Update for 2.2.11

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
avcodec/motion_est: use 2x8x8 for interlaced qpel
2014-12-08 19:27:19 +01:00 · 2014-12-08 19:08:30 +01:00 · 2014-12-08 19:08:30 +01:00 · 2014-12-08 19:08:30 +01:00 · 2014-12-08 19:08:30 +01:00 · 2014-12-08 19:08:30 +01:00
2330 changed files with 162733 additions and 46133 deletions
--- a/.gitattributes
+++ b/.gitattributes
@@ -0,0 +1 @@
+*.pnm -diff -text
--- a/.gitignore
+++ b/.gitignore
@@ -27,7 +27,6 @@
 /ffserver
 /config.*
 /coverage.info
-/version.h
 /doc/*.1
 /doc/*.3
 /doc/*.html
@@ -35,23 +34,28 @@
 /doc/config.texi
 /doc/avoptions_codec.texi
 /doc/avoptions_format.texi
-/doc/examples/decoding_encoding
-/doc/examples/demuxing
+/doc/doxy/html/
+/doc/examples/avio_reading
+/doc/examples/avcodec
+/doc/examples/demuxing_decoding
+/doc/examples/filter_audio
 /doc/examples/filtering_audio
 /doc/examples/filtering_video
 /doc/examples/metadata
 /doc/examples/muxing
 /doc/examples/pc-uninstalled
+/doc/examples/remuxing
 /doc/examples/resampling_audio
 /doc/examples/scaling_video
+/doc/examples/transcode_aac
 /doc/fate.txt
-/doc/doxy/html/
 /doc/print_options
 /lcov/
 /libavcodec/*_tablegen
 /libavcodec/*_tables.c
 /libavcodec/*_tables.h
 /libavutil/avconfig.h
+/libavutil/ffversion.h
 /tests/audiogen
 /tests/base64
 /tests/data/
@@ -63,6 +67,7 @@
 /tools/aviocat
 /tools/ffbisect
 /tools/bisect.need
+/tools/crypto_bench
 /tools/cws2fws
 /tools/fourcc2pixfmt
 /tools/ffescape
--- a/103
+++ b/103
@@ -1,6 +1,99 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.

+version 2.2.7
+- snow: fix null pointer dereference
+- iff: fix out of array access
+- svq1dec: fix input data corruption
+- proresenc_ks: check buffer size
+
+
+version 2.2.6
+- fix infinite loop in dvbsub parser
+- fix some interlaced MPEG-2 videos
+- fix decoding issues in dv (Ticket2340, 2341)
+- fix v4l2 and v4l2enc crashes
+- fix theoretical librtmp crash
+- fix theoretical eamad crash
+- support dimension change in g2meet
+
+
+version 2.2:
+- HNM version 4 demuxer and video decoder
+- Live HDS muxer
+- setsar/setdar filters now support variables in ratio expressions
+- elbg filter
+- string validation in ffprobe
+- support for decoding through VDPAU in ffmpeg (the -hwaccel option)
+- complete Voxware MetaSound decoder
+- remove mp3_header_compress bitstream filter
+- Windows resource files for shared libraries
+- aeval filter
+- stereoscopic 3d metadata handling
+- WebP encoding via libwebp
+- ATRAC3+ decoder
+- VP8 in Ogg demuxing
+- side & metadata support in NUT
+- framepack filter
+- XYZ12 rawvideo support in NUT
+- Exif metadata support in WebP decoder
+- OpenGL device
+- Use metadata_header_padding to control padding in ID3 tags (currently used in
+  MP3, AIFF, and OMA files), FLAC header, and the AVI "junk" block.
+- Mirillis FIC video decoder
+- Support DNx444
+- libx265 encoder
+- dejudder filter
+- Autodetect VDA like all other hardware accelerations
+
+
+version 2.1:
+
+- aecho filter
+- perspective filter ported from libmpcodecs
+- ffprobe -show_programs option
+- compand filter
+- RTMP seek support
+- when transcoding with ffmpeg (i.e. not streamcopying), -ss is now accurate
+  even when used as an input option. Previous behavior can be restored with
+  the -noaccurate_seek option.
+- ffmpeg -t option can now be used for inputs, to limit the duration of
+  data read from an input file
+- incomplete Voxware MetaSound decoder
+- read EXIF metadata from JPEG
+- DVB teletext decoder
+- phase filter ported from libmpcodecs
+- w3fdif filter
+- Opus support in Matroska
+- FFV1 version 1.3 is stable and no longer experimental
+- FFV1: YUVA(444,422,420) 9, 10 and 16 bit support
+- changed DTS stream id in lavf mpeg ps muxer from 0x8a to 0x88, to be
+  more consistent with other muxers.
+- adelay filter
+- pullup filter ported from libmpcodecs
+- ffprobe -read_intervals option
+- Lossless and alpha support for WebP decoder
+- Error Resilient AAC syntax (ER AAC LC) decoding
+- Low Delay AAC (ER AAC LD) decoding
+- mux chapters in ASF files
+- SFTP protocol (via libssh)
+- libx264: add ability to encode in YUVJ422P and YUVJ444P
+- Fraps: use BT.709 colorspace by default for yuv, as reference fraps decoder does
+- make decoding alpha optional for prores, ffv1 and vp6 by setting
+  the skip_alpha flag.
+- ladspa wrapper filter
+- native VP9 decoder
+- dpx parser
+- max_error_rate parameter in ffmpeg
+- PulseAudio output device
+- ReplayGain scanner
+- Enhanced Low Delay AAC (ER AAC ELD) decoding (no LD SBR support)
+- Linux framebuffer output device
+- HEVC decoder
+- raw HEVC, HEVC in MOV/MP4, HEVC in Matroska, HEVC in MPEG-TS demuxing
+- mergeplanes filter
+
+
 version 2.0:

 - curves filter
@@ -13,7 +106,7 @@ version 2.0:
 - 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
+- ffmpeg options -filter_script and -filter_complex_script, which allow a
  filtergraph description to be read from a file
 - OpenCL support
 - audio phaser filter
@@ -21,7 +114,7 @@ version 2.0:
 - libquvi demuxer
 - uniform options syntax across all filters
 - telecine filter
- new interlace filter
+- interlace filter
 - smptehdbars source
 - inverse telecine filters (fieldmatch and decimate)
 - colorbalance filter
@@ -607,7 +700,7 @@ version 0.6:
 - LPCM support in MPEG-TS (HDMV RID as found on Blu-ray disks)
 - WMA Pro decoder
 - Core Audio Format demuxer
- Atrac1 decoder
+- ATRAC1 decoder
 - MD STUDIO audio demuxer
 - RF64 support in WAV demuxer
 - MPEG-4 Audio Lossless Coding (ALS) decoder
@@ -707,7 +800,7 @@ version 0.5:
 - MXF demuxer
 - VC-1/WMV3/WMV9 video decoder
 - MacIntel support
- AVISynth support
+- AviSynth support
 - VMware video decoder
 - VP5 video decoder
 - VP6 video decoder
@@ -735,7 +828,7 @@ version 0.5:
 - Interplay C93 demuxer and video decoder
 - Bethsoft VID demuxer and video decoder
 - CRYO APC demuxer
- Atrac3 decoder
+- ATRAC3 decoder
 - V.Flash PTX decoder
 - RoQ muxer, RoQ audio encoder
 - Renderware TXD demuxer and decoder
--- a/8
+++ b/8
@@ -33,20 +33,21 @@ Specifically, the GPL parts of FFmpeg are
    - vf_geq.c
    - vf_histeq.c
    - vf_hqdn3d.c
-    - vf_hue.c
+    - vf_interlace.c
    - vf_kerndeint.c
    - vf_mcdeint.c
    - vf_mp.c
-    - vf_noise.c
    - vf_owdenoise.c
+    - vf_perspective.c
+    - vf_phase.c
    - vf_pp.c
+    - vf_pullup.c
    - vf_sab.c
    - vf_smartblur.c
    - vf_spp.c
    - vf_stereo3d.c
    - vf_super2xsai.c
    - vf_tinterlace.c
-    - vf_yadif.c
    - vsrc_mptestsrc.c

 Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
@@ -79,6 +80,7 @@ The following libraries are under GPL:
    - libutvideo
    - libvidstab
    - libx264
+    - libx265
    - libxavs
    - libxvid
 When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
--- a/110
+++ b/110
@@ -31,7 +31,7 @@ ffprobe:
  ffprobe.c                             Stefano Sabatini

 ffserver:
-  ffserver.c, ffserver.h                Baptiste Coudurier
+  ffserver.c                            Reynaldo H. Verdejo Pinochet

 Commandline utility code:
  cmdutils.c, cmdutils.h                Michael Niedermayer
@@ -43,16 +43,24 @@ QuickTime faststart:
 Miscellaneous Areas
 ===================

-documentation                           Mike Melanson
-website                                 Robert Swain, Lou Logan
+documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu
 build system (configure,Makefiles)      Diego Biurrun, Mans Rullgard
-project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger
-mailinglists                            Michael Niedermayer, Baptiste Coudurier, Lou Logan
+project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger, Alexander Strasser
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
 release management                      Michael Niedermayer


+Communication
+=============
+
+website                                 Robert Swain, Lou Logan
+mailinglists                            Michael Niedermayer, Baptiste Coudurier, Lou Logan
+Google+                                 Paul B Mahol, Michael Niedermayer, Alexander Strasser
+Twitter                                 Lou Logan
+Launchpad                               Timothy Gu
+
+
 libavutil
 =========

@@ -62,13 +70,23 @@ 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
+  eval.c, eval.h                        Michael Niedermayer
+  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
+  mem.c, mem.h                          Michael Niedermayer
  opencl.c, opencl.h                    Wei Gao
+  opt.c, opt.h                          Michael Niedermayer
+  rational.c, rational.h                Michael Niedermayer
+  rc4                                   Reimar Doeffinger
  ripemd.c, ripemd.h                    James Almer
+  timecode                              Clément Bœsch


 libavcodec
@@ -79,10 +97,6 @@ Generic Parts:
    avcodec.h                           Michael Niedermayer
  utility code:
    utils.c                             Michael Niedermayer
-    mem.c                               Michael Niedermayer
-    opt.c, opt.h                        Michael Niedermayer
-  arithmetic expression evaluator:
-    eval.c                              Michael Niedermayer
  audio and video frame extraction:
    parser.c                            Michael Niedermayer
  bitstream reading:
@@ -113,6 +127,8 @@ Generic Parts:
    libpostproc/*                       Michael Niedermayer
  table generation:
    tableprint.c, tableprint.h          Reimar Doeffinger
+  fixed point FFT:
+    fft*                                Zeljko Lukac

 Codecs:
  4xm.c                                 Michael Niedermayer
@@ -126,14 +142,16 @@ Codecs:
  ass*                                  Aurelien Jacobs
  asv*                                  Michael Niedermayer
  atrac3*                               Benjamin Larsson
+  atrac3plus*                           Maxim Poliakovski
  bgmc.c, bgmc.h                        Thilo Borgmann
  bink.c                                Kostya Shishkov
  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
+  cinepakenc.c                          Rl / Aetey G.T. AB
  cljr                                  Alex Beregszaszi
  cllc.c                                Derek Buitenhuis
  cook.c, cookdata.h                    Benjamin Larsson
@@ -143,9 +161,10 @@ 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
+  exif.c, exif.h                        Thilo Borgmann
  ffv1.c                                Michael Niedermayer
  ffwavesynth.c                         Nicolas George
  flac*                                 Justin Ruggles
@@ -154,9 +173,9 @@ 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
@@ -171,8 +190,8 @@ Codecs:
  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
@@ -180,8 +199,11 @@ Codecs:
  libtheoraenc.c                        David Conrad
  libutvideo*                           Derek Buitenhuis
  libvorbis.c                           David Conrad
-  libxavs.c                             Stefan Gehrer
+  libvpx*                               James Zern
  libx264.c                             Mans Rullgard, Jason Garrett-Glaser
+  libx265.c                             Derek Buitenhuis
+  libxavs.c                             Stefan Gehrer
+  libzvbi-teletextdec.c                 Marton Balint
  loco.c                                Kostya Shishkov
  lzo.h, lzo.c                          Reimar Doeffinger
  mdec.c                                Michael Niedermayer
@@ -243,12 +265,13 @@ 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
  vp8                                   David Conrad, Jason Garrett-Glaser, Ronald Bultje
+  vp9                                   Ronald Bultje, Clément Bœsch
  vqavideo.c                            Mike Melanson
  wavpack.c                             Kostya Shishkov
  wmaprodec.c                           Sascha Sommer
@@ -257,6 +280,7 @@ Codecs:
  wnv1.c                                Kostya Shishkov
  xan.c                                 Mike Melanson
  xbm*                                  Paul B Mahol
+  xface                                 Stefano Sabatini
  xl.c                                  Kostya Shishkov
  xvmc.c                                Ivan Kalvachev
  xwd*                                  Paul B Mahol
@@ -278,11 +302,16 @@ libavdevice
    libavdevice/avdevice.h


+  dshow.c                               Roger Pack
+  fbdev_enc.c                           Lukasz Marek
  iec61883.c                            Georg Lippitsch
+  lavfi                                 Stefano Sabatini
  libdc1394.c                           Roman Shaposhnik
+  opengl_enc.c                          Lukasz Marek
+  pulse_audio_enc.c                     Lukasz Marek
+  sdl                                   Stefano Sabatini
  v4l2.c                                Luca Abeni
  vfwcap.c                              Ramiro Polla
-  dshow.c                               Roger Pack

 libavfilter
 ===========
@@ -291,12 +320,34 @@ Generic parts:
  graphdump.c                           Nicolas George

 Filters:
+  af_adelay.c                           Paul B Mahol
+  af_aecho.c                            Paul B Mahol
+  af_afade.c                            Paul B Mahol
  af_amerge.c                           Nicolas George
+  af_aphaser.c                          Paul B Mahol
+  af_aresample.c                        Michael Niedermayer
+  af_astats.c                           Paul B Mahol
  af_astreamsync.c                      Nicolas George
  af_atempo.c                           Pavel Koshevoy
+  af_biquads.c                          Paul B Mahol
+  af_compand.c                          Paul B Mahol
+  af_ladspa.c                           Paul B Mahol
  af_pan.c                              Nicolas George
+  avf_avectorscope.c                    Paul B Mahol
+  vf_blend.c                            Paul B Mahol
+  vf_colorbalance.c                     Paul B Mahol
+  vf_dejudder.c                         Nicholas Robbins
  vf_delogo.c                           Jean Delvare (CC <khali@linux-fr.org>)
  vf_drawbox.c/drawgrid                 Andrey Utkin
+  vf_extractplanes.c                    Paul B Mahol
+  vf_histogram.c                        Paul B Mahol
+  vf_il.c                               Paul B Mahol
+  vf_mergeplanes.c                      Paul B Mahol
+  vf_psnr.c                             Paul B Mahol
+  vf_scale.c                            Michael Niedermayer
+  vf_separatefields.c                   Paul B Mahol
+  vf_stereo3d.c                         Paul B Mahol
+  vf_telecine.c                         Paul B Mahol
  vf_yadif.c                            Michael Niedermayer

 Sources:
@@ -316,7 +367,8 @@ 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
@@ -341,11 +393,12 @@ Muxers/Demuxers:
  flvdec.c, flvenc.c                    Michael Niedermayer
  gxf.c                                 Reimar Doeffinger
  gxfenc.c                              Baptiste Coudurier
+  hls.c                                 Anssi Hannula
  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
@@ -359,11 +412,11 @@ 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
@@ -400,6 +453,7 @@ Muxers/Demuxers:
  siff.c                                Kostya Shishkov
  smacker.c                             Kostya Shishkov
  smjpeg*                               Paul B Mahol
+  spdif*                                Anssi Hannula
  srtdec.c                              Aurelien Jacobs
  swf.c                                 Baptiste Coudurier
  takdec.c                              Paul B Mahol
@@ -418,6 +472,7 @@ Protocols:
  bluray.c                              Petri Hintukainen
  ftp.c                                 Lukasz Marek
  http.c                                Ronald S. Bultje
+  libssh.c                              Lukasz Marek
  mms*.c                                Ronald S. Bultje
  udp.c                                 Luca Abeni

@@ -442,7 +497,7 @@ Operating systems / CPU architectures
 Alpha                                   Mans Rullgard, Falk Hueffner
 ARM                                     Mans Rullgard
 AVR32                                   Mans Rullgard
-MIPS                                    Mans Rullgard
+MIPS                                    Mans Rullgard, Nedeljko Babic
 Mac OS X / PowerPC                      Romain Dolbeau, Guillaume Poirier
 Amiga / PowerPC                         Colin Ward
 Linux / PowerPC                         Luca Barbato
@@ -456,9 +511,9 @@ x86                                     Michael Niedermayer
 Releases
 ========

-2.0                                     Michael Niedermayer
+2.2                                     Michael Niedermayer
+2.1                                     Michael Niedermayer
 1.2                                     Michael Niedermayer
-1.1                                     Michael Niedermayer

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

@@ -466,6 +521,7 @@ If you want to maintain an older release, please contact us
 GnuPG Fingerprints of maintainers and contributors
 ==================================================

+Alexander Strasser            1C96 78B7 83CB 8AA7 9AF5 D1EB A7D8 A57B A876 E58F
 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
--- a/68
+++ b/68
@@ -6,29 +6,34 @@ vpath %.cpp  $(SRC_PATH)
 vpath %.h    $(SRC_PATH)
 vpath %.S    $(SRC_PATH)
 vpath %.asm  $(SRC_PATH)
+vpath %.rc   $(SRC_PATH)
 vpath %.v    $(SRC_PATH)
 vpath %.texi $(SRC_PATH)
 vpath %/fate_config.sh.template $(SRC_PATH)

-PROGS-$(CONFIG_FFMPEG)   += ffmpeg
-PROGS-$(CONFIG_FFPLAY)   += ffplay
-PROGS-$(CONFIG_FFPROBE)  += ffprobe
-PROGS-$(CONFIG_FFSERVER) += ffserver
+AVPROGS-$(CONFIG_FFMPEG)   += ffmpeg
+AVPROGS-$(CONFIG_FFPLAY)   += ffplay
+AVPROGS-$(CONFIG_FFPROBE)  += ffprobe
+AVPROGS-$(CONFIG_FFSERVER) += ffserver

-PROGS      := $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
-INSTPROGS   = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+AVPROGS    := $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+INSTPROGS   = $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+PROGS      += $(AVPROGS)

