *ctx) {
        AacPsyCoeffs *coeffs = pctx->psy_coef[j];
        const uint8_t *band_sizes = ctx->bands[j];
        float line_to_frequency = ctx->avctx->sample_rate / (j ? 256.f : 2048.0f);
-        float avg_chan_bits = chan_bitrate / ctx->avctx->sample_rate * (j ? 128.0f : 1024.0f);
+        float avg_chan_bits = chan_bitrate * (j ? 128.0f : 1024.0f) / ctx->avctx->sample_rate;
        /* reference encoder uses 2.4% here instead of 60% like the spec says */
        float bark_pe = 0.024f * PSY_3GPP_BITS_TO_PE(avg_chan_bits) / num_bark;
        float en_spread_low = j ? PSY_3GPP_EN_SPREAD_LOW_S : PSY_3GPP_EN_SPREAD_LOW_L;
@@ -560,25 +566,12 @@ static float calc_reduced_thr_3gpp(AacPsyBand *band, float min_snr,
    return thr;
 }

-/**
- * Calculate band thresholds as suggested in 3GPP TS26.403
- */
-static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
-                                     const float *coefs, const FFPsyWindowInfo *wi)
+#ifndef calc_thr_3gpp
+static void calc_thr_3gpp(const FFPsyWindowInfo *wi, const int num_bands, AacPsyChannel *pch,
+                          const uint8_t *band_sizes, const float *coefs)
 {
-    AacPsyContext *pctx = (AacPsyContext*) ctx->model_priv_data;
-    AacPsyChannel *pch  = &pctx->ch[channel];
-    int start = 0;
    int i, w, g;
-    float desired_bits, desired_pe, delta_pe, reduction= NAN, spread_en[128] = {0};
-    float a = 0.0f, active_lines = 0.0f, norm_fac = 0.0f;
-    float pe = pctx->chan_bitrate > 32000 ? 0.0f : FFMAX(50.0f, 100.0f - pctx->chan_bitrate * 100.0f / 32000.0f);
-    const int      num_bands   = ctx->num_bands[wi->num_windows == 8];
-    const uint8_t *band_sizes  = ctx->bands[wi->num_windows == 8];
-    AacPsyCoeffs  *coeffs      = pctx->psy_coef[wi->num_windows == 8];
-    const float avoid_hole_thr = wi->num_windows == 8 ? PSY_3GPP_AH_THR_SHORT : PSY_3GPP_AH_THR_LONG;
-
-    //calculate energies, initial thresholds and related values - 5.4.2 "Threshold Calculation"
+    int start = 0;
    for (w = 0; w < wi->num_windows*16; w += 16) {
        for (g = 0; g < num_bands; g++) {
            AacPsyBand *band = &pch->band[w+g];
@@ -597,6 +590,48 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
            start += band_sizes[g];
        }
    }
+}
+#endif /* calc_thr_3gpp */
+
+#ifndef psy_hp_filter
+static void psy_hp_filter(const float *firbuf, float *hpfsmpl, const float *psy_fir_coeffs)
+{
+    int i, j;
+    for (i = 0; i < AAC_BLOCK_SIZE_LONG; i++) {
+        float sum1, sum2;
+        sum1 = firbuf[i + (PSY_LAME_FIR_LEN - 1) / 2];
+        sum2 = 0.0;
+        for (j = 0; j < ((PSY_LAME_FIR_LEN - 1) / 2) - 1; j += 2) {
+            sum1 += psy_fir_coeffs[j] * (firbuf[i + j] + firbuf[i + PSY_LAME_FIR_LEN - j]);
+            sum2 += psy_fir_coeffs[j + 1] * (firbuf[i + j + 1] + firbuf[i + PSY_LAME_FIR_LEN - j - 1]);
+        }
+        /* NOTE: The LAME psymodel expects it's input in the range -32768 to 32768.
+         *       Tuning this for normalized floats would be difficult. */
+        hpfsmpl[i] = (sum1 + sum2) * 32768.0f;
+    }
+}
+#endif /* psy_hp_filter */
+
+/**
+ * Calculate band thresholds as suggested in 3GPP TS26.403
+ */
+static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
+                                     const float *coefs, const FFPsyWindowInfo *wi)
+{
+    AacPsyContext *pctx = (AacPsyContext*) ctx->model_priv_data;
+    AacPsyChannel *pch  = &pctx->ch[channel];
+    int i, w, g;
+    float desired_bits, desired_pe, delta_pe, reduction= NAN, spread_en[128] = {0};
+    float a = 0.0f, active_lines = 0.0f, norm_fac = 0.0f;
+    float pe = pctx->chan_bitrate > 32000 ? 0.0f : FFMAX(50.0f, 100.0f - pctx->chan_bitrate * 100.0f / 32000.0f);
+    const int      num_bands   = ctx->num_bands[wi->num_windows == 8];
+    const uint8_t *band_sizes  = ctx->bands[wi->num_windows == 8];
+    AacPsyCoeffs  *coeffs      = pctx->psy_coef[wi->num_windows == 8];
+    const float avoid_hole_thr = wi->num_windows == 8 ? PSY_3GPP_AH_THR_SHORT : PSY_3GPP_AH_THR_LONG;
+
+    //calculate energies, initial thresholds and related values - 5.4.2 "Threshold Calculation"
+    calc_thr_3gpp(wi, num_bands, pch, band_sizes, coefs);
+
    //modify thresholds and energies - spread, threshold in quiet, pre-echo control
    for (w = 0; w < wi->num_windows*16; w += 16) {
        AacPsyBand *bands = &pch->band[w];
@@ -802,20 +837,10 @@ static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio,
        float energy_subshort[(AAC_NUM_BLOCKS_SHORT + 1) * PSY_LAME_NUM_SUBBLOCKS];
        float energy_short[AAC_NUM_BLOCKS_SHORT + 1] = { 0 };
        const float *firbuf = la + (AAC_BLOCK_SIZE_SHORT/4 - PSY_LAME_FIR_LEN);
-        int j, att_sum = 0;
+        int att_sum = 0;

        /* LAME comment: apply high pass filter of fs/4 */
-        for (i = 0; i < AAC_BLOCK_SIZE_LONG; i++) {
-            float sum1, sum2;
-            sum1 = firbuf[i + (PSY_LAME_FIR_LEN - 1) / 2];
-            sum2 = 0.0;
-            for (j = 0; j < ((PSY_LAME_FIR_LEN - 1) / 2) - 1; j += 2) {
-                sum1 += psy_fir_coeffs[j] * (firbuf[i + j] + firbuf[i + PSY_LAME_FIR_LEN - j]);
-                sum2 += psy_fir_coeffs[j + 1] * (firbuf[i + j + 1] + firbuf[i + PSY_LAME_FIR_LEN - j - 1]);
-            }
-            /* NOTE: The LAME psymodel expects it's input in the range -32768 to 32768. Tuning this for normalized floats would be difficult. */
-            hpfsmpl[i] = (sum1 + sum2) * 32768.0f;
-        }
+        psy_hp_filter(firbuf, hpfsmpl, psy_fir_coeffs);

        /* Calculate the energies of each sub-shortblock */
        for (i = 0; i < PSY_LAME_NUM_SUBBLOCKS; i++) {
--- a/libavcodec/aacpsy.h
+++ b/libavcodec/aacpsy.h
@@ -1,50 +0,0 @@
-/*
- * AAC encoder psychoacoustic model
- * Copyright (C) 2008 Konstantin Shishkov
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef AVCODEC_AACPSY_H
-#define AVCODEC_AACPSY_H
-
-#include "avcodec.h"
-#include "aac.h"
-//#include "lowpass.h"
-
-enum AACPsyModelType{
-    AAC_PSY_TEST,              ///< a sample model to exercise encoder
-    AAC_PSY_3GPP,              ///< model following recommendations from 3GPP TS 26.403
-
-    AAC_NB_PSY_MODELS          ///< total number of psychoacoustic models, since it's not a part of the ABI new models can be added freely
-};
-
-/**
- * context used by psychoacoustic model
- */
-typedef struct AACPsyContext {
-    AVCodecContext *avctx;            ///< encoder context
-}AACPsyContext;
-
-/**
- * Cleanup model context at the end.
- *
- * @param ctx model context
- */
-void ff_aac_psy_end(AACPsyContext *ctx);
-
-#endif /* AVCODEC_AACPSY_H */
--- a/libavcodec/aacsbr.c
+++ b/libavcodec/aacsbr.c
@@ -933,7 +933,7 @@ static void read_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
 #if 1
            *num_bits_left -= ff_ps_read_data(ac->avctx, gb, &sbr->ps, *num_bits_left);
 #else
-            av_log_missing_feature(ac->avctx, "Parametric Stereo", 0);
+            avpriv_report_missing_feature(ac->avctx, "Parametric Stereo");
            skip_bits_long(gb, *num_bits_left); // bs_fill_bits
            *num_bits_left = 0;
 #endif
@@ -942,7 +942,7 @@ static void read_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
    default:
        // some files contain 0-padding
        if (bs_extension_id || *num_bits_left > 16 || show_bits(gb, *num_bits_left))
-            av_log_missing_feature(ac->avctx, "Reserved SBR extensions", 1);
+            avpriv_request_sample(ac->avctx, "Reserved SBR extensions");
        skip_bits_long(gb, *num_bits_left); // bs_fill_bits
        *num_bits_left = 0;
        break;
--- a/libavcodec/aacsbrdata.h
+++ b/libavcodec/aacsbrdata.h
@@ -352,7 +352,7 @@ static DECLARE_ALIGNED(32, float, sbr_qmf_window_us)[640] = {
     0.8537385600,
 };