-OBJS        = cmdutils.o $(EXEOBJS)
-OBJS-ffmpeg = ffmpeg_opt.o ffmpeg_filter.o
+AVBASENAMES  = ffmpeg ffplay ffprobe ffserver
+ALLAVPROGS   = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF))
+ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
+
+$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog) += cmdutils.o))
+$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog)-$(CONFIG_OPENCL) += cmdutils_opencl.o))
+
+OBJS-ffmpeg                   += ffmpeg_opt.o ffmpeg_filter.o
+OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o
 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
 HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
-TOOLS       = qt-faststart trasher
+TOOLS       = qt-faststart trasher uncoded_frame
 TOOLS-$(CONFIG_ZLIB) += cws2fws

-BASENAMES   = ffmpeg ffplay ffprobe ffserver
-ALLPROGS    = $(BASENAMES:%=%$(PROGSSUF)$(EXESUF))
-ALLPROGS_G  = $(BASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
-
 FFLIBS-$(CONFIG_AVDEVICE) += avdevice
 FFLIBS-$(CONFIG_AVFILTER) += avfilter
 FFLIBS-$(CONFIG_AVFORMAT) += avformat
@@ -50,16 +55,14 @@ include $(SRC_PATH)/common.mak
 FF_EXTRALIBS := $(FFEXTRALIBS)
 FF_DEP_LIBS  := $(DEP_LIBS)

-all: $(PROGS)
-
-$(PROGS): %$(EXESUF): %_g$(EXESUF)
-	$(CP) $< $@
-	$(STRIP) $@
+all: $(AVPROGS)

 $(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
 	$(LD) $(LDFLAGS) $(LD_O) $^ $(ELIBS)

 tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
+tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
+tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)

 config.h: .config
 .config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
@@ -73,7 +76,7 @@ SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS      \
               ALTIVEC-OBJS VIS-OBJS                                     \
               MMX-OBJS YASM-OBJS                                        \
               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MIPS32R2-OBJS  \
-               OBJS HOSTOBJS TESTOBJS
+               OBJS SLIBOBJS HOSTOBJS TESTOBJS

 define RESET
 $(1) :=
@@ -90,8 +93,10 @@ endef

 $(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))

+include $(SRC_PATH)/doc/Makefile
+
 define DOPROG
-OBJS-$(1) += $(1).o cmdutils.o $(EXEOBJS)
+OBJS-$(1) += $(1).o $(EXEOBJS) $(OBJS-$(1)-yes)
 $(1)$(PROGSSUF)_g$(EXESUF): $$(OBJS-$(1))
 $$(OBJS-$(1)): CFLAGS  += $(CFLAGS-$(1))
 $(1)$(PROGSSUF)_g$(EXESUF): LDFLAGS += $(LDFLAGS-$(1))
@@ -99,7 +104,13 @@ $(1)$(PROGSSUF)_g$(EXESUF): FF_EXTRALIBS += $(LIBS-$(1))
 -include $$(OBJS-$(1):.o=.d)
 endef

-$(foreach P,$(PROGS-yes),$(eval $(call DOPROG,$(P))))
+$(foreach P,$(PROGS),$(eval $(call DOPROG,$(P:$(PROGSSUF)$(EXESUF)=))))
+
+ffprobe.o cmdutils.o : libavutil/ffversion.h
+
+$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
+	$(CP) $< $@
+	$(STRIP) $@

 %$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
 	$(LD) $(LDFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
@@ -114,14 +125,14 @@ GIT_LOG     = $(SRC_PATH)/.git/logs/HEAD
 .version: $(wildcard $(GIT_LOG)) $(VERSION_SH) config.mak
 .version: M=@

-version.h .version:
-	$(M)$(VERSION_SH) $(SRC_PATH) version.h $(EXTRA_VERSION)
+libavutil/ffversion.h .version:
+	$(M)$(VERSION_SH) $(SRC_PATH) libavutil/ffversion.h $(EXTRA_VERSION)
 	$(Q)touch .version

 # force version.sh to run whenever version might have changed
 -include .version

-ifdef PROGS
+ifdef AVPROGS
 install: install-progs install-data
 endif

@@ -132,7 +143,7 @@ install-libs: install-libs-yes
 install-progs-yes:
 install-progs-$(CONFIG_SHARED): install-libs

-install-progs: install-progs-yes $(PROGS)
+install-progs: install-progs-yes $(AVPROGS)
 	$(Q)mkdir -p "$(BINDIR)"
 	$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"

@@ -144,13 +155,13 @@ install-data: $(DATA_FILES) $(EXAMPLES_FILES)
 uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data

 uninstall-progs:
-	$(RM) $(addprefix "$(BINDIR)/", $(ALLPROGS))
+	$(RM) $(addprefix "$(BINDIR)/", $(ALLAVPROGS))

 uninstall-data:
 	$(RM) -r "$(DATADIR)"

 clean::
-	$(RM) $(ALLPROGS) $(ALLPROGS_G)
+	$(RM) $(ALLAVPROGS) $(ALLAVPROGS_G)
 	$(RM) $(CLEANSUFFIXES)
 	$(RM) $(CLEANSUFFIXES:%=tools/%)
 	$(RM) -r coverage-html
@@ -158,14 +169,13 @@ clean::

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

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

 check: all alltools examples testprogs fate

-include $(SRC_PATH)/doc/Makefile
 include $(SRC_PATH)/tests/Makefile

 $(sort $(OBJDIRS)):
--- a/2
+++ b/2
@@ -1 +1 @@
-2.0
+2.2.11
--- a/1
+++ b/1
@@ -1 +0,0 @@
-2.0
--- a/cmdutils.c
+++ b/cmdutils.c
@@ -20,6 +20,7 @@
 */

 #include <string.h>
+#include <stdint.h>
 #include <stdlib.h>
 #include <errno.h>
 #include <math.h>
@@ -47,8 +48,9 @@
 #include "libavutil/eval.h"
 #include "libavutil/dict.h"
 #include "libavutil/opt.h"
+#include "libavutil/cpu.h"
+#include "libavutil/ffversion.h"
 #include "cmdutils.h"
-#include "version.h"
 #if CONFIG_NETWORK
 #include "libavformat/network.h"
 #endif
@@ -56,10 +58,6 @@
 #include <sys/time.h>
 #include <sys/resource.h>
 #endif
-#if CONFIG_OPENCL
-#include "libavutil/opencl.h"
-#endif
-

 static int init_report(const char *env);

@@ -67,9 +65,8 @@ struct SwsContext *sws_opts;
 AVDictionary *swr_opts;
 AVDictionary *format_opts, *codec_opts, *resample_opts;

-const int this_year = 2013;
-
 static FILE *report_file;
+int hide_banner = 0;

 void init_opts(void)
 {
@@ -213,7 +210,10 @@ static const OptionDef *find_option(const OptionDef *po, const char *name)
    return po;
 }

-#if HAVE_COMMANDLINETOARGVW
+/* _WIN32 means using the windows libc - cygwin doesn't define that
+ * by default. HAVE_COMMANDLINETOARGVW is true on cygwin, while
+ * it doesn't provide the actual command line via GetCommandLineW(). */
+#if HAVE_COMMANDLINETOARGVW && defined(_WIN32)
 #include <windows.h>
 #include <shellapi.h>
 /* Will be leaked on exit */
@@ -492,6 +492,18 @@ void parse_loglevel(int argc, char **argv, const OptionDef *options)
            fflush(report_file);
        }
    }
+    idx = locate_option(argc, argv, options, "hide_banner");
+    if (idx)
+        hide_banner = 1;
+}
+
+static const AVOption *opt_find(void *obj, const char *name, const char *unit,
+                            int opt_flags, int search_flags)
+{
+    const AVOption *o = av_opt_find(obj, name, unit, opt_flags, search_flags);
+    if(o && !o->flags)
+        return NULL;
+    return o;
 }

 #define FLAGS (o->type == AV_OPT_TYPE_FLAGS) ? AV_DICT_APPEND : 0
@@ -514,14 +526,14 @@ int opt_default(void *optctx, const char *opt, const char *arg)
        p = opt + strlen(opt);
    av_strlcpy(opt_stripped, opt, FFMIN(sizeof(opt_stripped), p - opt + 1));

-    if ((o = av_opt_find(&cc, opt_stripped, NULL, 0,
+    if ((o = opt_find(&cc, opt_stripped, NULL, 0,
                         AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ)) ||
        ((opt[0] == 'v' || opt[0] == 'a' || opt[0] == 's') &&
-         (o = av_opt_find(&cc, opt + 1, NULL, 0, AV_OPT_SEARCH_FAKE_OBJ)))) {
+         (o = opt_find(&cc, opt + 1, NULL, 0, AV_OPT_SEARCH_FAKE_OBJ)))) {
        av_dict_set(&codec_opts, opt, arg, FLAGS);
        consumed = 1;
    }
-    if ((o = av_opt_find(&fc, opt, NULL, 0,
+    if ((o = opt_find(&fc, opt, NULL, 0,
                         AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
        av_dict_set(&format_opts, opt, arg, FLAGS);
        if (consumed)
@@ -530,7 +542,7 @@ int opt_default(void *optctx, const char *opt, const char *arg)
    }
 #if CONFIG_SWSCALE
    sc = sws_get_class();
-    if (!consumed && av_opt_find(&sc, opt, NULL, 0,
+    if (!consumed && opt_find(&sc, opt, NULL, 0,
                         AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ)) {
        // XXX we only support sws_flags, not arbitrary sws options
        int ret = av_opt_set(sws_opts, opt, arg, 0);
@@ -543,7 +555,7 @@ int opt_default(void *optctx, const char *opt, const char *arg)
 #endif
 #if CONFIG_SWRESAMPLE
    swr_class = swr_get_class();
-    if (!consumed && (o=av_opt_find(&swr_class, opt, NULL, 0,
+    if (!consumed && (o=opt_find(&swr_class, opt, NULL, 0,
                                    AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
        struct SwrContext *swr = swr_alloc();
        int ret = av_opt_set(swr, opt, arg, 0);
@@ -557,7 +569,7 @@ int opt_default(void *optctx, const char *opt, const char *arg)
    }
 #endif
 #if CONFIG_AVRESAMPLE
-    if ((o=av_opt_find(&rc, opt, NULL, 0,
+    if ((o=opt_find(&rc, opt, NULL, 0,
                       AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
        av_dict_set(&resample_opts, opt, arg, FLAGS);
        consumed = 1;
@@ -796,6 +808,18 @@ do {                                                                           \
    return 0;
 }

+int opt_cpuflags(void *optctx, const char *opt, const char *arg)
+{
+    int ret;
+    unsigned flags = av_get_cpu_flags();
+
+    if ((ret = av_parse_cpu_caps(&flags, arg)) < 0)
+        return ret;
+
+    av_force_cpu_flags(flags);
+    return 0;
+}
+
 int opt_loglevel(void *optctx, const char *opt, const char *arg)
 {
    const struct { const char *name; int level; } log_levels[] = {
@@ -948,18 +972,6 @@ int opt_max_alloc(void *optctx, const char *opt, const char *arg)
    return 0;
 }

-int opt_cpuflags(void *optctx, const char *opt, const char *arg)
-{
-    int ret;
-    unsigned flags = av_get_cpu_flags();
-
-    if ((ret = av_parse_cpu_caps(&flags, arg)) < 0)
-        return ret;
-
-    av_force_cpu_flags(flags);
-    return 0;
-}
-
 int opt_timelimit(void *optctx, const char *opt, const char *arg)
 {
 #if HAVE_SETRLIMIT
@@ -973,26 +985,6 @@ 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];
@@ -1058,7 +1050,7 @@ static void print_program_info(int flags, int level)
    av_log(NULL, level, "%s version " FFMPEG_VERSION, program_name);
    if (flags & SHOW_COPYRIGHT)
        av_log(NULL, level, " Copyright (c) %d-%d the FFmpeg developers",
-               program_birth_year, this_year);
+               program_birth_year, CONFIG_THIS_YEAR);
    av_log(NULL, level, "\n");
    av_log(NULL, level, "%sbuilt on %s %s with %s\n",
           indent, __DATE__, __TIME__, CC_IDENT);
@@ -1066,10 +1058,36 @@ static void print_program_info(int flags, int level)
    av_log(NULL, level, "%sconfiguration: " FFMPEG_CONFIGURATION "\n", indent);
 }

+static void print_buildconf(int flags, int level)
+{
+    const char *indent = flags & INDENT ? "  " : "";
+    char str[] = { FFMPEG_CONFIGURATION };
+    char *conflist, *remove_tilde, *splitconf;
+
+    // Change all the ' --' strings to '~--' so that
+    // they can be identified as tokens.
+    while ((conflist = strstr(str, " --")) != NULL) {
+        strncpy(conflist, "~--", 3);
+    }
+
+    // Compensate for the weirdness this would cause
+    // when passing 'pkg-config --static'.
+    while ((remove_tilde = strstr(str, "pkg-config~")) != NULL) {
+        strncpy(remove_tilde, "pkg-config ", 11);
+    }
+
+    splitconf = strtok(str, "~");
+    av_log(NULL, level, "\n%sconfiguration:\n", indent);
+    while (splitconf != NULL) {
+        av_log(NULL, level, "%s%s%s\n", indent, indent, splitconf);
+        splitconf = strtok(NULL, "~");
+    }
+}
+
 void show_banner(int argc, char **argv, const OptionDef *options)
 {
    int idx = locate_option(argc, argv, options, "version");
-    if (idx)
+    if (hide_banner || idx)
        return;

    print_program_info (INDENT|SHOW_COPYRIGHT, AV_LOG_INFO);
@@ -1086,6 +1104,14 @@ int show_version(void *optctx, const char *opt, const char *arg)
    return 0;
 }

+int show_buildconf(void *optctx, const char *opt, const char *arg)
+{
+    av_log_set_callback(log_callback_help);
+    print_buildconf      (INDENT|0, AV_LOG_INFO);
+
+    return 0;
+}
+
 int show_license(void *optctx, const char *opt, const char *arg)
 {
 #if CONFIG_NONFREE
@@ -1469,8 +1495,9 @@ int show_filters(void *optctx, const char *opt, const char *arg)
    const AVFilterPad *pad;

    printf("Filters:\n"
-           "  T. = Timeline support\n"
-           "  .S = Slice threading\n"
+           "  T.. = Timeline support\n"
+           "  .S. = Slice threading\n"
+           "  ..C = Commmand support\n"
           "  A = Audio input/output\n"
           "  V = Video input/output\n"
           "  N = Dynamic number and/or type of input/output\n"
@@ -1494,15 +1521,30 @@ int show_filters(void *optctx, const char *opt, const char *arg)
                                  ( i && (filter->flags & AVFILTER_FLAG_DYNAMIC_OUTPUTS))) ? 'N' : '|';
        }
        *descr_cur = 0;
-        printf(" %c%c %-16s %-10s %s\n",
+        printf(" %c%c%c %-16s %-10s %s\n",
               filter->flags & AVFILTER_FLAG_SUPPORT_TIMELINE ? 'T' : '.',
               filter->flags & AVFILTER_FLAG_SLICE_THREADS    ? 'S' : '.',
+               filter->process_command                        ? 'C' : '.',
               filter->name, descr, filter->description);
    }
 #endif
    return 0;
 }

+int show_colors(void *optctx, const char *opt, const char *arg)
+{
+    const char *name;
+    const uint8_t *rgb;
+    int i;
+
+    printf("%-32s #RRGGBB\n", "name");
+
+    for (i = 0; name = av_get_known_color_name(i, &rgb); i++)
+        printf("%-32s #%02x%02x%02x\n", name, rgb[0], rgb[1], rgb[2]);
+
+    return 0;
+}
+
 int show_pix_fmts(void *optctx, const char *opt, const char *arg)
 {
    const AVPixFmtDescriptor *pix_desc = NULL;
@@ -1762,7 +1804,7 @@ int read_yesno(void)
 int cmdutils_read_file(const char *filename, char **bufptr, size_t *size)
 {
    int ret;
-    FILE *f = fopen(filename, "rb");
+    FILE *f = av_fopen_utf8(filename, "rb");

    if (!f) {
        av_log(NULL, AV_LOG_ERROR, "Cannot read file '%s': %s\n", filename,
--- a/cmdutils.h
+++ b/cmdutils.h
@@ -43,16 +43,12 @@ extern const char program_name[];
 */
 extern const int program_birth_year;

-/**
- * this year, defined by the program for show_banner()
- */
-extern const int this_year;
-
 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
 extern struct SwsContext *sws_opts;
 extern AVDictionary *swr_opts;
 extern AVDictionary *format_opts, *codec_opts, *resample_opts;
+extern int hide_banner;

 /**
 * Register a program-specific cleanup routine.
@@ -81,6 +77,11 @@ void uninit_opts(void);
 */
 void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);

+/**
+ * Override the cpuflags.
+ */
+int opt_cpuflags(void *optctx, const char *opt, const char *arg);
+
 /**
 * Fallback for options that are not explicitly handled, these will be
 * parsed through AVOptions.
@@ -96,12 +97,14 @@ int opt_report(const char *opt);

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

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

+#if CONFIG_OPENCL
 int opt_opencl(void *optctx, const char *opt, const char *arg);

+int opt_opencl_bench(void *optctx, const char *opt, const char *arg);
+#endif
+
 /**
 * Limit the execution time.
 */
@@ -411,6 +414,13 @@ void show_banner(int argc, char **argv, const OptionDef *options);
 */
 int show_version(void *optctx, const char *opt, const char *arg);

+/**
+ * Print the build configuration of the program to stdout. The contents
+ * depend on the definition of FFMPEG_CONFIGURATION.
+ * This option processing function does not utilize the arguments.
+ */
+int show_buildconf(void *optctx, const char *opt, const char *arg);
+
 /**
 * Print the license of the program to stdout. The license depends on
 * the license of the libraries compiled into the program.
@@ -485,6 +495,12 @@ int show_layouts(void *optctx, const char *opt, const char *arg);
 */
 int show_sample_fmts(void *optctx, const char *opt, const char *arg);