-/* First two entries repeated at end to simplify SIMD implementations. */
+/* First eight entries repeated at end to simplify SIMD implementations. */
 const DECLARE_ALIGNED(16, float, ff_sbr_noise_table)[][2] = {
 {-0.99948153278296, -0.59483417516607}, { 0.97113454393991, -0.67528515225647},
 { 0.14130051758487, -0.95090983575689}, {-0.47005496701697, -0.37340549728647},
@@ -610,7 +610,11 @@ const DECLARE_ALIGNED(16, float, ff_sbr_noise_table)[][2] = {
 {-0.93412041758744,  0.41374052024363}, { 0.96063943315511,  0.93116709541280},
 { 0.97534253457837,  0.86150930812689}, { 0.99642466504163,  0.70190043427512},
 {-0.94705089665984, -0.29580042814306}, { 0.91599807087376, -0.98147830385781},
+// Start of duplicated table
 {-0.99948153278296, -0.59483417516607}, { 0.97113454393991, -0.67528515225647},
+{ 0.14130051758487, -0.95090983575689}, {-0.47005496701697, -0.37340549728647},
+{ 0.80705063769351,  0.29653668284408}, {-0.38981478896926,  0.89572605717087},
+{-0.01053049862020, -0.66959058036166}, {-0.91266367957293, -0.11522938140034},
 };

 #endif /* AVCODEC_AACSBRDATA_H */
--- a/libavcodec/aasc.c
+++ b/libavcodec/aasc.c
@@ -29,12 +29,13 @@
 #include <string.h>

 #include "avcodec.h"
+#include "internal.h"
 #include "msrledec.h"

 typedef struct AascContext {
    AVCodecContext *avctx;
    GetByteContext gb;
-    AVFrame frame;
+    AVFrame *frame;

    uint32_t palette[AVPALETTE_COUNT];
    int palette_size;
@@ -68,7 +69,10 @@ static av_cold int aasc_decode_init(AVCodecContext *avctx)
        av_log(avctx, AV_LOG_ERROR, "Unsupported bit depth: %d\n", avctx->bits_per_coded_sample);
        return -1;
    }
-    avcodec_get_frame_defaults(&s->frame);
+
+    s->frame = av_frame_alloc();
+    if (!s->frame)
+        return AVERROR(ENOMEM);

    return 0;
 }
@@ -87,12 +91,8 @@ static int aasc_decode_frame(AVCodecContext *avctx,
        return AVERROR_INVALIDDATA;
    }

-    s->frame.reference = 3;
-    s->frame.buffer_hints = FF_BUFFER_HINTS_VALID | FF_BUFFER_HINTS_PRESERVE | FF_BUFFER_HINTS_REUSABLE;
-    if ((ret = avctx->reget_buffer(avctx, &s->frame)) < 0) {
-        av_log(avctx, AV_LOG_ERROR, "reget_buffer() failed\n");
+    if ((ret = ff_reget_buffer(avctx, s->frame)) < 0)
        return ret;
-    }

    compr     = AV_RL32(buf);
    buf      += 4;
@@ -101,7 +101,7 @@ static int aasc_decode_frame(AVCodecContext *avctx,
    switch (avctx->codec_tag) {
    case MKTAG('A', 'A', 'S', '4'):
        bytestream2_init(&s->gb, buf - 4, buf_size + 4);
-        ff_msrle_decode(avctx, (AVPicture*)&s->frame, 8, &s->gb);
+        ff_msrle_decode(avctx, (AVPicture*)s->frame, 8, &s->gb);
        break;
    case MKTAG('A', 'A', 'S', 'C'):
    switch (compr) {
@@ -112,14 +112,14 @@ static int aasc_decode_frame(AVCodecContext *avctx,
                av_log(avctx, AV_LOG_ERROR, "Next line is beyond buffer bounds\n");
                break;
            }
-            memcpy(s->frame.data[0] + i*s->frame.linesize[0], buf, avctx->width * psize);
+            memcpy(s->frame->data[0] + i * s->frame->linesize[0], buf, avctx->width * psize);
            buf += stride;
            buf_size -= stride;
        }
        break;
    case 1:
        bytestream2_init(&s->gb, buf, buf_size);
-        ff_msrle_decode(avctx, (AVPicture*)&s->frame, 8, &s->gb);
+        ff_msrle_decode(avctx, (AVPicture*)s->frame, 8, &s->gb);
        break;
    default:
        av_log(avctx, AV_LOG_ERROR, "Unknown compression type %d\n", compr);
@@ -132,10 +132,11 @@ static int aasc_decode_frame(AVCodecContext *avctx,
    }