+/**
+ * Print a listing containing all the color names and values recognized
+ * by the program.
+ */
+int show_colors(void *optctx, const char *opt, const char *arg);
+
 /**
 * Return a positive value if a line read from standard input
 * starts with [yY], otherwise return 0.
@@ -498,7 +514,7 @@ int read_yesno(void);
 * @param filename file to read from
 * @param bufptr location where pointer to buffer is returned
 * @param size   location where size of buffer is returned
- * @return 0 in case of success, a negative value corresponding to an
+ * @return >= 0 in case of success, a negative value corresponding to an
 * AVERROR error code in case of failure.
 */
 int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
--- a/cmdutils_common_opts.h
+++ b/cmdutils_common_opts.h
@@ -4,6 +4,7 @@
    { "help"       , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
    { "-help"      , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
    { "version"    , OPT_EXIT, {.func_arg = show_version},      "show version" },
+    { "buildconf"  , OPT_EXIT, {.func_arg = show_buildconf},    "show build configuration" },
    { "formats"    , OPT_EXIT, {.func_arg = show_formats  },    "show available formats" },
    { "codecs"     , OPT_EXIT, {.func_arg = show_codecs   },    "show available codecs" },
    { "decoders"   , OPT_EXIT, {.func_arg = show_decoders },    "show available decoders" },
@@ -14,11 +15,14 @@
    { "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" },
+    { "colors"     , OPT_EXIT, {.func_arg = show_colors },      "show available color names" },
    { "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" },
+    { "cpuflags"   , HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags }, "force specific cpu flags", "flags" },
+    { "hide_banner", OPT_BOOL | OPT_EXPERT, {&hide_banner},     "do not show program banner", "hide_banner" },
 #if CONFIG_OPENCL
+    { "opencl_bench", OPT_EXIT, {.func_arg = opt_opencl_bench}, "run benchmark on all OpenCL devices and show results" },
    { "opencl_options", HAS_ARG, {.func_arg = opt_opencl},      "set OpenCL environment options" },
 #endif
--- a/cmdutils_opencl.c
+++ b/cmdutils_opencl.c
@@ -0,0 +1,274 @@
+/*
+ * Copyright (C) 2013 Lenny Wang
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/opt.h"
+#include "libavutil/time.h"
+#include "libavutil/log.h"
+#include "libavutil/opencl.h"
+#include "cmdutils.h"
+
+typedef struct {
+    int platform_idx;
+    int device_idx;
+    char device_name[64];
+    int64_t runtime;
+} OpenCLDeviceBenchmark;
+
+const char *ocl_bench_source = AV_OPENCL_KERNEL(
+inline unsigned char clip_uint8(int a)
+{
+    if (a & (~0xFF))
+        return (-a)>>31;
+    else
+        return a;
+}
+
+kernel void unsharp_bench(
+                    global unsigned char *src,
+                    global unsigned char *dst,
+                    global int *mask,
+                    int width,
+                    int height)
+{
+    int i, j, local_idx, lc_idx, sum = 0;
+    int2 thread_idx, block_idx, global_idx, lm_idx;
+    thread_idx.x = get_local_id(0);
+    thread_idx.y = get_local_id(1);
+    block_idx.x = get_group_id(0);
+    block_idx.y = get_group_id(1);
+    global_idx.x = get_global_id(0);
+    global_idx.y = get_global_id(1);
+    local uchar data[32][32];
+    local int lc[128];
+
+    for (i = 0; i <= 1; i++) {
+        lm_idx.y = -8 + (block_idx.y + i) * 16 + thread_idx.y;
+        lm_idx.y = lm_idx.y < 0 ? 0 : lm_idx.y;
+        lm_idx.y = lm_idx.y >= height ? height - 1: lm_idx.y;
+        for (j = 0; j <= 1; j++) {
+            lm_idx.x = -8 + (block_idx.x + j) * 16 + thread_idx.x;
+            lm_idx.x = lm_idx.x < 0 ? 0 : lm_idx.x;
+            lm_idx.x = lm_idx.x >= width ? width - 1: lm_idx.x;
+            data[i*16 + thread_idx.y][j*16 + thread_idx.x] = src[lm_idx.y*width + lm_idx.x];
+        }
+    }
+    local_idx = thread_idx.y*16 + thread_idx.x;
+    if (local_idx < 128)
+        lc[local_idx] = mask[local_idx];
+    barrier(CLK_LOCAL_MEM_FENCE);
+
+    \n#pragma unroll\n
+    for (i = -4; i <= 4; i++) {
+        lm_idx.y = 8 + i + thread_idx.y;
+        \n#pragma unroll\n
+        for (j = -4; j <= 4; j++) {
+            lm_idx.x = 8 + j + thread_idx.x;
+            lc_idx = (i + 4)*8 + j + 4;
+            sum += (int)data[lm_idx.y][lm_idx.x] * lc[lc_idx];
+        }
+    }
+    int temp = (int)data[thread_idx.y + 8][thread_idx.x + 8];
+    int res = temp + (((temp - (int)((sum + 1<<15) >> 16))) >> 16);
+    if (global_idx.x < width && global_idx.y < height)
+        dst[global_idx.x + global_idx.y*width] = clip_uint8(res);
+}
+);
+
+#define OCLCHECK(method, ... )                                                 \
+do {                                                                           \
+    status = method(__VA_ARGS__);                                              \
+    if (status != CL_SUCCESS) {                                                \
+        av_log(NULL, AV_LOG_ERROR, # method " error '%s'\n",                   \
+               av_opencl_errstr(status));                                      \
+        ret = AVERROR_EXTERNAL;                                                \
+        goto end;                                                              \
+    }                                                                          \
+} while (0)
+
+#define CREATEBUF(out, flags, size)                                            \
+do {                                                                           \
+    out = clCreateBuffer(ext_opencl_env->context, flags, size, NULL, &status); \
+    if (status != CL_SUCCESS) {                                                \
+        av_log(NULL, AV_LOG_ERROR, "Could not create OpenCL buffer\n");        \
+        ret = AVERROR_EXTERNAL;                                                \
+        goto end;                                                              \
+    }                                                                          \
+} while (0)
+
+static void fill_rand_int(int *data, int n)
+{
+    int i;
+    srand(av_gettime());
+    for (i = 0; i < n; i++)
+        data[i] = rand();
+}
+
+#define OPENCL_NB_ITER 5
+static int64_t run_opencl_bench(AVOpenCLExternalEnv *ext_opencl_env)
+{
+    int i, arg = 0, width = 1920, height = 1088;
+    int64_t start, ret = 0;
+    cl_int status;
+    size_t kernel_len;
+    char *inbuf;
+    int *mask;
+    int buf_size = width * height * sizeof(char);
+    int mask_size = sizeof(uint32_t) * 128;
+
+    cl_mem cl_mask, cl_inbuf, cl_outbuf;
+    cl_kernel kernel = NULL;
+    cl_program program = NULL;
+    size_t local_work_size_2d[2] = {16, 16};
+    size_t global_work_size_2d[2] = {(size_t)width, (size_t)height};
+
+    if (!(inbuf = av_malloc(buf_size)) || !(mask = av_malloc(mask_size))) {
+        av_log(NULL, AV_LOG_ERROR, "Out of memory\n");
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+    fill_rand_int((int*)inbuf, buf_size/4);
+    fill_rand_int(mask, mask_size/4);
+
+    CREATEBUF(cl_mask, CL_MEM_READ_ONLY, mask_size);
+    CREATEBUF(cl_inbuf, CL_MEM_READ_ONLY, buf_size);
+    CREATEBUF(cl_outbuf, CL_MEM_READ_WRITE, buf_size);
+
+    kernel_len = strlen(ocl_bench_source);
+    program = clCreateProgramWithSource(ext_opencl_env->context, 1, &ocl_bench_source,
+                                        &kernel_len, &status);
+    if (status != CL_SUCCESS || !program) {
+        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark program\n");
+        ret = AVERROR_EXTERNAL;
+        goto end;
+    }
+    status = clBuildProgram(program, 1, &(ext_opencl_env->device_id), NULL, NULL, NULL);
+    if (status != CL_SUCCESS) {
+        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to build benchmark program\n");
+        ret = AVERROR_EXTERNAL;
+        goto end;
+    }
+    kernel = clCreateKernel(program, "unsharp_bench", &status);
+    if (status != CL_SUCCESS) {
+        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark kernel\n");
+        ret = AVERROR_EXTERNAL;
+        goto end;
+    }
+
+    OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_inbuf, CL_TRUE, 0,
+             buf_size, inbuf, 0, NULL, NULL);
+    OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_mask, CL_TRUE, 0,
+             mask_size, mask, 0, NULL, NULL);
+    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_inbuf);
+    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_outbuf);
+    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_mask);
+    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &width);
+    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &height);
+
+    start = av_gettime();
+    for (i = 0; i < OPENCL_NB_ITER; i++)
+        OCLCHECK(clEnqueueNDRangeKernel, ext_opencl_env->command_queue, kernel, 2, NULL,
+                 global_work_size_2d, local_work_size_2d, 0, NULL, NULL);
+    clFinish(ext_opencl_env->command_queue);
+    ret = (av_gettime() - start)/OPENCL_NB_ITER;
+end:
+    if (kernel)
+        clReleaseKernel(kernel);
+    if (program)
+        clReleaseProgram(program);
+    if (cl_inbuf)
+        clReleaseMemObject(cl_inbuf);
+    if (cl_outbuf)
+        clReleaseMemObject(cl_outbuf);
+    if (cl_mask)
+        clReleaseMemObject(cl_mask);
+    av_free(inbuf);
+    av_free(mask);
+    return ret;
+}
+
+static int compare_ocl_device_desc(const void *a, const void *b)
+{
+    return ((OpenCLDeviceBenchmark*)a)->runtime - ((OpenCLDeviceBenchmark*)b)->runtime;
+}
+
+int opt_opencl_bench(void *optctx, const char *opt, const char *arg)
+{
+    int i, j, nb_devices = 0, count = 0;
+    int64_t score = 0;
+    AVOpenCLDeviceList *device_list;
+    AVOpenCLDeviceNode *device_node = NULL;
+    OpenCLDeviceBenchmark *devices = NULL;
+    cl_platform_id platform;
+
+    av_opencl_get_device_list(&device_list);
+    for (i = 0; i < device_list->platform_num; i++)
+        nb_devices += device_list->platform_node[i]->device_num;
+    if (!nb_devices) {
+        av_log(NULL, AV_LOG_ERROR, "No OpenCL device detected!\n");
+        return AVERROR(EINVAL);
+    }
+    if (!(devices = av_malloc_array(nb_devices, sizeof(OpenCLDeviceBenchmark)))) {
+        av_log(NULL, AV_LOG_ERROR, "Could not allocate buffer\n");
+        return AVERROR(ENOMEM);
+    }
+
+    for (i = 0; i < device_list->platform_num; i++) {
+        for (j = 0; j < device_list->platform_node[i]->device_num; j++) {
+            device_node = device_list->platform_node[i]->device_node[j];
+            platform = device_list->platform_node[i]->platform_id;
+            score = av_opencl_benchmark(device_node, platform, run_opencl_bench);
+            if (score > 0) {
+                devices[count].platform_idx = i;
+                devices[count].device_idx = j;
+                devices[count].runtime = score;
+                strcpy(devices[count].device_name, device_node->device_name);
+                count++;
+            }
+        }
+    }
+    qsort(devices, count, sizeof(OpenCLDeviceBenchmark), compare_ocl_device_desc);
+    fprintf(stderr, "platform_idx\tdevice_idx\tdevice_name\truntime\n");
+    for (i = 0; i < count; i++)
+        fprintf(stdout, "%d\t%d\t%s\t%"PRId64"\n",
+                devices[i].platform_idx, devices[i].device_idx,
+                devices[i].device_name, devices[i].runtime);
+
+    av_opencl_free_device_list(&device_list);
+    av_free(devices);
+    return 0;
+}
+
+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;
+}
--- a/common.mak
+++ b/common.mak
@@ -10,7 +10,7 @@ ifndef SUBDIR
 ifndef V
 Q      = @
 ECHO   = printf "$(1)\t%s\n" $(2)
-BRIEF  = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP
+BRIEF  = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES
 SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM

 MSG    = $@
@@ -43,6 +43,7 @@ endef
 COMPILE_C = $(call COMPILE,CC)
 COMPILE_CXX = $(call COMPILE,CXX)
 COMPILE_S = $(call COMPILE,AS)
+COMPILE_HOSTC = $(call COMPILE,HOSTCC)

 %.o: %.c
 	$(COMPILE_C)
@@ -56,6 +57,12 @@ COMPILE_S = $(call COMPILE,AS)
 %.o: %.S
 	$(COMPILE_S)

+%_host.o: %.c
+	$(COMPILE_HOSTC)
+
+%.o: %.rc
+	$(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $<
+
 %.i: %.c
 	$(CC) $(CCFLAGS) $(CC_E) $<

@@ -82,14 +89,15 @@ endif
 include $(SRC_PATH)/arch.mak

 OBJS      += $(OBJS-yes)
+SLIBOBJS  += $(SLIBOBJS-yes)
 FFLIBS    := $(FFLIBS-yes) $(FFLIBS)
 TESTPROGS += $(TESTPROGS-yes)

 LDLIBS       = $(FFLIBS:%=%$(BUILDSUF))
 FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)

-EXAMPLES  := $(EXAMPLES:%=$(SUBDIR)%-example$(EXESUF))
 OBJS      := $(sort $(OBJS:%=$(SUBDIR)%))
+SLIBOBJS  := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
 TESTOBJS  := $(TESTOBJS:%=$(SUBDIR)%) $(TESTPROGS:%=$(SUBDIR)%-test.o)
 TESTPROGS := $(TESTPROGS:%=$(SUBDIR)%-test$(EXESUF))
 HOSTOBJS  := $(HOSTPROGS:%=$(SUBDIR)%.o)
@@ -99,7 +107,8 @@ TOOLOBJS  := $(TOOLS:%=tools/%.o)
 TOOLS     := $(TOOLS:%=tools/%$(EXESUF))
 HEADERS   += $(HEADERS-yes)

-DEP_LIBS := $(foreach NAME,$(FFLIBS),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
+PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
+DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))

 SRC_DIR    := $(SRC_PATH)/lib$(NAME)
 ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
@@ -112,18 +121,19 @@ checkheaders: $(HOBJS)
 alltools: $(TOOLS)

 $(HOSTOBJS): %.o: %.c
-	$(call COMPILE,HOSTCC)
+	$(COMPILE_HOSTC)

 $(HOSTPROGS): %$(HOSTEXESUF): %.o
-	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $< $(HOSTLIBS)
+	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTLIBS)

 $(OBJS):     | $(sort $(dir $(OBJS)))
 $(HOBJS):    | $(sort $(dir $(HOBJS)))
 $(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
+$(SLIBOBJS): | $(sort $(dir $(SLIBOBJS)))
 $(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
 $(TOOLOBJS): | tools

-OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(TESTOBJS))
+OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))

 CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
 DISTCLEANSUFFIXES = *.pc
@@ -138,4 +148,4 @@ endef

 $(eval $(RULES))

-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d))
+-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d))
--- a/compat/aix/math.h
+++ b/compat/aix/math.h
@@ -1,9 +1,26 @@
 /*
- * Workaround aix-specific class() function clashing with ffmpeg class usage
+ * Work around the class() function in AIX math.h clashing with
+ * identifiers named "class".
+ *
+ * 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 COMPAT_AIX_MATH_H
-#define COMPAT_AIX_MATH_H
+#ifndef FFMPEG_COMPAT_AIX_MATH_H
+#define FFMPEG_COMPAT_AIX_MATH_H

 #define class class_in_math_h_causes_problems

@@ -11,4 +28,4 @@

 #undef class

-#endif /* COMPAT_AIX_MATH_H */
+#endif /* FFMPEG_COMPAT_AIX_MATH_H */
--- a/compat/avisynth/avisynth_c.h
+++ b/compat/avisynth/avisynth_c.h
@@ -13,7 +13,8 @@
 //
 // 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
+// Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
+// MA 02110-1301 USA, or visit
 // http://www.gnu.org/copyleft/gpl.html .
 //
 // As a special exception, I give you permission to link to the
--- 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
@@ -13,7 +13,8 @@
 //
 // 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
+// Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
+// MA 02110-1301 USA, or visit
 // http://www.gnu.org/copyleft/gpl.html .
 //
 // As a special exception, I give you permission to link to the
--- a/compat/getopt.c
+++ b/compat/getopt.c
@@ -38,8 +38,6 @@ static int optind = 1;
 static int optopt;
 static char *optarg;

-#undef fprintf
-
 static int getopt(int argc, char *argv[], char *opts)
 {
    static int sp = 1;
--- a/compat/os2threads.h
+++ b/compat/os2threads.h
@@ -32,6 +32,8 @@
 #undef __STRICT_ANSI__          /* for _beginthread() */
 #include <stdlib.h>

+#include "libavutil/mem.h"
+
 typedef TID  pthread_t;
 typedef void pthread_attr_t;

--- a/compat/tms470/math.h
+++ b/compat/tms470/math.h
@@ -1,3 +1,24 @@
+/*
+ * 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 FFMPEG_COMPAT_TMS470_MATH_H
+#define FFMPEG_COMPAT_TMS470_MATH_H
+
 #include_next <math.h>

 #undef INFINITY
@@ -5,3 +26,5 @@

 #define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 }))
 #define NAN      (*(const float*)((const unsigned []){ 0x7fc00000 }))
+
+#endif /* FFMPEG_COMPAT_TMS470_MATH_H */
--- a/compat/va_copy.h
+++ b/compat/va_copy.h
@@ -24,3 +24,6 @@
 #if !defined(va_copy) && defined(_MSC_VER)
 #define va_copy(dst, src) ((dst) = (src))
 #endif
+#if !defined(va_copy) && defined(__GNUC__) && __GNUC__ < 3
+#define va_copy(dst, src) __va_copy(dst, src)
+#endif
--- a/compat/w32pthreads.h
+++ b/compat/w32pthreads.h
@@ -26,8 +26,8 @@
 * w32threads to pthreads wrapper
 */

-#ifndef LIBAV_W32PTHREADS_H
-#define LIBAV_W32PTHREADS_H
+#ifndef FFMPEG_COMPAT_W32PTHREADS_H
+#define FFMPEG_COMPAT_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
@@ -62,11 +62,18 @@ typedef struct pthread_cond_t {
 } pthread_cond_t;

 /* function pointers to conditional variable API on windows 6.0+ kernels */
+#if _WIN32_WINNT < 0x0600
 static void (WINAPI *cond_broadcast)(pthread_cond_t *cond);
 static void (WINAPI *cond_init)(pthread_cond_t *cond);
 static void (WINAPI *cond_signal)(pthread_cond_t *cond);
 static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
                                DWORD milliseconds);
+#else
+#define cond_init      InitializeConditionVariable
+#define cond_broadcast WakeAllConditionVariable
+#define cond_signal    WakeConditionVariable
+#define cond_wait      SleepConditionVariableCS
+#endif

 static unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
 {
@@ -268,13 +275,8 @@ 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 /* LIBAV_W32PTHREADS_H */
+#endif /* FFMPEG_COMPAT_W32PTHREADS_H */
--- a/compat/windows/makedef
+++ b/compat/windows/makedef
@@ -0,0 +1,132 @@
+#!/bin/sh
+
+# Copyright (c) 2013, Derek Buitenhuis
+#
+# Permission to use, copy, modify, and/or distribute this software for any
+# purpose with or without fee is hereby granted, provided that the above
+# copyright notice and this permission notice appear in all copies.
+#
+# THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
+# WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
+# MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
+# ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
+# WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
+# ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
+# OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
+
+# mktemp isn't POSIX, so supply an implementation
+mktemp() {
+    echo "${2%%XXX*}.${HOSTNAME}.${UID}.$$"
+}
+
+if [ $# -lt 2 ]; then
+    echo "Usage: makedef <version_script> <objects>" >&2
+    exit 0
+fi
+
+vscript=$1
+shift
+
+if [ ! -f "$vscript" ]; then
+    echo "Version script does not exist" >&2
+    exit 1
+fi
+
+for object in "$@"; do
+    if [ ! -f "$object" ]; then
+        echo "Object does not exist: ${object}" >&2
+        exit 1
+    fi
+done
+
+# Create a lib temporarily to dump symbols from.
+# It's just much easier to do it this way
+libname=$(mktemp -u "library").lib
+
+trap 'rm -f -- $libname' EXIT
+
+lib -out:${libname} $@ >/dev/null
+if [ $? != 0 ]; then
+    echo "Could not create temporary library." >&2
+    exit 1
+fi
+
+IFS='
+'
+
+# Determine if we're building for x86 or x86_64 and
+# set the symbol prefix accordingly.
+prefix=""
+arch=$(dumpbin -headers ${libname} |
+       tr '\t' ' ' |
+       grep '^ \+.\+machine \+(.\+)' |
+       head -1 |
+       sed -e 's/^ \{1,\}.\{1,\} \{1,\}machine \{1,\}(\(...\)).*/\1/')
+
+if [ "${arch}" = "x86" ]; then
+    prefix="_"
+else
+    if [ "${arch}" != "ARM" ] && [ "${arch}" != "x64" ]; then
+        echo "Unknown machine type." >&2
+        exit 1
+    fi
+fi
+
+started=0
+regex="none"
+
+for line in $(cat ${vscript} | tr '\t' ' '); do
+    # We only care about global symbols
+    echo "${line}" | grep -q '^ \+global:'
+    if [ $? = 0 ]; then
+        started=1
+        line=$(echo "${line}" | sed -e 's/^ \{1,\}global: *//')
+    else
+        echo "${line}" | grep -q '^ \+local:'
+        if [ $? = 0 ]; then
+            started=0
+        fi
+    fi
+
+    if [ ${started} = 0 ]; then
+        continue
+    fi
+
+    # Handle multiple symbols on one line
+    IFS=';'
+
+    # Work around stupid expansion to filenames
+    line=$(echo "${line}" | sed -e 's/\*/.\\+/g')
+    for exp in ${line}; do
+        # Remove leading and trailing whitespace
+        exp=$(echo "${exp}" | sed -e 's/^ *//' -e 's/ *$//')
+
+        if [ "${regex}" = "none" ]; then
+            regex="${exp}"
+        else
+            regex="${regex};${exp}"
+        fi
+    done
+
+    IFS='
+'
+done
+
+dump=$(dumpbin -linkermember:1 ${libname})
+
+rm ${libname}
+
+IFS=';'
+list=""
+for exp in ${regex}; do
+    list="${list}"'
+'$(echo "${dump}" |
+          sed -e '/public symbols/,$!d' -e '/^ \{1,\}Summary/,$d' -e "s/ \{1,\}${prefix}/ /" -e 's/ \{1,\}/ /g' |
+          tail -n +2 |
+          cut -d' ' -f3 |
+          grep "^${exp}" |
+          sed -e 's/^/    /')
+done
+
+echo "EXPORTS"
+echo "${list}" | sort | uniq | tail -n +2
--- a/823
+++ b/823
--- a/doc/APIchanges
+++ b/doc/APIchanges
@@ -4,7 +4,7 @@ since the last major version increase or the API was added.
 The last version increases were:
 libavcodec:    2013-03-xx
 libavdevice:   2013-03-xx
-libavfilter:   2012-06-22
+libavfilter:   2013-12-xx
 libavformat:   2013-03-xx
 libavresample: 2012-10-05
 libpostproc:   2011-04-18
@@ -15,24 +15,226 @@ libavutil:     2012-10-22

 API changes, most recent first:

-2013-07-03 - xxxxxxx - lavfi 3.78.100 - avfilter.h
+2014-03-18 - e9c8a9a - lsws 2.5.102
+  Make gray16 full-scale.
+
+2014-xx-xx - xxxxxxx - lavu 53.05.0 - frame.h
+  Add av_frame_copy() for copying the frame data.
+
+2014-02-xx - xxxxxxx - lswr 0.18.100 - swresample.h
+  Add swr_is_initialized() for checking whether a resample context is initialized.
+
+2014-02-xx - xxxxxxx - lavr 1.2.0 - avresample.h
+  Add avresample_is_open() for checking whether a resample context is open.
+
+2014-xx-xx - xxxxxxx - lavu 53.04.0  - opt.h
+  Add AV_OPT_FLAG_EXPORT and AV_OPT_FLAG_READONLY to mark options meant (only)
+  for reading.
+
+2014-xx-xx - xxxxxxx - lavu 53.03.01 - opt.h
+  Deprecate unused AV_OPT_FLAG_METADATA.
+
+2014-02-xx - xxxxxxx - lavd 55.10.100 - avdevice.h
+  Add avdevice_list_devices() and avdevice_free_list_devices()
+
+2014-02-16 - db3c970 - lavf 55.33.100 - avio.h
+  Add avio_find_protocol_name() to find out the name of the protocol that would
+  be selected for a given URL.
+
+2014-02-xx - xxxxxxx - lavu 53.3.0 - frame.h
+  Add AV_FRAME_DATA_DOWNMIX_INFO value to the AVFrameSideDataType enum and
+  downmix_info.h API, which identify downmix-related metadata.
+
+2014-02-11 - 1b05ac2 - lavf 55.32.100 - avformat.h
+  Add av_write_uncoded_frame() and av_interleaved_write_uncoded_frame().
+
+2014-02-04 - 3adb5f8 / d9ae103 - lavf 55.30.100 / 55.11.0 - avformat.h
+  Add AVFormatContext.max_interleave_delta for controlling amount of buffering
+  when interleaving.
+
+2014-02-02 - xxxxxxx - lavf 55.29.100 - avformat.h
+  Add output_ts_offset muxing option to AVFormatContext.
+
+2014-01-27 - 102bd64 - lavd 55.7.100 - avdevice.h
+                       lavf 55.28.100 - avformat.h
+  Add avdevice_dev_to_app_control_message() function.
+
+2014-01-27 - 7151411 - lavd 55.6.100 - avdevice.h
+                       lavf 55.27.100 - avformat.h
+  Add avdevice_app_to_dev_control_message() function.
+
+2014-01-24 - 86bee79 - lavf 55.26.100 - avformat.h
+  Add AVFormatContext option metadata_header_padding to allow control over the
+  amount of padding added.
+
+2014-01-20 - eef74b2 / 93c553c - lavc 55.48.102 / 55.32.1 - avcodec.h
+  Edges are not required anymore on video buffers allocated by get_buffer2()
+  (i.e. as if the CODEC_FLAG_EMU_EDGE flag was always on). Deprecate
+  CODEC_FLAG_EMU_EDGE and avcodec_get_edge_width().
+
+2014-01-19 - xxxxxxx - lavf 55.25.100 - avformat.h
+  Add avformat_get_mov_video_tags() and avformat_get_mov_audio_tags().
+
+2014-01-19 - xxxxxxx - lavu 52.63.100 - rational.h
+  Add av_make_q() function.
+
+2014-01-05 - 4cf4da9 / 5b4797a - lavu 52.62.100 / 53.2.0 - frame.h
+  Add AV_FRAME_DATA_MATRIXENCODING value to the AVFrameSideDataType enum, which
+  identifies AVMatrixEncoding data.
+
+2014-01-05 - 751385f / 5c437fb - lavu 52.61.100 / 53.1.0 - channel_layout.h
+  Add values for various Dolby flags to the AVMatrixEncoding enum.
+
+2014-01-04 - xxxxxxx - lavu 52.60.100 - mathematics.h
+  Add av_add_stable() function.
+
+2013-12-22 - xxxxxxx - lavu 52.59.100 - avstring.h
+  Add av_strnlen() function.
+
+2013-12-xx - xxxxxxx - lavu 52.57.100 - opencl.h
+  Add av_opencl_benchmark() function.
+
+2013-11-xx - xxxxxxx - lavu 52.56.100 - ffversion.h
+  Moves version.h to libavutil/ffversion.h.
+  Install ffversion.h and make it public.
+
+2013-12-11 - 29c83d2 / b9fb59d,409a143 / 9431356,44967ab / d7b3ee9 - lavc 55.45.101 / 55.28.1 - avcodec.h
+  av_frame_alloc(), av_frame_unref() and av_frame_free() now can and should be
+  used instead of avcodec_alloc_frame(), avcodec_get_frame_defaults() and
+  avcodec_free_frame() respectively. The latter three functions are deprecated.
+
+2013-12-09 - 7a60348 / 7e244c6- - lavu 52.58.100 / 52.20.0 - frame.h
+  Add AV_FRAME_DATA_STEREO3D value to the AVFrameSideDataType enum and
+  stereo3d.h API, that identify codec-independent stereo3d information.
+
+2013-11-26 - 625b290 / 1eaac1d- - lavu 52.55.100 / 52.19.0 - frame.h
+  Add AV_FRAME_DATA_A53_CC value to the AVFrameSideDataType enum, which
+  identifies ATSC A53 Part 4 Closed Captions data.
+
+2013-11-XX - xxxxxxx - lavu 52.54.100 - avstring.h
+  Add av_utf8_decode() function.
+
+2013-11-22 - fb7d70c - lavc 55.44.100 - avcodec.h
+  Add HEVC profiles
+
+2013-11-xx - xxxxxxx - lavc 55.44.100 - avcodec.h
+  Add av_packet_{un,}pack_dictionary()
+  Add AV_PKT_METADATA_UPDATE side data type, used to transmit key/value
+  strings between a stream and the application.
+
+2013-11-14 - 7c888ae / cce3e0a - lavu 52.53.100 / 52.18.0 - mem.h
+  Move av_fast_malloc() and av_fast_realloc() for libavcodec to libavutil.
+
+2013-11-14 - b71e4d8 / 8941971 - lavc 55.43.100 / 55.27.0 - avcodec.h
+  Deprecate AVCodecContext.error_rate, it is replaced by the 'error_rate'
+  private option of the mpegvideo encoder family.
+
+2013-11-14 - 31c09b7 / 728c465 - lavc 55.26.0 - vdpau.h
+  Add av_vdpau_get_profile().
+  Add av_vdpau_alloc_context(). This function must from now on be
+  used for allocating AVVDPAUContext.
+
+2013-11-04 - be41f21 / cd8f772 - lavc 55.41.100 / 55.25.0 - avcodec.h
+                       lavu 52.51.100 - frame.h
+  Add ITU-R BT.2020 and other not yet included values to color primaries,
+  transfer characteristics and colorspaces.
+
+2013-11-04 - xxxxxxx - lavu 52.50.100 - avutil.h
+  Add av_fopen_utf8()
+
+2013-10-31 - 78265fc / 28096e0 - lavu 52.49.100 / 52.17.0 - frame.h
+  Add AVFrame.flags and AV_FRAME_FLAG_CORRUPT.
+
+2013-10-27 - xxxxxxx - lavc 55.39.100 - avcodec.h
+  Add CODEC_CAP_DELAY support to avcodec_decode_subtitle2.
+
+2013-10-27 - xxxxxxx - lavu 52.48.100 - parseutils.h
+  Add av_get_known_color_name().
+
+2013-10-17 - xxxxxxx - lavu 52.47.100 - opt.h
+  Add AV_OPT_TYPE_CHANNEL_LAYOUT and channel layout option handlers
+  av_opt_get_channel_layout() and av_opt_set_channel_layout().
+
+2013-10-xx - xxxxxxx -libswscale 2.5.101 - options.c
+  Change default scaler to bicubic
+
+2013-10-03 - xxxxxxx - lavc 55.34.100 - avcodec.h
+  Add av_codec_get_max_lowres()
+
+2013-10-02 - xxxxxxx - lavf 55.19.100 - avformat.h
+  Add audio/video/subtitle AVCodec fields to AVFormatContext to force specific
+  decoders
+
+2013-09-28 - 7381d31 / 0767bfd - lavfi 3.88.100 / 3.11.0 - avfilter.h
+  Add AVFilterGraph.execute and AVFilterGraph.opaque for custom slice threading
+  implementations.
+
+2013-09-21 - 85f8a3c / e208e6d - lavu 52.46.100 / 52.16.0 - pixfmt.h
+  Add interleaved 4:2:2 8/10-bit formats AV_PIX_FMT_NV16 and
+  AV_PIX_FMT_NV20.
+
+2013-09-16 - c74c3fb / 3feb3d6 - lavu 52.44.100 / 52.15.0 - mem.h
+  Add av_reallocp.
+
+2013-09-04 - 3e1f507 - lavc 55.31.101 - avcodec.h
+  avcodec_close() argument can be NULL.
+
+2013-09-04 - 36cd017 - lavf 55.16.101 - avformat.h
+  avformat_close_input() argument can be NULL and point on NULL.
+
+2013-08-29 - e31db62 - lavf 55.15.100 - avformat.h
+  Add av_format_get_probe_score().
+
+2013-08-15 - 1e0e193 - lsws 2.5.100 -
+  Add a sws_dither AVOption, allowing to set the dither algorithm used
+
+2013-08-xx - xxxxxxx - lavc 55.27.100 - vdpau.h
+  Add a render2 alternative to the render callback function.
+
+2013-08-xx - xxxxxxx - lavc 55.26.100 - vdpau.h
+  Add allocation function for AVVDPAUContext, allowing
+  to extend it in the future without breaking ABI/API.
+
+2013-08-10 - 67a580f / 5a9a9d4 - lavc 55.25.100 / 55.16.0 - avcodec.h
+  Extend AVPacket API with av_packet_unref, av_packet_ref,
+  av_packet_move_ref, av_packet_copy_props, av_packet_free_side_data.
+
+2013-08-05 - 9547e3e / f824535 - lavc 55.22.100 / 55.13.0 - avcodec.h
+  Deprecate the bitstream-related members from struct AVVDPAUContext.
+  The bistream buffers no longer need to be explicitly freed.
+
+2013-08-05 - 3b805dc / 549294f - lavc 55.21.100 / 55.12.0 - avcodec.h
+  Deprecate the CODEC_CAP_HWACCEL_VDPAU codec capability. Use CODEC_CAP_HWACCEL
+  and select the AV_PIX_FMT_VDPAU format with get_format() instead.
+
+2013-08-05 - 4ee0984 / a0ad5d0 - lavu 52.41.100 / 52.14.0 - pixfmt.h
+  Deprecate AV_PIX_FMT_VDPAU_*. Use AV_PIX_FMT_VDPAU instead.
+
+2013-08-02 - 82fdfe8 / a8b1927 - lavc 55.20.100 / 55.11.0 - avcodec.h
+  Add output_picture_number to AVCodecParserContext.
+
+2013-07-23 - abc8110 - lavc 55.19.100 - avcodec.h
+  Add avcodec_chroma_pos_to_enum()
+  Add avcodec_enum_to_chroma_pos()
+
+2013-07-03 - 838bd73 - 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
+2013-06-24 - af5f9c0 / 95d5246 - lavc 55.17.100 / 55.10.0 - avcodec.h
  Add MPEG-2 AAC profiles

-2013-06-xx - xxxxxxx - lavf 55.10.100 - avformat.h
+2013-06-25 - af5f9c0 / 95d5246 - lavf 55.10.100 - avformat.h
  Add AV_DISPOSITION_* flags to indicate text track kind.

-2013-06-xx - xxxxxxx - lavu 52.36.100
+2013-06-15 - 99b8cd0 - 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
+2013-06-04 - 30b491f / fc962d4 - lavu 52.35.100 / 52.13.0 - mem.h
  Add av_realloc_array and av_reallocp_array

 2013-05-30 - 682b227 - lavu 52.35.100
@@ -42,55 +244,55 @@ API changes, most recent first:
   av_sha512_update()
   av_sha512_final()

-2013-05-24 - xxxxxxx - lavfi 3.70.100 - avfilter.h
+2013-05-24 - 8d4e969 / 129bb23 - lavfi 3.10.0 / 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
+2013-05-24 - fe40a9f / 2a6eaea - lavu 52.12.0 / 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
+2013-05-24 - 0c25c39 / b493847 - lavc 55.7.0 / 55.12.100 - avcodec.h
  Add picture_structure to AVCodecParserContext.

-2013-05-17 - xxxxxxx - lavu 52.33.100 - opt.h
+2013-05-17 - 3a751ea - 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
+2013-05-13 - e398416 - lavu 52.31.100 - mem.h
  Add av_dynarray2_add().

-2013-05-12 - xxxxxxx - lavfi 3.65.100
+2013-05-12 - 1776177 - lavfi 3.65.100
  Add AVFILTER_FLAG_SUPPORT_TIMELINE* filter flags.

-2013-04-19 - xxxxxxx - lavc 55.4.100
+2013-04-19 - 380cfce - lavc 55.4.100
  Add AV_CODEC_PROP_TEXT_SUB property for text based subtitles codec.

-2013-04-18 - xxxxxxx - lavf 55.3.100
+2013-04-18 - 7c1a002 - 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
+2013-04-10 - af0d270 - 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
+2013-03-30 - 5c73645 - lavu 52.24.100 - samplefmt.h
  Add av_samples_alloc_array_and_samples().