    if (avctx->pix_fmt == AV_PIX_FMT_PAL8)
-        memcpy(s->frame.data[1], s->palette, s->palette_size);
+        memcpy(s->frame->data[1], s->palette, s->palette_size);

    *got_frame = 1;
-    *(AVFrame*)data = s->frame;
+    if ((ret = av_frame_ref(data, s->frame)) < 0)
+        return ret;

    /* report that the buffer was completely consumed */
    return buf_size;
@@ -145,9 +146,7 @@ static av_cold int aasc_decode_end(AVCodecContext *avctx)
 {
    AascContext *s = avctx->priv_data;

-    /* release the last frame */
-    if (s->frame.data[0])
-        avctx->release_buffer(avctx, &s->frame);
+    av_frame_free(&s->frame);

    return 0;
 }
--- a/libavcodec/ac3dec.c
+++ b/libavcodec/ac3dec.c
@@ -429,7 +429,7 @@ static void ac3_decode_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, ma
    int end_freq   = s->end_freq[ch_index];
    uint8_t *baps  = s->bap[ch_index];
    int8_t *exps   = s->dexps[ch_index];
-    int *coeffs    = s->fixed_coeffs[ch_index];
+    int32_t *coeffs    = s->fixed_coeffs[ch_index];
    int dither     = (ch_index == CPL_CH) || s->dither_flag[ch_index];
    GetBitContext *gbc = &s->gbc;
    int freq;
@@ -874,7 +874,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
            /* check for enhanced coupling */
            if (s->eac3 && get_bits1(gbc)) {
                /* TODO: parse enhanced coupling strategy info */
-                av_log_missing_feature(s->avctx, "Enhanced coupling", 1);
+                avpriv_request_sample(s->avctx, "Enhanced coupling");
                return AVERROR_PATCHWELCOME;
            }

@@ -1375,10 +1375,8 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,

    /* get output buffer */
    frame->nb_samples = s->num_blocks * 256;
-    if ((ret = ff_get_buffer(avctx, frame)) < 0) {
-        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
        return ret;
-    }

    /* decode the audio blocks */
    channel_map = ff_ac3_dec_channel_map[s->output_mode & ~AC3_OUTPUT_LFEON][s->lfe_on];
--- a/libavcodec/ac3dec.h
+++ b/libavcodec/ac3dec.h
@@ -209,7 +209,7 @@ typedef struct AC3DecodeContext {
    float *dlyptr[AC3_MAX_CHANNELS];

 ///@name Aligned arrays
-    DECLARE_ALIGNED(16, int,   fixed_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS];       ///< fixed-point transform coefficients
+    DECLARE_ALIGNED(16, int32_t, fixed_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS];     ///< fixed-point transform coefficients
    DECLARE_ALIGNED(32, float, transform_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS];   ///< transform coefficients
    DECLARE_ALIGNED(32, float, delay)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE];             ///< delay - added to the next block
    DECLARE_ALIGNED(32, float, window)[AC3_BLOCK_SIZE];                              ///< window coefficients
--- a/libavcodec/ac3enc.c
+++ b/libavcodec/ac3enc.c
@@ -26,10 +26,9 @@
 * The simplest AC-3 encoder.
 */

-//#define ASSERT_LEVEL 2
-
 #include <stdint.h>

+#include "libavutil/attributes.h"
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
 #include "libavutil/channel_layout.h"
@@ -754,7 +753,7 @@ static void count_frame_bits_fixed(AC3EncodeContext *s)
 * Initialize bit allocation.
 * Set default parameter codes and calculate parameter values.
 */
-static void bit_alloc_init(AC3EncodeContext *s)
+static av_cold void bit_alloc_init(AC3EncodeContext *s)
 {
    int ch;