-2013-03-29 - xxxxxxx - lavf 55.1.100 - avformat.h
+2013-03-29 - ef7b6b4 - lavf 55.1.100 - avformat.h
  Add av_guess_frame_rate()

-2013-03-20 - xxxxxxx - lavu 52.22.100 - opt.h
+2013-03-20 - 8d928a9 - 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
+2013-03-17 - 7aa9af5 - 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
+2013-03-07 - 9767ec6 - lavu 52.18.100 - avstring.h,bprint.h
  Add av_escape() and av_bprint_escape() API.

-2013-02-24 - xxxxxx - lavfi 3.41.100 - buffersink.h
+2013-02-24 - b59cd08 - lavfi 3.41.100 - buffersink.h
  Add sample_rates field to AVABufferSinkParams.

 2013-01-17 - a1a707f - lavf 54.61.100
@@ -103,7 +305,7 @@ API changes, most recent first:
  Add AVFilterLink.channels, avfilter_link_get_channels()
  and avfilter_ref_get_channels().

-2012-12-15 - 2ada584d - lavc 54.80.100 - avcodec.h
+2012-12-15 - 96d815fc - lavc 54.80.100 - avcodec.h
  Add pkt_size field to AVFrame.

 2012-11-25 - c70ec631 - lavu 52.9.100 - opt.h
@@ -216,52 +418,52 @@ API changes, most recent first:
 2012-03-26 - a67d9cf - lavfi 2.66.100
  Add avfilter_fill_frame_from_{audio_,}buffer_ref() functions.

-2013-05-xx - xxxxxxx - lavu 52.11.0 - pixdesc.h
+2013-05-15 - ff46809 / e6c4ac7 - lavu 52.32.100 / 52.11.0 - pixdesc.h
  Replace PIX_FMT_* flags with AV_PIX_FMT_FLAG_*.

-2013-04-xx - xxxxxxx - lavc 55.4.0 - avcodec.h
+2013-04-03 - 6fc58a8 / 507b1e4 - lavc 55.7.100 / 55.4.0 - avcodec.h
  Add field_order to AVCodecParserContext.

-2013-03-xx - xxxxxxx - lavc 55.2.0 - avcodec.h
+2013-04-19 - f4b05cd / 5e83d9a - lavc 55.5.100 / 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
+2013-04-11 - lavfi 3.53.100 / 3.8.0
+  231fd44 / 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
+  86070b8 / 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
+  4fde705 / 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().
+  710b0aa / 48a5ada - Add avfilter_init_str(), deprecate avfilter_init_filter().
+  46de9ba / 1ba95a9 - Add avfilter_init_dict().
+  16fc24b / 7cdd737 - Add AVFilter.flags field and AVFILTER_FLAG_DYNAMIC_{INPUTS,OUTPUTS} flags.
+  f4db6bf / 7e8fe4b - Add avfilter_pad_count() for counting filter inputs/outputs.
+  835cc0f / 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
+2013-04-09 - lavfi 3.51.100 / 3.7.0 - avfilter.h
+  0594ef0 / 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().
+  44d4488 / 8114c10 - Add avfilter_get_class().
  Switch all filters to use AVOptions.

-2013-03-19 - 2c328a9 - lavu 52.9.0 - pixdesc.h
+2013-03-19 - 17ebef2 / 2c328a9 - lavu 52.20.100 / 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
+2013-03-16 - ecade98 / 42c7c61 - lavfi 3.47.100 / 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
+2013-03-08 - Reference counted buffers - lavu 52.8.0, lavc 55.0.100 / 55.0.0, lavf 55.0.100 / 55.0.0,
+lavd 54.4.100 / 54.0.0, lavfi 3.5.0
+  36099df / 8e401db, 532f31a / 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.
+  2653e12 / 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
+  c4e8821 / 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
+  80e9e63 / 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()
@@ -279,30 +481,30 @@ lavd 54.0.0, lavfi 3.5.0
                * 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
+  a05a44e / 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
+2013-03-17 - 6c17ff8 / 12c5c1d - lavu 52.19.100 / 52.8.0 - avstring.h
  Add av_isdigit, av_isgraph, av_isspace, av_isxdigit.

-2013-02-23 - 9f12235 - lavfi 3.4.0 - avfiltergraph.h
+2013-02-23 - 71cf094 / 9f12235 - lavfi 3.40.100 / 3.4.0 - avfiltergraph.h
  Add resample_lavr_opts to AVFilterGraph for setting libavresample options
  for auto-inserted resample filters.

-2013-01-25 - 38c1466 - lavu 52.7.0 - dict.h
+2013-01-25 - e7e14bc / 38c1466 - lavu 52.17.100 / 52.7.0 - dict.h
  Add av_dict_parse_string() to set multiple key/value pairs at once from a
  string.

-2013-01-25 - b85a5e8 - lavu 52.6.0 - avstring.h
+2013-01-25 - 25be630 / b85a5e8 - lavu 52.16.100 / 52.6.0 - avstring.h
  Add av_strnstr()

-2013-01-15 - 8ee288d - lavu 52.5.0 - hmac.h
+2013-01-15 - e7e0186 / 8ee288d - lavu 52.15.100 / 52.5.0 - hmac.h
  Add AVHMAC.

-2013-01-13 - 44e065d - lavc 54.87.100 / 54.36.0 - vdpau.h
+2013-01-13 - 8ee7b38 / 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
@@ -390,7 +592,7 @@ lavd 54.0.0, lavfi 3.5.0
 2012-07-29 - 7c26761 / 681ed00 - lavf 54.22.100 / 54.13.0 - avformat.h
  Add AVFMT_FLAG_NOBUFFER for low latency use cases.

-2012-07-10 - 5fade8a - lavu 51.37.0
+2012-07-10 - fbe0245 / f3e5e6f - lavu 51.65.100 / 51.37.0
  Add av_malloc_array() and av_mallocz_array()

 2012-06-22 - e847f41 / d3d3a32 - lavu 51.61.100 / 51.34.0
--- a/doc/Doxyfile
+++ b/doc/Doxyfile
@@ -31,9 +31,9 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.

-PROJECT_NUMBER         = 2.0
+PROJECT_NUMBER         = 2.2.11

-# With the PROJECT_LOGO tag one can specify an logo or icon that is included
+# With the PROJECT_LOGO tag one can specify a logo or icon that is included
 # in the documentation. The maximum height of the logo should not exceed 55
 # pixels and the maximum width should not exceed 200 pixels. Doxygen will
 # copy the logo to the output directory.
@@ -793,13 +793,13 @@ HTML_FILE_EXTENSION    = .html
 # each generated HTML page. If it is left blank doxygen will generate a
 # standard header.

-#HTML_HEADER            = doc/doxy/header.html
+HTML_HEADER            =

 # The HTML_FOOTER tag can be used to specify a personal HTML footer for
 # each generated HTML page. If it is left blank doxygen will generate a
 # standard footer.

-#HTML_FOOTER            = doc/doxy/footer.html
+HTML_FOOTER            =

 # The HTML_STYLESHEET tag can be used to specify a user-defined cascading
 # style sheet that is used by each HTML page. It can be used to
@@ -808,7 +808,7 @@ HTML_FILE_EXTENSION    = .html
 # the style sheet file to the HTML output directory, so don't put your own
 # stylesheet in the HTML output directory as well, or it will be erased!

-#HTML_STYLESHEET        = doc/doxy/doxy_stylesheet.css
+HTML_STYLESHEET        =

 # The HTML_COLORSTYLE_HUE tag controls the color of the HTML output.
 # Doxygen will adjust the colors in the stylesheet and background images
--- a/doc/Makefile
+++ b/doc/Makefile
@@ -14,11 +14,11 @@ COMPONENTS-$(CONFIG_AVFORMAT)   += ffmpeg-formats ffmpeg-protocols
 COMPONENTS-$(CONFIG_AVDEVICE)   += ffmpeg-devices
 COMPONENTS-$(CONFIG_AVFILTER)   += ffmpeg-filters

-MANPAGES1   = $(PROGS-yes:%=doc/%.1)    $(PROGS-yes:%=doc/%-all.1)    $(COMPONENTS-yes:%=doc/%.1)
+MANPAGES1   = $(AVPROGS-yes:%=doc/%.1)    $(AVPROGS-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) \
+PODPAGES    = $(AVPROGS-yes:%=doc/%.pod)  $(AVPROGS-yes:%=doc/%-all.pod)  $(COMPONENTS-yes:%=doc/%.pod)  $(LIBRARIES-yes:%=doc/%.pod)
+HTMLPAGES   = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
              doc/developer.html                                        \
              doc/faq.html                                              \
              doc/fate.html                                             \
@@ -36,6 +36,25 @@ DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
 DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
 DOCS = $(DOCS-yes)

+DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
+DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE)           += avcodec
+DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
+DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE)      += filter_audio
+DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE)   += filtering_audio
+DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE)   += filtering_video
+DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE)          += metadata
+DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE)            += muxing
+DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE)          += remuxing
+DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)  += resampling_audio
+DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)     += scaling_video
+DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE)     += transcode_aac
+ALL_DOC_EXAMPLES_LIST = $(DOC_EXAMPLES-) $(DOC_EXAMPLES-yes)
+
+DOC_EXAMPLES       := $(DOC_EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
+ALL_DOC_EXAMPLES   := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)$(EXESUF))
+ALL_DOC_EXAMPLES_G := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
+PROGS              += $(DOC_EXAMPLES)
+
 all-$(CONFIG_DOC): doc

 doc: documentation
@@ -43,7 +62,9 @@ doc: documentation
 apidoc: doc/doxy/html
 documentation: $(DOCS)

-TEXIDEP = awk '/^@(verbatim)?include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)
+examples: $(DOC_EXAMPLES)
+
+TEXIDEP = perl $(SRC_PATH)/doc/texidep.pl $(SRC_PATH) $< $@ >$(@:%=%.d)

 doc/%.txt: TAG = TXT
 doc/%.txt: doc/%.texi
@@ -84,12 +105,28 @@ doc/%.3: doc/%.pod $(GENTEXI)
 	$(M)pod2man --section=3 --center=" " --release=" " $< > $@

 $(DOCS) doc/doxy/html: | doc/
+$(DOC_EXAMPLES:%$(EXESUF)=%.o): | doc/examples
+OBJDIRS += doc/examples

-doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(INSTHEADERS)
-	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $^
+DOXY_INPUT      = $(addprefix $(SRC_PATH)/, $(INSTHEADERS) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c))
+
+doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(DOXY_INPUT)
+	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $< $(DOXY_INPUT)
+
+install-doc: install-html install-man
+
+install-html:

 install-man:

+ifdef CONFIG_HTMLPAGES
+install-progs-$(CONFIG_DOC): install-html
+
+install-html: $(HTMLPAGES)
+	$(Q)mkdir -p "$(DOCDIR)"
+	$(INSTALL) -m 644 $(HTMLPAGES) "$(DOCDIR)"
+endif
+
 ifdef CONFIG_MANPAGES
 install-progs-$(CONFIG_DOC): install-man

@@ -100,19 +137,29 @@ install-man: $(MANPAGES)
 	$(INSTALL) -m 644 $(MANPAGES3) "$(MANDIR)/man3"
 endif

-uninstall: uninstall-man
+uninstall: uninstall-doc
+
+uninstall-doc: uninstall-html uninstall-man
+
+uninstall-html:
+	$(RM) -r "$(DOCDIR)"

 uninstall-man:
-	$(RM) $(addprefix "$(MANDIR)/man1/",$(MANPAGES1))
-	$(RM) $(addprefix "$(MANDIR)/man3/",$(MANPAGES3))
+	$(RM) $(addprefix "$(MANDIR)/man1/",$(AVPROGS-yes:%=%.1) $(AVPROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
+	$(RM) $(addprefix "$(MANDIR)/man3/",$(LIBRARIES-yes:%=%.3))

 clean:: docclean

 distclean:: docclean
 	$(RM) doc/config.texi

-docclean:
-	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 $(CLEANSUFFIXES:%=doc/%) doc/avoptions_*.texi
+examplesclean:
+	$(RM) $(ALL_DOC_EXAMPLES) $(ALL_DOC_EXAMPLES_G)
+	$(RM) $(CLEANSUFFIXES:%=doc/examples/%)
+
+docclean: examplesclean
+	$(RM) $(CLEANSUFFIXES:%=doc/%)
+	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 doc/avoptions_*.texi
 	$(RM) -r doc/doxy/html

 -include $(wildcard $(DOCS:%=%.d))
--- a/doc/RELEASE_NOTES
+++ b/doc/RELEASE_NOTES
@@ -1,7 +1,7 @@
 Release Notes
 =============

-* 2.0 "Nameless"  July, 2013
+* 2.2 "Muybridge"  March, 2014


 General notes
@@ -14,7 +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.
-
-AVI/AVXSynth
--------
-If you want to use FFmpeg with AVISynth, you need AVISynth 2.6.0 at minimum.
--- a/doc/avutil.txt
+++ b/doc/avutil.txt
@@ -1,36 +0,0 @@
-AVUtil
-======
-libavutil is a small lightweight library of generally useful functions.
-It is not a library for code needed by both libavcodec and libavformat.
-
-
-Overview:
-=========
-adler32.c               adler32 checksum
-aes.c                   AES encryption and decryption
-fifo.c                  resizeable first in first out buffer
-intfloat_readwrite.c    portable reading and writing of floating point values
-log.c                   "printf" with context and level
-md5.c                   MD5 Message-Digest Algorithm
-rational.c              code to perform exact calculations with rational numbers
-tree.c                  generic AVL tree
-crc.c                   generic CRC checksumming code
-integer.c               128bit integer math
-lls.c
-mathematics.c           greatest common divisor, integer sqrt, integer log2, ...
-mem.c                   memory allocation routines with guaranteed alignment
-
-Headers:
-bswap.h                 big/little/native-endian conversion code
-x86_cpu.h               a few useful macros for unifying x86-64 and x86-32 code
-avutil.h
-common.h
-intreadwrite.h          reading and writing of unaligned big/little/native-endian integers
-
-
-Goals:
-======
-* Modular (few interdependencies and the possibility of disabling individual parts during ./configure)
-* Small (source and object)
-* Efficient (low CPU and memory usage)
-* Useful (avoid useless features almost no one needs)
--- a/doc/bitstream_filters.texi
+++ b/doc/bitstream_filters.texi
@@ -30,7 +30,33 @@ ADTS AAC container to a FLV or a MOV/MP4 file.

 Remove zero padding at the end of a packet.

-@section dump_extradata
+@section dump_extra
+
+Add extradata to the beginning of the filtered packets.
+
+The additional argument specifies which packets should be filtered.
+It accepts the values:
+@table @samp
+@item a
+add extradata to all key packets, but only if @var{local_header} is
+set in the @option{flags2} codec context field
+
+@item k
+add extradata to all key packets
+
+@item e
+add extradata to all packets
+@end table
+
+If not specified it is assumed @samp{k}.
+
+For example the following @command{ffmpeg} command forces a global
+header (thus disabling individual packet headers) in the H.264 packets
+generated by the @code{libx264} encoder, but corrects them by adding
+the header stored in extradata to the key packets:
+@example
+ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
+@end example

@section h264_mp4toannexb

@@ -91,12 +117,10 @@ ffmpeg -i frame_%d.jpg -c:v copy rotated.avi

@section movsub

-@section mp3_header_compress
-
@section mp3_header_decompress

@section noise

-@section remove_extradata
+@section remove_extra

@c man end BITSTREAM FILTERS
--- a/doc/codecs.texi
+++ b/doc/codecs.texi
@@ -1,3 +1,4 @@
+@anchor{codec-options}
@chapter Codec Options
@c man begin CODEC OPTIONS

@@ -83,9 +84,6 @@ Apply interlaced motion estimation.
 Use closed gop.
@end table

-@item sub_id @var{integer}
-Deprecated, currently unused.
-
@item me_method @var{integer} (@emph{encoding,video})
 Set motion estimation method.

@@ -174,7 +172,13 @@ Set max video quantizer scale (VBR). Must be included between -1 and
 Set max difference between the quantizer scale (VBR).

@item bf @var{integer} (@emph{encoding,video})
-Set max number of B frames.
+Set max number of B frames between non-B-frames.
+
+Must be an integer between -1 and 16. 0 means that B-frames are
+disabled. If a value of -1 is used, it will choose an automatic value
+depending on the encoder.
+
+Default value is 0.

@item b_qfactor @var{float} (@emph{encoding,video})
 Set qp factor between P and B frames.
@@ -385,10 +389,6 @@ Possible values:

@item simplemmx

-@item libmpeg2mmx
-
-@item mmi
-
@item arm

@item altivec
@@ -405,10 +405,6 @@ Possible values:

@item simplealpha

-@item h264
-
-@item vp3
-
@item ipp

@item xvidmmx
@@ -771,26 +767,26 @@ Set noise reduction.
 Set number of bits which should be loaded into the rc buffer before
 decoding starts.

-@item inter_threshold @var{integer} (@emph{encoding,video})
-
@item flags2 @var{flags} (@emph{decoding/encoding,audio,video})

 Possible values:
@table @samp
@item fast
-allow non spec compliant speedup tricks
+Allow non spec compliant speedup tricks.
@item sgop
-Deprecated, use mpegvideo private options instead
+Deprecated, use mpegvideo private options instead.
@item noout
-skip bitstream encoding
+Skip bitstream encoding.
+@item ignorecrop
+Ignore cropping information from sps.
@item local_header
-place global headers at every keyframe instead of in extradata
+Place global headers at every keyframe instead of in extradata.
@item chunks
-Frame data might be split into multiple chunks
+Frame data might be split into multiple chunks.
@item showall
-Show all frames before the first keyframe
+Show all frames before the first keyframe.
@item skiprd
-Deprecated, use mpegvideo private options instead
+Deprecated, use mpegvideo private options instead.
@end table

@item error @var{integer} (@emph{encoding,video})
@@ -846,6 +842,10 @@ Possible values:

@item aac_eld

+@item mpeg2_aac_low
+
+@item mpeg2_aac_he
+
@item dts

@item dts_es
@@ -877,6 +877,9 @@ Set frame skip factor.

@item skip_exp @var{integer} (@emph{encoding,video})
 Set frame skip exponent.
+Negative values behave identical to the corresponding positive ones, except
+that the score is normalized.
+Positive values exist primarly for compatibility reasons and are not so useful.