@@ -2018,6 +2017,7 @@ av_cold int ff_ac3_encode_close(AVCodecContext *avctx)
    AC3EncodeContext *s = avctx->priv_data;

    av_freep(&s->windowed_samples);
+    if (s->planar_samples)
    for (ch = 0; ch < s->channels; ch++)
        av_freep(&s->planar_samples[ch]);
    av_freep(&s->planar_samples);
@@ -2049,9 +2049,6 @@ av_cold int ff_ac3_encode_close(AVCodecContext *avctx)

    s->mdct_end(s);

-#if FF_API_OLD_ENCODE_AUDIO
-    av_freep(&avctx->coded_frame);
-#endif
    return 0;
 }

@@ -2481,14 +2478,6 @@ av_cold int ff_ac3_encode_init(AVCodecContext *avctx)
    if (ret)
        goto init_fail;

-#if FF_API_OLD_ENCODE_AUDIO
-    avctx->coded_frame= avcodec_alloc_frame();
-    if (!avctx->coded_frame) {
-        ret = AVERROR(ENOMEM);
-        goto init_fail;
-    }
-#endif
-
    ff_dsputil_init(&s->dsp, avctx);
    avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
    ff_ac3dsp_init(&s->ac3dsp, avctx->flags & CODEC_FLAG_BITEXACT);
--- a/libavcodec/adpcm.c
+++ b/libavcodec/adpcm.c
@@ -1,6 +1,18 @@
 /*
 * Copyright (c) 2001-2003 The ffmpeg Project
 *
+ * first version by Francois Revol (revol@free.fr)
+ * fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
+ *   by Mike Melanson (melanson@pcisys.net)
+ * CD-ROM XA ADPCM codec by BERO
+ * EA ADPCM decoder by Robin Kay (komadori@myrealbox.com)
+ * EA ADPCM R1/R2/R3 decoder by Peter Ross (pross@xvid.org)
+ * EA IMA EACS decoder by Peter Ross (pross@xvid.org)
+ * EA IMA SEAD decoder by Peter Ross (pross@xvid.org)
+ * EA ADPCM XAS decoder by Peter Ross (pross@xvid.org)
+ * MAXIS EA ADPCM decoder by Robert Marston (rmarston@gmail.com)
+ * THP ADPCM decoder by Marco Gerards (mgerards@xs4all.nl)
+ *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
@@ -19,7 +31,6 @@
 */
 #include "avcodec.h"
 #include "get_bits.h"
-#include "put_bits.h"
 #include "bytestream.h"
 #include "adpcm.h"
 #include "adpcm_data.h"
@@ -28,18 +39,6 @@
 /**
 * @file
 * ADPCM decoders
- * First version by Francois Revol (revol@free.fr)
- * Fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
- *   by Mike Melanson (melanson@pcisys.net)
- * CD-ROM XA ADPCM codec by BERO
- * EA ADPCM decoder by Robin Kay (komadori@myrealbox.com)
- * EA ADPCM R1/R2/R3 decoder by Peter Ross (pross@xvid.org)
- * EA IMA EACS decoder by Peter Ross (pross@xvid.org)
- * EA IMA SEAD decoder by Peter Ross (pross@xvid.org)
- * EA ADPCM XAS decoder by Peter Ross (pross@xvid.org)
- * MAXIS EA ADPCM decoder by Robert Marston (rmarston@gmail.com)
- * THP ADPCM decoder by Marco Gerards (mgerards@xs4all.nl)
- *
 * Features and limitations:
 *
 * Reference documents:
@@ -96,6 +95,7 @@ static av_cold int adpcm_decode_init(AVCodecContext * avctx)
    unsigned int max_channels = 2;

    switch(avctx->codec->id) {
+    case AV_CODEC_ID_ADPCM_DTK:
    case AV_CODEC_ID_ADPCM_EA:
        min_channels = 2;
        break;
@@ -118,10 +118,8 @@ static av_cold int adpcm_decode_init(AVCodecContext * avctx)
        c->status[0].step = c->status[1].step = 511;
        break;
    case AV_CODEC_ID_ADPCM_IMA_WAV:
-        if (avctx->bits_per_coded_sample != 4) {
-            av_log(avctx, AV_LOG_ERROR, "Only 4-bit ADPCM IMA WAV files are supported\n");
-            return -1;
-        }
+        if (avctx->bits_per_coded_sample < 2 || avctx->bits_per_coded_sample > 5)
+            return AVERROR_INVALIDDATA;
        break;
    case AV_CODEC_ID_ADPCM_IMA_APC:
        if (avctx->extradata && avctx->extradata_size >= 8) {
@@ -148,6 +146,7 @@ static av_cold int adpcm_decode_init(AVCodecContext * avctx)
        case AV_CODEC_ID_ADPCM_EA_XAS:
        case AV_CODEC_ID_ADPCM_THP:
        case AV_CODEC_ID_ADPCM_AFC:
+        case AV_CODEC_ID_ADPCM_DTK:
            avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
            break;
        case AV_CODEC_ID_ADPCM_IMA_WS:
@@ -187,6 +186,29 @@ static inline short adpcm_ima_expand_nibble(ADPCMChannelStatus *c, char nibble,
    return (short)c->predictor;
 }