@item skipcmp @var{integer} (@emph{encoding,video})
 Set frame skip compare function.
@@ -1064,9 +1067,34 @@ Set sample format audio decoders should prefer. Default value is

@item sub_charenc @var{encoding} (@emph{decoding,subtitles})
 Set the input subtitles character encoding.
+
+@item field_order  @var{field_order} (@emph{video})
+Set/override the field order of the video.
+Possible values:
+@table @samp
+@item progressive
+Progressive video
+@item tt
+Interlaced video, top field coded and displayed first
+@item bb
+Interlaced video, bottom field coded and displayed first
+@item tb
+Interlaced video, top coded first, bottom displayed first
+@item bt
+Interlaced video, bottom coded first, top displayed first
+@end table
+
+@item skip_alpha @var{integer} (@emph{decoding,video})
+Set to 1 to disable processing alpha (transparency). This works like the
+@samp{gray} flag in the @option{flags} option which skips chroma information
+instead of alpha. Default is 0.
@end table

@c man end CODEC OPTIONS

+@ifclear config-writeonly
@include decoders.texi
+@end ifclear
+@ifclear config-readonly
@include encoders.texi
+@end ifclear
--- a/doc/decoders.texi
+++ b/doc/decoders.texi
@@ -14,7 +14,7 @@ You can disable all the decoders with the configure option
 with the options @code{--enable-decoder=@var{DECODER}} /
@code{--disable-decoder=@var{DECODER}}.

-The option @code{-codecs} of the ff* tools will display the list of
+The option @code{-decoders} of the ff* tools will display the list of
 enabled decoders.

@c man end DECODERS
@@ -52,6 +52,37 @@ top-field-first is assumed
@chapter Audio Decoders
@c man begin AUDIO DECODERS

+A description of some of the currently available audio decoders
+follows.
+
+@section ac3
+
+AC-3 audio decoder.
+
+This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as well as
+the undocumented RealAudio 3 (a.k.a. dnet).
+
+@subsection AC-3 Decoder Options
+
+@table @option
+
+@item -drc_scale @var{value}
+Dynamic Range Scale Factor. The factor to apply to dynamic range values
+from the AC-3 stream. This factor is applied exponentially.
+There are 3 notable scale factor ranges:
+@table @option
+@item drc_scale == 0
+DRC disabled. Produces full range audio.
+@item 0 < drc_scale <= 1
+DRC enabled.  Applies a fraction of the stream DRC value.
+Audio reproduction is between full range and full compression.
+@item drc_scale > 1
+DRC enabled. Applies drc_scale asymmetrically.
+Loud sounds are fully compressed.  Soft sounds are enhanced.
+@end table
+
+@end table
+
@section ffwavesynth

 Internal wave synthetizer.
@@ -158,4 +189,45 @@ ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
@end table

+@section libzvbi-teletext
+
+Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext
+subtitles. Requires the presence of the libzvbi headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libzvbi}.
+
+@subsection Options
+
+@table @option
+@item txt_page
+List of teletext page numbers to decode. You may use the special * string to
+match all pages. Pages that do not match the specified list are dropped.
+Default value is *.
+@item txt_chop_top
+Discards the top teletext line. Default value is 1.
+@item txt_format
+Specifies the format of the decoded subtitles. The teletext decoder is capable
+of decoding the teletext pages to bitmaps or to simple text, you should use
+"bitmap" for teletext pages, because certain graphics and colors cannot be
+expressed in simple text. You might use "text" for teletext based subtitles if
+your application can handle simple text based subtitles. Default value is
+bitmap.
+@item txt_left
+X offset of generated bitmaps, default is 0.
+@item txt_top
+Y offset of generated bitmaps, default is 0.
+@item txt_chop_spaces
+Chops leading and trailing spaces and removes empty lines from the generated
+text. This option is useful for teletext based subtitles where empty spaces may
+be present at the start or at the end of the lines or empty lines may be
+present between the subtitle lines because of double-sized teletext charactes.
+Default value is 1.
+@item txt_duration
+Sets the display duration of the decoded teletext pages or subtitles in
+miliseconds. Default value is 30000 which is 30 seconds.
+@item txt_transparent
+Force transparent background of the generated teletext bitmaps. Default value
+is 0 which means an opaque (black) background.
+@end table
+
@c man end SUBTILES DECODERS
--- a/doc/default.css
+++ b/doc/default.css
@@ -1,3 +1,7 @@
+a.summary-letter {
+    text-decoration: none;
+}
+
 a {
    color: #2D6198;
 }
@@ -13,8 +17,8 @@ a:visited {
 }

 #banner img {
-    padding-bottom: 1px;
-    padding-top: 5px;
+    margin-bottom: 1px;
+    margin-top: 5px;
 }

 #body {
@@ -45,11 +49,16 @@ body {
    text-align: center;
 }

-h1, h2, h3 {
+h1 a, h2 a, h3 a, h4 a {
+    text-decoration: inherit;
+    color: inherit;
+}
+
+h1, h2, h3, h4 {
    padding-left: 0.4em;
    border-radius: 4px;
-    padding-bottom: 0.2em;
-    padding-top: 0.2em;
+    padding-bottom: 0.25em;
+    padding-top: 0.25em;
    border: 1px solid #6A996A;
 }

@@ -63,15 +72,22 @@ h1 {

 h2 {
    color: #313131;
-    font-size: 0.9em;
+    font-size: 1.0em;
    background-color: #ABE3AB;
 }

 h3 {
+    color: #313131;
+    font-size: 0.9em;
+    margin-bottom: -6px;
+    background-color: #BBF3BB;
+}
+
+h4 {
    color: #313131;
    font-size: 0.8em;
    margin-bottom: -8px;
-    background-color: #BBF3BB;
+    background-color: #D1FDD1;
 }

 img {
--- a/doc/demuxers.texi
+++ b/doc/demuxers.texi
@@ -1,7 +1,7 @@
@chapter Demuxers
@c man begin DEMUXERS

-Demuxers are configured elements in FFmpeg which allow to read the
+Demuxers are configured elements in FFmpeg that can read the
 multimedia streams from a particular type of file.

 When you configure your FFmpeg build, all the supported demuxers
@@ -29,6 +29,17 @@ the caller can decide which variant streams to actually receive.
 The total bitrate of the variant that the stream belongs to is
 available in a metadata key named "variant_bitrate".

+@section asf
+
+Advanced Systems Format demuxer.
+
+This demuxer is used to demux ASF files and MMS network streams.
+
+@table @option
+@item -no_resync_search @var{bool}
+Do not try to resynchronize by looking for a certain optional start code.
+@end table
+
@anchor{concat}
@section concat

@@ -103,6 +114,17 @@ probed and 0 otherwise.

@end table

+@section flv
+
+Adobe Flash Video Format demuxer.
+
+This demuxer is used to demux FLV files and RTMP network streams.
+
+@table @option
+@item -flv_metadata @var{bool}
+Allocate the streams according to the onMetaData array content.
+@end table
+
@section libgme

 The Game Music Emu library is a collection of video game music file emulators.
@@ -240,28 +262,41 @@ Use @command{ffmpeg} for creating a video from the images in the file
 sequence @file{img-001.jpeg}, @file{img-002.jpeg}, ..., assuming an
 input frame rate of 10 frames per second:
@example
-ffmpeg -i 'img-%03d.jpeg' -r 10 out.mkv
+ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv
@end example

@item
 As above, but start by reading from a file with index 100 in the sequence:
@example
-ffmpeg -start_number 100 -i 'img-%03d.jpeg' -r 10 out.mkv
+ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv
@end example

@item
 Read images matching the "*.png" glob pattern , that is all the files
 terminating with the ".png" suffix:
@example
-ffmpeg -pattern_type glob -i "*.png" -r 10 out.mkv
+ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
@end example
@end itemize

+@section mpegts
+
+MPEG-2 transport stream demuxer.
+
+@table @option
+
+@item fix_teletext_pts
+Overrides teletext packet PTS and DTS values with the timestamps calculated
+from the PCR of the first program which the teletext stream is part of and is
+not discarded. Default value is 1, set this option to 0 if you want your
+teletext packet PTS and DTS values untouched.
+@end table
+
@section rawvideo

 Raw video demuxer.

-This demuxer allows to read raw video data. Since there is no header
+This demuxer allows one to read raw video data. Since there is no header
 specifying the assumed video parameters, the user must specify them
 in order to be able to decode the data correctly.

--- a/doc/developer.texi
+++ b/doc/developer.texi
@@ -11,29 +11,23 @@

@chapter Developers Guide

-@section API
-@itemize @bullet
-@item libavcodec is the library containing the codecs (both encoding and
-decoding). Look at @file{doc/examples/decoding_encoding.c} to see how to use
-it.
+@section Notes for external developers

-@item libavformat is the library containing the file format handling (mux and
-demux code for several formats). Look at @file{ffplay.c} to use it in a
-player. See @file{doc/examples/muxing.c} to use it to generate audio or video
-streams.
+This document is mostly useful for internal FFmpeg developers.
+External developers who need to use the API in their application should
+refer to the API doxygen documentation in the public headers, and
+check the examples in @file{doc/examples} and in the source code to
+see how the public API is employed.

-@end itemize
+You can use the FFmpeg libraries in your commercial program, but you
+are encouraged to @emph{publish any patch you make}. In this case the
+best way to proceed is to send your patches to the ffmpeg-devel
+mailing list following the guidelines illustrated in the remainder of
+this document.

-@section Integrating libavcodec or libavformat in your program
-
-You can integrate all the source code of the libraries to link them
-statically to avoid any version problem. All you need is to provide a
-'config.mak' and a 'config.h' in the parent directory. See the defines
-generated by ./configure to understand what is needed.
-
-You can use libavcodec or libavformat in your commercial program, but
-@emph{any patch you make must be published}. The best way to proceed is
-to send your patches to the FFmpeg mailing list.
+For more detailed legal information about the use of FFmpeg in
+external programs read the @file{LICENSE} file in the source tree and
+consult @url{http://ffmpeg.org/legal.html}.

@section Contributing

@@ -57,13 +51,16 @@ and should try to fix issues their commit causes.
@subsection Code formatting conventions

 There are the following guidelines regarding the indentation in files:
+
@itemize @bullet
@item
 Indent size is 4.
+
@item
 The TAB character is forbidden outside of Makefiles as is any
 form of trailing whitespace. Commits containing either will be
 rejected by the git repository.
+
@item
 You should try to limit your code lines to 80 characters; however, do so if
 and only if this improves readability.
@@ -95,7 +92,7 @@ for markup commands, i.e. use @code{@@param} and not @code{\param}.
 * more text ...
 * ...
 */
-typedef struct Foobar@{
+typedef struct Foobar @{
    int var1; /**< var1 description */
    int var2; ///< var2 description
    /** var3 description */
@@ -117,13 +114,17 @@ int myfunc(int my_parameter)

 FFmpeg is programmed in the ISO C90 language with a few additional
 features from ISO C99, namely:
+
@itemize @bullet
@item
 the @samp{inline} keyword;
+
@item
@samp{//} comments;
+
@item
 designated struct initializers (@samp{struct s x = @{ .i = 17 @};})
+
@item
 compound literals (@samp{x = (struct s) @{ 17, 23 @};})
@end itemize
@@ -135,13 +136,17 @@ clarity and performance.
 All code must compile with recent versions of GCC and a number of other
 currently supported compilers. To ensure compatibility, please do not use
 additional C99 features or GCC extensions. Especially watch out for:
+
@itemize @bullet
@item
 mixing statements and declarations;
+
@item
@samp{long long} (use @samp{int64_t} instead);
+
@item
@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
+
@item
 GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
@end itemize
@@ -153,20 +158,25 @@ All names should be composed with underscores (_), not CamelCase. For example,
 for example structs and enums; they should always be in the CamelCase

 There are the following conventions for naming variables and functions:
+
@itemize @bullet
@item
 For local variables no prefix is required.
+
@item
 For file-scope variables and functions declared as @code{static}, no prefix
 is required.
+
@item
 For variables and functions visible outside of file scope, but only used
 internally by a library, an @code{ff_} prefix should be used,
 e.g. @samp{ff_w64_demuxer}.
+
@item
 For variables and functions visible outside of file scope, used internally
 across multiple libraries, use @code{avpriv_} as prefix, for example,
@samp{avpriv_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,
@@ -186,10 +196,12 @@ are reserved at the file level and may not be used for externally visible
 symbols. If in doubt, just avoid names starting with @code{_} altogether.

@subsection Miscellaneous conventions
+
@itemize @bullet
@item
 fprintf and printf are forbidden in libavformat and libavcodec,
 please use av_log() instead.
+
@item
 Casts should be used only when necessary. Unneeded parentheses
 should also be avoided if they don't make the code easier to understand.
@@ -232,131 +244,154 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:

@enumerate
@item
-   Contributions should be licensed under the
-   @uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
-   including an "or any later version" clause, or, if you prefer
-   a gift-style license, the
-   @uref{http://www.isc.org/software/license/, ISC} or
-   @uref{http://mit-license.org/, MIT} license.
-   @uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
-   an "or any later version" clause is also acceptable, but LGPL is
-   preferred.
-@item
-   You must not commit code which breaks FFmpeg! (Meaning unfinished but
-   enabled code which breaks compilation or compiles but does not work or
-   breaks the regression tests)
-   You can commit unfinished stuff (for testing etc), but it must be disabled
-   (#ifdef etc) by default so it does not interfere with other developers'
-   work.
-@item
-   The commit message should have a short first line in the form of
-   a @samp{topic: short description} as a header, separated by a newline
-   from the body consisting of an explanation of why the change is necessary.
-   If the commit fixes a known bug on the bug tracker, the commit message
-   should include its bug ID. Referring to the issue on the bug tracker does
-   not exempt you from writing an excerpt of the bug in the commit message.
-@item
-   You do not have to over-test things. If it works for you, and you think it
-   should work for others, then commit. If your code has problems
-   (portability, triggers compiler bugs, unusual environment etc) they will be
-   reported and eventually fixed.
-@item
-   Do not commit unrelated changes together, split them into self-contained
-   pieces. Also do not forget that if part B depends on part A, but A does not
-   depend on B, then A can and should be committed first and separate from B.
-   Keeping changes well split into self-contained parts makes reviewing and
-   understanding them on the commit log mailing list easier. This also helps
-   in case of debugging later on.
-   Also if you have doubts about splitting or not splitting, do not hesitate to
-   ask/discuss it on the developer mailing list.
-@item
-   Do not change behavior of the programs (renaming options etc) or public
-   API or ABI without first discussing it on the ffmpeg-devel mailing list.
-   Do not remove functionality from the code. Just improve!
+Contributions should be licensed under the
+@uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
+including an "or any later version" clause, or, if you prefer
+a gift-style license, the
+@uref{http://opensource.org/licenses/isc-license.txt, ISC} or
+@uref{http://mit-license.org/, MIT} license.
+@uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
+an "or any later version" clause is also acceptable, but LGPL is
+preferred.
+If you add a new file, give it a proper license header. Do not copy and
+paste it from a random place, use an existing file as template.

-   Note: Redundant code can be removed.
@item
-   Do not commit changes to the build system (Makefiles, configure script)
-   which change behavior, defaults etc, without asking first. The same
-   applies to compiler warning fixes, trivial looking fixes and to code
-   maintained by other developers. We usually have a reason for doing things
-   the way we do. Send your changes as patches to the ffmpeg-devel mailing
-   list, and if the code maintainers say OK, you may commit. This does not
-   apply to files you wrote and/or maintain.
-@item
-   We refuse source indentation and other cosmetic changes if they are mixed
-   with functional changes, such commits will be rejected and removed. Every
-   developer has his own indentation style, you should not change it. Of course
-   if you (re)write something, you can use your own style, even though we would
-   prefer if the indentation throughout FFmpeg was consistent (Many projects
-   force a given indentation style - we do not.). If you really need to make
-   indentation changes (try to avoid this), separate them strictly from real
-   changes.
+You must not commit code which breaks FFmpeg! (Meaning unfinished but
+enabled code which breaks compilation or compiles but does not work or
+breaks the regression tests)
+You can commit unfinished stuff (for testing etc), but it must be disabled
+(#ifdef etc) by default so it does not interfere with other developers'
+work.

-   NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
-   then either do NOT change the indentation of the inner part within (do not
-   move it to the right)! or do so in a separate commit
@item
-   Always fill out the commit log message. Describe in a few lines what you
-   changed and why. You can refer to mailing list postings if you fix a
-   particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
-   Recommended format:
-   area changed: Short 1 line description
+The commit message should have a short first line in the form of
+a @samp{topic: short description} as a header, separated by a newline
+from the body consisting of an explanation of why the change is necessary.
+If the commit fixes a known bug on the bug tracker, the commit message
+should include its bug ID. Referring to the issue on the bug tracker does
+not exempt you from writing an excerpt of the bug in the commit message.