+static inline int16_t adpcm_ima_wav_expand_nibble(ADPCMChannelStatus *c, GetBitContext *gb, int bps)
+{
+    int nibble, step_index, predictor, sign, delta, diff, step, shift;
+
+    shift = bps - 1;
+    nibble = get_bits_le(gb, bps),
+    step = ff_adpcm_step_table[c->step_index];
+    step_index = c->step_index + ff_adpcm_index_tables[bps - 2][nibble];
+    step_index = av_clip(step_index, 0, 88);
+
+    sign = nibble & (1 << shift);
+    delta = nibble & ((1 << shift) - 1);
+    diff = ((2 * delta + 1) * step) >> shift;
+    predictor = c->predictor;
+    if (sign) predictor -= diff;
+    else predictor += diff;
+
+    c->predictor = av_clip_int16(predictor);
+    c->step_index = step_index;
+
+    return (int16_t)c->predictor;
+}
+
 static inline int adpcm_ima_qt_expand_nibble(ADPCMChannelStatus *c, int nibble, int shift)
 {
    int step_index;
@@ -323,7 +345,7 @@ static int xa_decode(AVCodecContext *avctx, int16_t *out0, int16_t *out1,
        shift  = 12 - (in[4+i*2] & 15);
        filter = in[4+i*2] >> 4;
        if (filter >= FF_ARRAY_ELEMS(xa_adpcm_table)) {
-            av_log_ask_for_sample(avctx, "unknown XA-ADPCM filter %d\n", filter);
+            avpriv_request_sample(avctx, "unknown XA-ADPCM filter %d", filter);
            filter=0;
        }
        f0 = xa_adpcm_table[filter][0];
@@ -352,7 +374,7 @@ static int xa_decode(AVCodecContext *avctx, int16_t *out0, int16_t *out1,
        shift  = 12 - (in[5+i*2] & 15);
        filter = in[5+i*2] >> 4;
        if (filter >= FF_ARRAY_ELEMS(xa_adpcm_table)) {
-            av_log_ask_for_sample(avctx, "unknown XA-ADPCM filter %d\n", filter);
+            avpriv_request_sample(avctx, "unknown XA-ADPCM filter %d", filter);
            filter=0;
        }

@@ -550,11 +572,20 @@ static int get_nb_samples(AVCodecContext *avctx, GetByteContext *gb,
            buf_size = FFMIN(buf_size, avctx->block_align);
        nb_samples = 1 + (buf_size - 4 * ch) * 2 / ch;
        break;
-    case AV_CODEC_ID_ADPCM_IMA_WAV:
+    case AV_CODEC_ID_ADPCM_IMA_RAD:
        if (avctx->block_align > 0)
            buf_size = FFMIN(buf_size, avctx->block_align);
-        nb_samples = 1 + (buf_size - 4 * ch) / (4 * ch) * 8;
+        nb_samples = (buf_size - 4 * ch) * 2 / ch;
        break;
+    case AV_CODEC_ID_ADPCM_IMA_WAV:
+    {
+        int bsize = ff_adpcm_ima_block_sizes[avctx->bits_per_coded_sample - 2];
+        int bsamples = ff_adpcm_ima_block_samples[avctx->bits_per_coded_sample - 2];
+        if (avctx->block_align > 0)
+            buf_size = FFMIN(buf_size, avctx->block_align);
+        nb_samples = 1 + (buf_size - 4 * ch) / (bsize * ch) * bsamples;
+        break;
+    }
    case AV_CODEC_ID_ADPCM_MS:
        if (avctx->block_align > 0)
            buf_size = FFMIN(buf_size, avctx->block_align);
@@ -591,6 +622,10 @@ static int get_nb_samples(AVCodecContext *avctx, GetByteContext *gb,
        break;
    }
    case AV_CODEC_ID_ADPCM_THP:
+        if (avctx->extradata) {
+            nb_samples = buf_size / (8 * ch) * 14;
+            break;
+        }
        has_coded_samples = 1;
        bytestream2_skip(gb, 4); // channel size
        *coded_samples  = bytestream2_get_be32(gb);
@@ -603,6 +638,9 @@ static int get_nb_samples(AVCodecContext *avctx, GetByteContext *gb,
    case AV_CODEC_ID_ADPCM_XA:
        nb_samples = (buf_size / 128) * 224 / ch;
        break;
+    case AV_CODEC_ID_ADPCM_DTK:
+        nb_samples = buf_size / (16 * ch) * 28;
+        break;
    }

    /* validate coded sample count */
@@ -637,10 +675,8 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,

    /* get output buffer */
    frame->nb_samples = nb_samples;
-    if ((ret = ff_get_buffer(avctx, frame)) < 0) {
-        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
        return ret;
-    }
    samples = (short *)frame->data[0];
    samples_p = (int16_t **)frame->extended_data;

@@ -709,6 +745,23 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
            }
        }

+        if (avctx->bits_per_coded_sample != 4) {
+            int samples_per_block = ff_adpcm_ima_block_samples[avctx->bits_per_coded_sample - 2];
+            GetBitContext g;
+
+            init_get_bits8(&g, gb.buffer, bytestream2_get_bytes_left(&gb));
+            for (n = 0; n < (nb_samples - 1) / samples_per_block; n++) {
+                for (i = 0; i < avctx->channels; i++) {
+                    cs = &c->status[i];
+                    samples = &samples_p[i][1 + n * samples_per_block];
+                    for (m = 0; m < samples_per_block; m++) {
+                        samples[m] = adpcm_ima_wav_expand_nibble(cs, &g,
+                                          avctx->bits_per_coded_sample);
+                    }
+                }
+            }
+            bytestream2_skip(&gb, avctx->block_align - avctx->channels * 4);
+        } else {
        for (n = 0; n < (nb_samples - 1) / 8; n++) {
            for (i = 0; i < avctx->channels; i++) {
                cs = &c->status[i];
@@ -720,6 +773,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
                }
            }
        }
+        }
        break;
    case AV_CODEC_ID_ADPCM_4XM:
        for (i = 0; i < avctx->channels; i++)
@@ -906,6 +960,31 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
            *samples++ = adpcm_ima_oki_expand_nibble(&c->status[st], v & 0x0F);
        }
        break;
+    case AV_CODEC_ID_ADPCM_IMA_RAD:
+        for (channel = 0; channel < avctx->channels; channel++) {
+            cs = &c->status[channel];
+            cs->step_index = sign_extend(bytestream2_get_le16u(&gb), 16);
+            cs->predictor  = sign_extend(bytestream2_get_le16u(&gb), 16);
+            if (cs->step_index > 88u){
+                av_log(avctx, AV_LOG_ERROR, "ERROR: step_index[%d] = %i\n",
+                       channel, cs->step_index);
+                return AVERROR_INVALIDDATA;
+            }
+        }
+        for (n = 0; n < nb_samples / 2; n++) {
+            int byte[2];
+
+            byte[0] = bytestream2_get_byteu(&gb);
+            if (st)
+                byte[1] = bytestream2_get_byteu(&gb);
+            for(channel = 0; channel < avctx->channels; channel++) {
+                *samples++ = adpcm_ima_expand_nibble(&c->status[channel], byte[channel] & 0x0F, 3);
+            }
+            for(channel = 0; channel < avctx->channels; channel++) {
+                *samples++ = adpcm_ima_expand_nibble(&c->status[channel], byte[channel] >> 4  , 3);
+            }
+        }
+        break;
    case AV_CODEC_ID_ADPCM_IMA_WS:
        if (c->vqa_version == 3) {
            for (channel = 0; channel < avctx->channels; channel++) {
@@ -1323,6 +1402,18 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
        int table[6][16];
        int ch;

+        if (avctx->extradata) {
+            GetByteContext tb;
+            if (avctx->extradata_size < 32 * avctx->channels) {
+                av_log(avctx, AV_LOG_ERROR, "Missing coeff table\n");
+                return AVERROR_INVALIDDATA;
+            }
+
+            bytestream2_init(&tb, avctx->extradata, avctx->extradata_size);
+            for (i = 0; i < avctx->channels; i++)
+                for (n = 0; n < 16; n++)
+                    table[i][n] = sign_extend(bytestream2_get_be16u(&tb), 16);
+        } else {
        for (i = 0; i < avctx->channels; i++)
            for (n = 0; n < 16; n++)
                table[i][n] = sign_extend(bytestream2_get_be16u(&gb), 16);
@@ -1332,6 +1423,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
            c->status[i].sample1 = sign_extend(bytestream2_get_be16u(&gb), 16);
            c->status[i].sample2 = sign_extend(bytestream2_get_be16u(&gb), 16);
        }
+        }