-   details describing what and why and giving references.
@item
-   Make sure the author of the commit is set correctly. (see git commit --author)
-   If you apply a patch, send an
-   answer to ffmpeg-devel (or wherever you got the patch from) saying that
-   you applied the patch.
+You do not have to over-test things. If it works for you, and you think it
+should work for others, then commit. If your code has problems
+(portability, triggers compiler bugs, unusual environment etc) they will be
+reported and eventually fixed.
+
@item
-   When applying patches that have been discussed (at length) on the mailing
-   list, reference the thread in the log message.
+Do not commit unrelated changes together, split them into self-contained
+pieces. Also do not forget that if part B depends on part A, but A does not
+depend on B, then A can and should be committed first and separate from B.
+Keeping changes well split into self-contained parts makes reviewing and
+understanding them on the commit log mailing list easier. This also helps
+in case of debugging later on.
+Also if you have doubts about splitting or not splitting, do not hesitate to
+ask/discuss it on the developer mailing list.
+
@item
-    Do NOT commit to code actively maintained by others without permission.
-    Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
-    timeframe (12h for build failures and security fixes, 3 days small changes,
-    1 week for big patches) then commit your patch if you think it is OK.
-    Also note, the maintainer can simply ask for more time to review!
+Do not change behavior of the programs (renaming options etc) or public
+API or ABI without first discussing it on the ffmpeg-devel mailing list.
+Do not remove functionality from the code. Just improve!
+
+Note: Redundant code can be removed.
+
@item
-    Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
-    are sent there and reviewed by all the other developers. Bugs and possible
-    improvements or general questions regarding commits are discussed there. We
-    expect you to react if problems with your code are uncovered.
+Do not commit changes to the build system (Makefiles, configure script)
+which change behavior, defaults etc, without asking first. The same
+applies to compiler warning fixes, trivial looking fixes and to code
+maintained by other developers. We usually have a reason for doing things
+the way we do. Send your changes as patches to the ffmpeg-devel mailing
+list, and if the code maintainers say OK, you may commit. This does not
+apply to files you wrote and/or maintain.
+
@item
-    Update the documentation if you change behavior or add features. If you are
-    unsure how best to do this, send a patch to ffmpeg-devel, the documentation
-    maintainer(s) will review and commit your stuff.
+We refuse source indentation and other cosmetic changes if they are mixed
+with functional changes, such commits will be rejected and removed. Every
+developer has his own indentation style, you should not change it. Of course
+if you (re)write something, you can use your own style, even though we would
+prefer if the indentation throughout FFmpeg was consistent (Many projects
+force a given indentation style - we do not.). If you really need to make
+indentation changes (try to avoid this), separate them strictly from real
+changes.
+
+NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
+then either do NOT change the indentation of the inner part within (do not
+move it to the right)! or do so in a separate commit
+
@item
-    Try to keep important discussions and requests (also) on the public
-    developer mailing list, so that all developers can benefit from them.
+Always fill out the commit log message. Describe in a few lines what you
+changed and why. You can refer to mailing list postings if you fix a
+particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
+Recommended format:
+area changed: Short 1 line description
+
+details describing what and why and giving references.
+
@item
-    Never write to unallocated memory, never write over the end of arrays,
-    always check values read from some untrusted source before using them
-    as array index or other risky things.
+Make sure the author of the commit is set correctly. (see git commit --author)
+If you apply a patch, send an
+answer to ffmpeg-devel (or wherever you got the patch from) saying that
+you applied the patch.
+
@item
-    Remember to check if you need to bump versions for the specific libav*
-    parts (libavutil, libavcodec, libavformat) you are changing. You need
-    to change the version integer.
-    Incrementing the first component means no backward compatibility to
-    previous versions (e.g. removal of a function from the public API).
-    Incrementing the second component means backward compatible change
-    (e.g. addition of a function to the public API or extension of an
-    existing data structure).
-    Incrementing the third component means a noteworthy binary compatible
-    change (e.g. encoder bug fix that matters for the decoder). The third
-    component always starts at 100 to distinguish FFmpeg from Libav.
+When applying patches that have been discussed (at length) on the mailing
+list, reference the thread in the log message.
+
@item
-    Compiler warnings indicate potential bugs or code with bad style. If a type of
-    warning always points to correct and clean code, that warning should
-    be disabled, not the code changed.
-    Thus the remaining warnings can either be bugs or correct code.
-    If it is a bug, the bug has to be fixed. If it is not, the code should
-    be changed to not generate a warning unless that causes a slowdown
-    or obfuscates the code.
+Do NOT commit to code actively maintained by others without permission.
+Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
+timeframe (12h for build failures and security fixes, 3 days small changes,
+1 week for big patches) then commit your patch if you think it is OK.
+Also note, the maintainer can simply ask for more time to review!
+
@item
-    If you add a new file, give it a proper license header. Do not copy and
-    paste it from a random place, use an existing file as template.
+Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
+are sent there and reviewed by all the other developers. Bugs and possible
+improvements or general questions regarding commits are discussed there. We
+expect you to react if problems with your code are uncovered.
+
+@item
+Update the documentation if you change behavior or add features. If you are
+unsure how best to do this, send a patch to ffmpeg-devel, the documentation
+maintainer(s) will review and commit your stuff.
+
+@item
+Try to keep important discussions and requests (also) on the public
+developer mailing list, so that all developers can benefit from them.
+
+@item
+Never write to unallocated memory, never write over the end of arrays,
+always check values read from some untrusted source before using them
+as array index or other risky things.
+
+@item
+Remember to check if you need to bump versions for the specific libav*
+parts (libavutil, libavcodec, libavformat) you are changing. You need
+to change the version integer.
+Incrementing the first component means no backward compatibility to
+previous versions (e.g. removal of a function from the public API).
+Incrementing the second component means backward compatible change
+(e.g. addition of a function to the public API or extension of an
+existing data structure).
+Incrementing the third component means a noteworthy binary compatible
+change (e.g. encoder bug fix that matters for the decoder). The third
+component always starts at 100 to distinguish FFmpeg from Libav.
+
+@item
+Compiler warnings indicate potential bugs or code with bad style. If a type of
+warning always points to correct and clean code, that warning should
+be disabled, not the code changed.
+Thus the remaining warnings can either be bugs or correct code.
+If it is a bug, the bug has to be fixed. If it is not, the code should
+be changed to not generate a warning unless that causes a slowdown
+or obfuscates the code.
+
+@item
+Make sure that no parts of the codebase that you maintain are missing from the
+@file{MAINTAINERS} file. If something that you want to maintain is missing add it with
+your name after it.
+If at some point you no longer want to maintain some code, then please help
+finding a new maintainer and also don't forget updating the @file{MAINTAINERS} file.
@end enumerate

 We think our rules are not too hard. If you have comments, contact us.
@@ -411,40 +446,51 @@ send a reminder by email. Your patch should eventually be dealt with.

@enumerate
@item
-    Did you use av_cold for codec initialization and close functions?
+Did you use av_cold for codec initialization and close functions?
+
@item
-    Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
-    AVInputFormat/AVOutputFormat struct?
+Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
+AVInputFormat/AVOutputFormat struct?
+
@item
-    Did you bump the minor version number (and reset the micro version
-    number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
+Did you bump the minor version number (and reset the micro version
+number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
+
@item
-    Did you register it in @file{allcodecs.c} or @file{allformats.c}?
+Did you register it in @file{allcodecs.c} or @file{allformats.c}?
+
@item
-    Did you add the AVCodecID to @file{avcodec.h}?
-    When adding new codec IDs, also add an entry to the codec descriptor
-    list in @file{libavcodec/codec_desc.c}.
+Did you add the AVCodecID to @file{avcodec.h}?
+When adding new codec IDs, also add an entry to the codec descriptor
+list in @file{libavcodec/codec_desc.c}.
+
@item
-    If it has a FourCC, did you add it to @file{libavformat/riff.c},
-    even if it is only a decoder?
+If it has a FourCC, did you add it to @file{libavformat/riff.c},
+even if it is only a decoder?
+
@item
-    Did you add a rule to compile the appropriate files in the Makefile?
-    Remember to do this even if you're just adding a format to a file that is
-    already being compiled by some other rule, like a raw demuxer.
+Did you add a rule to compile the appropriate files in the Makefile?
+Remember to do this even if you're just adding a format to a file that is
+already being compiled by some other rule, like a raw demuxer.
+
@item
-    Did you add an entry to the table of supported formats or codecs in
-    @file{doc/general.texi}?
+Did you add an entry to the table of supported formats or codecs in
+@file{doc/general.texi}?
+
@item
-    Did you add an entry in the Changelog?
+Did you add an entry in the Changelog?
+
@item
-    If it depends on a parser or a library, did you add that dependency in
-    configure?
+If it depends on a parser or a library, did you add that dependency in
+configure?
+
@item
-    Did you @code{git add} the appropriate files before committing?
+Did you @code{git add} the appropriate files before committing?
+
@item
-    Did you make sure it compiles standalone, i.e. with
-    @code{configure --disable-everything --enable-decoder=foo}
-    (or @code{--enable-demuxer} or whatever your component is)?
+Did you make sure it compiles standalone, i.e. with
+@code{configure --disable-everything --enable-decoder=foo}
+(or @code{--enable-demuxer} or whatever your component is)?
@end enumerate


@@ -452,82 +498,109 @@ send a reminder by email. Your patch should eventually be dealt with.

@enumerate
@item
-    Does @code{make fate} pass with the patch applied?
+Does @code{make fate} pass with the patch applied?
+
@item
-    Was the patch generated with git format-patch or send-email?
+Was the patch generated with git format-patch or send-email?
+
@item
-    Did you sign off your patch? (git commit -s)
-    See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
-    of sign off.
+Did you sign off your patch? (git commit -s)
+See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
+of sign off.
+
@item
-    Did you provide a clear git commit log message?
+Did you provide a clear git commit log message?
+
@item
-    Is the patch against latest FFmpeg git master branch?
+Is the patch against latest FFmpeg git master branch?
+
@item
-    Are you subscribed to ffmpeg-devel?
-    (the list is subscribers only due to spam)
+Are you subscribed to ffmpeg-devel?
+(the list is subscribers only due to spam)
+
@item
-    Have you checked that the changes are minimal, so that the same cannot be
-    achieved with a smaller patch and/or simpler final code?
+Have you checked that the changes are minimal, so that the same cannot be
+achieved with a smaller patch and/or simpler final code?
+
@item
-    If the change is to speed critical code, did you benchmark it?
+If the change is to speed critical code, did you benchmark it?
+
@item
-    If you did any benchmarks, did you provide them in the mail?
+If you did any benchmarks, did you provide them in the mail?
+
@item
-    Have you checked that the patch does not introduce buffer overflows or
-    other security issues?
+Have you checked that the patch does not introduce buffer overflows or
+other security issues?
+
@item
-    Did you test your decoder or demuxer against damaged data? If no, see
-    tools/trasher, the noise bitstream filter, and
-    @uref{http://caca.zoy.org/wiki/zzuf, zzuf}. Your decoder or demuxer
-    should not crash, end in a (near) infinite loop, or allocate ridiculous
-    amounts of memory when fed damaged data.
+Did you test your decoder or demuxer against damaged data? If no, see
+tools/trasher, the noise bitstream filter, and
+@uref{http://caca.zoy.org/wiki/zzuf, zzuf}. Your decoder or demuxer
+should not crash, end in a (near) infinite loop, or allocate ridiculous
+amounts of memory when fed damaged data.
+
@item
-    Does the patch not mix functional and cosmetic changes?
+Does the patch not mix functional and cosmetic changes?
+
@item
-    Did you add tabs or trailing whitespace to the code? Both are forbidden.
+Did you add tabs or trailing whitespace to the code? Both are forbidden.
+
@item
-    Is the patch attached to the email you send?
+Is the patch attached to the email you send?
+
@item
-    Is the mime type of the patch correct? It should be text/x-diff or
-    text/x-patch or at least text/plain and not application/octet-stream.
+Is the mime type of the patch correct? It should be text/x-diff or
+text/x-patch or at least text/plain and not application/octet-stream.
+
@item
-    If the patch fixes a bug, did you provide a verbose analysis of the bug?
+If the patch fixes a bug, did you provide a verbose analysis of the bug?
+
@item
-    If the patch fixes a bug, did you provide enough information, including
-    a sample, so the bug can be reproduced and the fix can be verified?
-    Note please do not attach samples >100k to mails but rather provide a
-    URL, you can upload to ftp://upload.ffmpeg.org
+If the patch fixes a bug, did you provide enough information, including
+a sample, so the bug can be reproduced and the fix can be verified?
+Note please do not attach samples >100k to mails but rather provide a
+URL, you can upload to ftp://upload.ffmpeg.org
+
@item
-    Did you provide a verbose summary about what the patch does change?
+Did you provide a verbose summary about what the patch does change?
+
@item
-    Did you provide a verbose explanation why it changes things like it does?
+Did you provide a verbose explanation why it changes things like it does?
+
@item
-    Did you provide a verbose summary of the user visible advantages and
-    disadvantages if the patch is applied?
+Did you provide a verbose summary of the user visible advantages and
+disadvantages if the patch is applied?
+
@item
-    Did you provide an example so we can verify the new feature added by the
-    patch easily?
+Did you provide an example so we can verify the new feature added by the
+patch easily?
+
@item
-    If you added a new file, did you insert a license header? It should be
-    taken from FFmpeg, not randomly copied and pasted from somewhere else.
+If you added a new file, did you insert a license header? It should be
+taken from FFmpeg, not randomly copied and pasted from somewhere else.
+
@item
-    You should maintain alphabetical order in alphabetically ordered lists as
-    long as doing so does not break API/ABI compatibility.
+You should maintain alphabetical order in alphabetically ordered lists as
+long as doing so does not break API/ABI compatibility.
+
@item
-    Lines with similar content should be aligned vertically when doing so
-    improves readability.
+Lines with similar content should be aligned vertically when doing so
+improves readability.
+
@item
-    Consider to add a regression test for your code.
+Consider to add a regression test for your code.
+
@item
-    If you added YASM code please check that things still work with --disable-yasm
+If you added YASM code please check that things still work with --disable-yasm
+
@item
-    Make sure you check the return values of function and return appropriate
-    error codes. Especially memory allocation functions like @code{av_malloc()}
-    are notoriously left unchecked, which is a serious problem.
+Make sure you check the return values of function and return appropriate
+error codes. Especially memory allocation functions like @code{av_malloc()}
+are notoriously left unchecked, which is a serious problem.
+
@item
-    Test your code with valgrind and or Address Sanitizer to ensure it's free
-    of leaks, out of array accesses, etc.
+Test your code with valgrind and or Address Sanitizer to ensure it's free
+of leaks, out of array accesses, etc.
@end enumerate

@section Patch review process
@@ -590,12 +663,15 @@ the following steps:
@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
@@ -630,12 +706,13 @@ There are two kinds of releases:

@enumerate
@item
-    @strong{Major releases} always include the latest and greatest
-    features and functionality.
+@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.
+@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
@@ -656,15 +733,18 @@ 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/}.
+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}.
+Fixes a documented bug in @url{https://trac.ffmpeg.org}.
+
@item
-    Improves the included documentation.
+Improves the included documentation.
+
@item
-    Retains both source code and binary compatibility with previous
-    point releases of the same release branch.
+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.
@@ -676,33 +756,42 @@ The release process involves the following steps:

@enumerate
@item
-    Ensure that the @file{RELEASE} file contains the version number for
-    the upcoming release.
+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}.
+Add the release at @url{https://trac.ffmpeg.org/admin/ticket/versions}.
+
@item
-    Announce the intent to do a release to the mailing list.
+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}.
+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}).
+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
+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.
+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.
+Propose and send a patch to the @strong{ffmpeg-devel} mailing list
+with a news entry for the website.
+
@item
-    Publish the news entry.
+Publish the news entry.
+
@item
-    Send announcement to the mailing list.
+Send announcement to the mailing list.
@end enumerate

@bye
--- a/doc/devices.texi
+++ b/doc/devices.texi
@@ -17,5 +17,9 @@ for programmatic use.

@c man end DEVICE OPTIONS

+@ifclear config-writeonly
@include indevs.texi
+@end ifclear
+@ifclear config-readonly
@include outdevs.texi
+@end ifclear
--- a/doc/doxy-wrapper.sh
+++ b/doc/doxy-wrapper.sh
@@ -8,7 +8,5 @@ shift 2
 doxygen - <<EOF
@INCLUDE        = ${DOXYFILE}
 INPUT           = $@
-HTML_HEADER     = ${SRC_PATH}/doc/doxy/header.html
-HTML_FOOTER     = ${SRC_PATH}/doc/doxy/footer.html
-HTML_STYLESHEET = ${SRC_PATH}/doc/doxy/doxy_stylesheet.css
+EXAMPLE_PATH    = ${SRC_PATH}/doc/examples
 EOF
--- a/doc/doxy/doxy_stylesheet.css
+++ b/doc/doxy/doxy_stylesheet.css
--- a/doc/doxy/footer.html
+++ b/doc/doxy/footer.html
@@ -1,9 +0,0 @@
-
- <footer class="footer pagination-right">
-   <span class="label label-info">
-    Generated on $datetime for $projectname by&#160;<a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion
-   </span>
- </footer>
-</div>
-</body>
-</html>
--- a/doc/doxy/header.html
+++ b/doc/doxy/header.html
@@ -1,16 +0,0 @@
-<!DOCTYPE html>
-<html>
-<head>
-<meta http-equiv="Content-Type" content="text/html; charset=UTF-8"/>
-<meta http-equiv="X-UA-Compatible" content="IE=9"/>
-<!--BEGIN PROJECT_NAME--><title>$projectname: $title</title><!--END PROJECT_NAME-->
-<!--BEGIN !PROJECT_NAME--><title>$title</title><!--END !PROJECT_NAME-->
-<link href="$relpath$doxy_stylesheet.css" rel="stylesheet" type="text/css" />
-<!--Header replace -->
-
-</head>
-
-<div class="container">
-
-<!--Header replace -->
-<div class="menu">
--- a/doc/encoders.texi
+++ b/doc/encoders.texi
--- a/doc/examples/Makefile
+++ b/doc/examples/Makefile
@@ -11,20 +11,24 @@ CFLAGS += -Wall -g
 CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
 LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)

-EXAMPLES=       decoding_encoding                  \
-                demuxing                           \
+EXAMPLES=       avio_reading                       \
+                avcodec                            \
+                demuxing_decoding                  \
                filtering_video                    \
                filtering_audio                    \
                metadata                           \
                muxing                             \
+                remuxing                           \
                resampling_audio                   \
                scaling_video                      \
+                transcode_aac                      \

 OBJS=$(addsuffix .o,$(EXAMPLES))

 # the following examples make explicit use of the math library
-decoding_encoding: LDLIBS += -lm
+avcodec:           LDLIBS += -lm
 muxing:            LDLIBS += -lm
+resampling_audio:  LDLIBS += -lm

 .phony: all clean-test clean

--- a/doc/examples/README
+++ b/doc/examples/README
@@ -5,14 +5,19 @@ Both following use cases rely on pkg-config and make, thus make sure
 that you have them installed and working on your system.


-1) Build the installed examples in a generic read/write user directory
+Method 1: build the installed examples in a generic read/write user directory

 Copy to a read/write user directory and just use "make", it will link
 to the libraries on your system, assuming the PKG_CONFIG_PATH is
 correctly configured.