        for (ch = 0; ch < avctx->channels; ch++) {
            samples = samples_p[ch];
@@ -1365,6 +1457,54 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
        }
        break;
    }
+    case AV_CODEC_ID_ADPCM_DTK:
+        for (channel = 0; channel < avctx->channels; channel++) {
+            samples = samples_p[channel];
+
+            /* Read in every sample for this channel.  */
+            for (i = 0; i < nb_samples / 28; i++) {
+                int byte, header;
+                if (channel)
+                    bytestream2_skipu(&gb, 1);
+                header = bytestream2_get_byteu(&gb);
+                bytestream2_skipu(&gb, 3 - channel);
+
+                /* Decode 28 samples.  */
+                for (n = 0; n < 28; n++) {
+                    int32_t sampledat, prev;
+
+                    switch (header >> 4) {
+                    case 1:
+                        prev = (c->status[channel].sample1 * 0x3c);
+                        break;
+                    case 2:
+                        prev = (c->status[channel].sample1 * 0x73) - (c->status[channel].sample2 * 0x34);
+                        break;
+                    case 3:
+                        prev = (c->status[channel].sample1 * 0x62) - (c->status[channel].sample2 * 0x37);
+                        break;
+                    default:
+                        prev = 0;
+                    }
+
+                    prev = av_clip((prev + 0x20) >> 6, -0x200000, 0x1fffff);
+
+                    byte = bytestream2_get_byteu(&gb);
+                    if (!channel)
+                        sampledat = sign_extend(byte, 4);
+                    else
+                        sampledat = sign_extend(byte >> 4, 4);
+
+                    sampledat = (((sampledat << 12) >> (header & 0xf)) << 6) + prev;
+                    *samples++ = av_clip_int16(sampledat >> 6);
+                    c->status[channel].sample2 = c->status[channel].sample1;
+                    c->status[channel].sample1 = sampledat;
+                }
+            }
+            if (!channel)
+                bytestream2_seek(&gb, 0, SEEK_SET);
+        }
+        break;

    default:
        return -1;
@@ -1406,6 +1546,7 @@ AVCodec ff_ ## name_ ## _decoder = {                        \
 ADPCM_DECODER(AV_CODEC_ID_ADPCM_4XM,         sample_fmts_s16p, adpcm_4xm,         "ADPCM 4X Movie");
 ADPCM_DECODER(AV_CODEC_ID_ADPCM_AFC,         sample_fmts_s16p, adpcm_afc,         "ADPCM Nintendo Gamecube AFC");
 ADPCM_DECODER(AV_CODEC_ID_ADPCM_CT,          sample_fmts_s16,  adpcm_ct,          "ADPCM Creative Technology");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_DTK,         sample_fmts_s16p, adpcm_dtk,         "ADPCM Nintendo Gamecube DTK");
 ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA,          sample_fmts_s16,  adpcm_ea,          "ADPCM Electronic Arts");
 ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_MAXIS_XA, sample_fmts_s16,  adpcm_ea_maxis_xa, "ADPCM Electronic Arts Maxis CDROM XA");
 ADPCM_DECODER(AV_CODEC_ID_ADPCM_EA_R1,       sample_fmts_s16p, adpcm_ea_r1,       "ADPCM Electronic Arts R1");
@@ -1421,6 +1562,7 @@ ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_EA_SEAD, sample_fmts_s16,  adpcm_ima_ea_sead
 ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_ISS,     sample_fmts_s16,  adpcm_ima_iss,     "ADPCM IMA Funcom ISS");
 ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_OKI,     sample_fmts_s16,  adpcm_ima_oki,     "ADPCM IMA Dialogic OKI");
 ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_QT,      sample_fmts_s16p, adpcm_ima_qt,      "ADPCM IMA QuickTime");
+ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_RAD,     sample_fmts_s16,  adpcm_ima_rad,     "ADPCM IMA Radical");
 ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_SMJPEG,  sample_fmts_s16,  adpcm_ima_smjpeg,  "ADPCM IMA Loki SDL MJPEG");
 ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_WAV,     sample_fmts_s16p, adpcm_ima_wav,     "ADPCM IMA WAV");
 ADPCM_DECODER(AV_CODEC_ID_ADPCM_IMA_WS,      sample_fmts_both, adpcm_ima_ws,      "ADPCM IMA Westwood");
--- a/Show More
+++ b/Show More
@@ -1 +1 @@
 .1.git
 .0.3