-2) Build the examples in-tree
+Method 2: build the examples in-tree

 Assuming you are in the source FFmpeg checkout directory, you need to build
-FFmpeg (no need to make install in any prefix). Then you can go into the
-doc/examples and run a command such as PKG_CONFIG_PATH=pc-uninstalled make.
+FFmpeg (no need to make install in any prefix). Then just run "make examples".
+This will build the examples using the FFmpeg build system. You can clean those
+examples using "make examplesclean"
+
+If you want to try the dedicated Makefile examples (to emulate the first
+method), go into doc/examples and run a command such as
+PKG_CONFIG_PATH=pc-uninstalled make.
--- a/doc/examples/decoding_encoding.c
+++ b/doc/examples/decoding_encoding.c
@@ -24,10 +24,10 @@
 * @file
 * libavcodec API use example.
 *
+ * @example avcodec.c
 * Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),
 * not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the
 * format handling
- * @example doc/examples/decoding_encoding.c
 */

 #include <math.h>
@@ -156,7 +156,7 @@ static void audio_encode_example(const char *filename)
    }

    /* frame containing input raw audio */
-    frame = avcodec_alloc_frame();
+    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate audio frame\n");
        exit(1);
@@ -170,6 +170,10 @@ static void audio_encode_example(const char *filename)
     * we calculate the size of the samples buffer in bytes */
    buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
                                             c->sample_fmt, 0);
+    if (buffer_size < 0) {
+        fprintf(stderr, "Could not get sample buffer size\n");
+        exit(1);
+    }
    samples = av_malloc(buffer_size);
    if (!samples) {
        fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
@@ -187,7 +191,7 @@ static void audio_encode_example(const char *filename)
    /* encode a single tone sound */
    t = 0;
    tincr = 2 * M_PI * 440.0 / c->sample_rate;
-    for(i=0;i<200;i++) {
+    for (i = 0; i < 200; i++) {
        av_init_packet(&pkt);
        pkt.data = NULL; // packet data will be allocated by the encoder
        pkt.size = 0;
@@ -227,7 +231,7 @@ static void audio_encode_example(const char *filename)
    fclose(f);

    av_freep(&samples);
-    avcodec_free_frame(&frame);
+    av_frame_free(&frame);
    avcodec_close(c);
    av_free(c);
 }
@@ -287,12 +291,11 @@ static void audio_decode_example(const char *outfilename, const char *filename)
        int got_frame = 0;

        if (!decoded_frame) {
-            if (!(decoded_frame = avcodec_alloc_frame())) {
+            if (!(decoded_frame = av_frame_alloc())) {
                fprintf(stderr, "Could not allocate audio frame\n");
                exit(1);
            }
-        } else
-            avcodec_get_frame_defaults(decoded_frame);
+        }

        len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
        if (len < 0) {
@@ -304,6 +307,11 @@ static void audio_decode_example(const char *outfilename, const char *filename)
            int data_size = av_samples_get_buffer_size(NULL, c->channels,
                                                       decoded_frame->nb_samples,
                                                       c->sample_fmt, 1);
+            if (data_size < 0) {
+                /* This should not occur, checking just for paranoia */
+                fprintf(stderr, "Failed to calculate data size\n");
+                exit(1);
+            }
            fwrite(decoded_frame->data[0], 1, data_size, outfile);
        }
        avpkt.size -= len;
@@ -329,7 +337,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)

    avcodec_close(c);
    av_free(c);
-    avcodec_free_frame(&decoded_frame);
+    av_frame_free(&decoded_frame);
 }

 /*
@@ -366,12 +374,12 @@ static void video_encode_example(const char *filename, int codec_id)
    c->width = 352;
    c->height = 288;
    /* frames per second */
-    c->time_base= (AVRational){1,25};
+    c->time_base = (AVRational){1,25};
    c->gop_size = 10; /* emit one intra frame every ten frames */
-    c->max_b_frames=1;
+    c->max_b_frames = 1;
    c->pix_fmt = AV_PIX_FMT_YUV420P;

-    if(codec_id == AV_CODEC_ID_H264)
+    if (codec_id == AV_CODEC_ID_H264)
        av_opt_set(c->priv_data, "preset", "slow", 0);

    /* open it */
@@ -386,7 +394,7 @@ static void video_encode_example(const char *filename, int codec_id)
        exit(1);
    }

-    frame = avcodec_alloc_frame();
+    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate video frame\n");
        exit(1);
@@ -405,7 +413,7 @@ static void video_encode_example(const char *filename, int codec_id)
    }

    /* encode 1 second of video */
-    for(i=0;i<25;i++) {
+    for (i = 0; i < 25; i++) {
        av_init_packet(&pkt);
        pkt.data = NULL;    // packet data will be allocated by the encoder
        pkt.size = 0;
@@ -413,15 +421,15 @@ static void video_encode_example(const char *filename, int codec_id)
        fflush(stdout);
        /* prepare a dummy image */
        /* Y */
-        for(y=0;y<c->height;y++) {
-            for(x=0;x<c->width;x++) {
+        for (y = 0; y < c->height; y++) {
+            for (x = 0; x < c->width; x++) {
                frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
            }
        }

        /* Cb and Cr */
-        for(y=0;y<c->height/2;y++) {
-            for(x=0;x<c->width/2;x++) {
+        for (y = 0; y < c->height/2; y++) {
+            for (x = 0; x < c->width/2; x++) {
                frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
                frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
            }
@@ -467,7 +475,7 @@ static void video_encode_example(const char *filename, int codec_id)
    avcodec_close(c);
    av_free(c);
    av_freep(&frame->data[0]);
-    avcodec_free_frame(&frame);
+    av_frame_free(&frame);
    printf("\n");
 }

@@ -481,10 +489,10 @@ static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
    FILE *f;
    int i;

-    f=fopen(filename,"w");
-    fprintf(f,"P5\n%d %d\n%d\n",xsize,ysize,255);
-    for(i=0;i<ysize;i++)
-        fwrite(buf + i * wrap,1,xsize,f);
+    f = fopen(filename,"w");
+    fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
+    for (i = 0; i < ysize; i++)
+        fwrite(buf + i * wrap, 1, xsize, f);
    fclose(f);
 }

@@ -565,14 +573,14 @@ static void video_decode_example(const char *outfilename, const char *filename)
        exit(1);
    }

-    frame = avcodec_alloc_frame();
+    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate video frame\n");
        exit(1);
    }

    frame_count = 0;
-    for(;;) {
+    for (;;) {
        avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
        if (avpkt.size == 0)
            break;
@@ -609,7 +617,7 @@ static void video_decode_example(const char *outfilename, const char *filename)

    avcodec_close(c);
    av_free(c);
-    avcodec_free_frame(&frame);
+    av_frame_free(&frame);
    printf("\n");
 }

--- a/doc/examples/avio_reading.c
+++ b/doc/examples/avio_reading.c
@@ -0,0 +1,134 @@
+/*
+ * Copyright (c) 2014 Stefano Sabatini
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libavformat AVIOContext API example.
+ *
+ * Make libavformat demuxer access media content through a custom
+ * AVIOContext read callback.
+ * @example avio_reading.c
+ */
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavformat/avio.h>
+#include <libavutil/file.h>
+
+struct buffer_data {
+    uint8_t *ptr;
+    size_t size; ///< size left in the buffer
+};
+
+static int read_packet(void *opaque, uint8_t *buf, int buf_size)
+{
+    struct buffer_data *bd = (struct buffer_data *)opaque;
+    buf_size = FFMIN(buf_size, bd->size);
+
+    printf("ptr:%p size:%zu\n", bd->ptr, bd->size);
+
+    /* copy internal buffer data to buf */
+    memcpy(buf, bd->ptr, buf_size);
+    bd->ptr  += buf_size;
+    bd->size -= buf_size;
+
+    return buf_size;
+}
+
+int main(int argc, char *argv[])
+{
+    AVFormatContext *fmt_ctx = NULL;
+    AVIOContext *avio_ctx = NULL;
+    uint8_t *buffer = NULL, *avio_ctx_buffer = NULL;
+    size_t buffer_size, avio_ctx_buffer_size = 4096;
+    char *input_filename = NULL;
+    int ret = 0;
+    struct buffer_data bd = { 0 };
+
+    if (argc != 2) {
+        fprintf(stderr, "usage: %s input_file\n"
+                "API example program to show how to read from a custom buffer "
+                "accessed through AVIOContext.\n", argv[0]);
+        return 1;
+    }
+    input_filename = argv[1];
+
+    /* register codecs and formats and other lavf/lavc components*/
+    av_register_all();
+
+    /* slurp file content into buffer */
+    ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL);
+    if (ret < 0)
+        goto end;
+
+    /* fill opaque structure used by the AVIOContext read callback */
+    bd.ptr  = buffer;
+    bd.size = buffer_size;
+
+    if (!(fmt_ctx = avformat_alloc_context())) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    avio_ctx_buffer = av_malloc(avio_ctx_buffer_size);
+    if (!avio_ctx_buffer) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+    avio_ctx = avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size,
+                                  0, &bd, &read_packet, NULL, NULL);
+    if (!avio_ctx) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+    fmt_ctx->pb = avio_ctx;
+
+    ret = avformat_open_input(&fmt_ctx, NULL, NULL, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Could not open input\n");
+        goto end;
+    }
+
+    ret = avformat_find_stream_info(fmt_ctx, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Could not find stream information\n");
+        goto end;
+    }
+
+    av_dump_format(fmt_ctx, 0, input_filename, 0);
+
+end:
+    avformat_close_input(&fmt_ctx);
+    /* note: the internal buffer could have changed, and be != avio_ctx_buffer */
+    if (avio_ctx) {
+        av_freep(&avio_ctx->buffer);
+        av_freep(&avio_ctx);
+    }
+    av_file_unmap(buffer, buffer_size);
+
+    if (ret < 0) {
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        return 1;
+    }
+
+    return 0;
+}
--- a/doc/examples/demuxing_decoding.c
+++ b/doc/examples/demuxing_decoding.c
@@ -22,11 +22,11 @@

 /**
 * @file
- * libavformat demuxing API use example.
+ * Demuxing and decoding example.
 *
 * Show how to use the libavformat and libavcodec API to demux and
 * decode audio and video data.
- * @example doc/examples/demuxing.c
+ * @example demuxing_decoding.c
 */

 #include <libavutil/imgutils.h>
@@ -47,25 +47,36 @@ static uint8_t *video_dst_data[4] = {NULL};
 static int      video_dst_linesize[4];
 static int video_dst_bufsize;

-static uint8_t **audio_dst_data = NULL;
-static int       audio_dst_linesize;
-static int audio_dst_bufsize;
-
 static int video_stream_idx = -1, audio_stream_idx = -1;
 static AVFrame *frame = NULL;
 static AVPacket pkt;
 static int video_frame_count = 0;
 static int audio_frame_count = 0;

+/* The different ways of decoding and managing data memory. You are not
+ * supposed to support all the modes in your application but pick the one most
+ * appropriate to your needs. Look for the use of api_mode in this example to
+ * see what are the differences of API usage between them */
+enum {
+    API_MODE_OLD                  = 0, /* old method, deprecated */
+    API_MODE_NEW_API_REF_COUNT    = 1, /* new method, using the frame reference counting */
+    API_MODE_NEW_API_NO_REF_COUNT = 2, /* new method, without reference counting */
+};
+
+static int api_mode = API_MODE_OLD;
+
 static int decode_packet(int *got_frame, int cached)
 {
    int ret = 0;
+    int decoded = pkt.size;
+
+    *got_frame = 0;

    if (pkt.stream_index == video_stream_idx) {
        /* decode video frame */
        ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
        if (ret < 0) {
-            fprintf(stderr, "Error decoding video frame\n");
+            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
            return ret;
        }

@@ -88,40 +99,40 @@ static int decode_packet(int *got_frame, int cached)
        /* decode audio frame */
        ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
        if (ret < 0) {
-            fprintf(stderr, "Error decoding audio frame\n");
+            fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret));
            return ret;
        }
+        /* Some audio decoders decode only part of the packet, and have to be
+         * called again with the remainder of the packet data.
+         * Sample: fate-suite/lossless-audio/luckynight-partial.shn
+         * Also, some decoders might over-read the packet. */
+        decoded = FFMIN(ret, pkt.size);

        if (*got_frame) {
+            size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
            printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
                   cached ? "(cached)" : "",
                   audio_frame_count++, frame->nb_samples,
                   av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));

-            ret = av_samples_alloc(audio_dst_data, &audio_dst_linesize, av_frame_get_channels(frame),
-                                   frame->nb_samples, frame->format, 1);
-            if (ret < 0) {
-                fprintf(stderr, "Could not allocate audio buffer\n");
-                return AVERROR(ENOMEM);
-            }
-
-            /* TODO: extend return code of the av_samples_* functions so that this call is not needed */
-            audio_dst_bufsize =
-                av_samples_get_buffer_size(NULL, av_frame_get_channels(frame),
-                                           frame->nb_samples, frame->format, 1);
-
-            /* copy audio data to destination buffer:
-             * this is required since rawaudio expects non aligned data */
-            av_samples_copy(audio_dst_data, frame->data, 0, 0,
-                            frame->nb_samples, av_frame_get_channels(frame), frame->format);
-
-            /* write to rawaudio file */
-            fwrite(audio_dst_data[0], 1, audio_dst_bufsize, audio_dst_file);
-            av_freep(&audio_dst_data[0]);
+            /* Write the raw audio data samples of the first plane. This works
+             * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
+             * most audio decoders output planar audio, which uses a separate
+             * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
+             * In other words, this code will write only the first audio channel
+             * in these cases.
+             * You should use libswresample or libavfilter to convert the frame
+             * to packed data. */
+            fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
        }
    }

-    return ret;
+    /* If we use the new API with reference counting, we own the data and need
+     * to de-reference it when we don't use it anymore */
+    if (*got_frame && api_mode == API_MODE_NEW_API_REF_COUNT)
+        av_frame_unref(frame);
+
+    return decoded;
 }

 static int open_codec_context(int *stream_idx,
@@ -131,6 +142,7 @@ static int open_codec_context(int *stream_idx,
    AVStream *st;
    AVCodecContext *dec_ctx = NULL;
    AVCodec *dec = NULL;
+    AVDictionary *opts = NULL;

    ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
    if (ret < 0) {
@@ -147,10 +159,13 @@ static int open_codec_context(int *stream_idx,
        if (!dec) {
            fprintf(stderr, "Failed to find %s codec\n",
                    av_get_media_type_string(type));
-            return ret;
+            return AVERROR(EINVAL);
        }

-        if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
+        /* Init the decoders, with or without reference counting */
+        if (api_mode == API_MODE_NEW_API_REF_COUNT)
+            av_dict_set(&opts, "refcounted_frames", "1", 0);
+        if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
            fprintf(stderr, "Failed to open %s codec\n",
                    av_get_media_type_string(type));
            return ret;
@@ -193,15 +208,31 @@ int main (int argc, char **argv)
 {
    int ret = 0, got_frame;

-    if (argc != 4) {
-        fprintf(stderr, "usage: %s input_file video_output_file audio_output_file\n"
+    if (argc != 4 && argc != 5) {
+        fprintf(stderr, "usage: %s [-refcount=<old|new_norefcount|new_refcount>] "
+                "input_file video_output_file audio_output_file\n"
                "API example program to show how to read frames from an input file.\n"
                "This program reads frames from a file, decodes them, and writes decoded\n"
                "video frames to a rawvideo file named video_output_file, and decoded\n"
-                "audio frames to a rawaudio file named audio_output_file.\n"
+                "audio frames to a rawaudio file named audio_output_file.\n\n"
+                "If the -refcount option is specified, the program use the\n"
+                "reference counting frame system which allows keeping a copy of\n"
+                "the data for longer than one decode call. If unset, it's using\n"
+                "the classic old method.\n"
                "\n", argv[0]);
        exit(1);
    }
+    if (argc == 5) {
+        const char *mode = argv[1] + strlen("-refcount=");
+        if      (!strcmp(mode, "old"))            api_mode = API_MODE_OLD;
+        else if (!strcmp(mode, "new_norefcount")) api_mode = API_MODE_NEW_API_NO_REF_COUNT;
+        else if (!strcmp(mode, "new_refcount"))   api_mode = API_MODE_NEW_API_REF_COUNT;
+        else {
+            fprintf(stderr, "unknow mode '%s'\n", mode);
+            exit(1);
+        }
+        argv++;
+    }
    src_filename = argv[1];
    video_dst_filename = argv[2];
    audio_dst_filename = argv[3];
@@ -244,8 +275,6 @@ int main (int argc, char **argv)
    }

    if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
-        int nb_planes;
-
        audio_stream = fmt_ctx->streams[audio_stream_idx];
        audio_dec_ctx = audio_stream->codec;
        audio_dst_file = fopen(audio_dst_filename, "wb");
@@ -254,15 +283,6 @@ int main (int argc, char **argv)
            ret = 1;
            goto end;
        }
-
-        nb_planes = av_sample_fmt_is_planar(audio_dec_ctx->sample_fmt) ?
-            audio_dec_ctx->channels : 1;
-        audio_dst_data = av_mallocz(sizeof(uint8_t *) * nb_planes);
-        if (!audio_dst_data) {
-            fprintf(stderr, "Could not allocate audio data buffers\n");
-            ret = AVERROR(ENOMEM);
-            goto end;
-        }
    }

    /* dump input information to stderr */
@@ -274,7 +294,12 @@ int main (int argc, char **argv)
        goto end;
    }

-    frame = avcodec_alloc_frame();
+    /* When using the new API, you need to use the libavutil/frame.h API, while
+     * the classic frame management is available in libavcodec */
+    if (api_mode == API_MODE_OLD)
+        frame = avcodec_alloc_frame();
+    else
+        frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate frame\n");
        ret = AVERROR(ENOMEM);
@@ -293,8 +318,15 @@ int main (int argc, char **argv)

    /* read frames from the file */
    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
-        decode_packet(&got_frame, 0);
-        av_free_packet(&pkt);
+        AVPacket orig_pkt = pkt;
+        do {
+            ret = decode_packet(&got_frame, 0);
+            if (ret < 0)
+                break;
+            pkt.data += ret;
+            pkt.size -= ret;
+        } while (pkt.size > 0);
+        av_free_packet(&orig_pkt);
    }

    /* flush cached frames */
@@ -314,29 +346,41 @@ int main (int argc, char **argv)
    }

    if (audio_stream) {
+        enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
+        int n_channels = audio_dec_ctx->channels;
        const char *fmt;

-        if ((ret = get_format_from_sample_fmt(&fmt, audio_dec_ctx->sample_fmt)) < 0)
+        if (av_sample_fmt_is_planar(sfmt)) {
+            const char *packed = av_get_sample_fmt_name(sfmt);
+            printf("Warning: the sample format the decoder produced is planar "
+                   "(%s). This example will output the first channel only.\n",
+                   packed ? packed : "?");
+            sfmt = av_get_packed_sample_fmt(sfmt);
+            n_channels = 1;
+        }
+
+        if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
            goto end;
+
        printf("Play the output audio file with the command:\n"
               "ffplay -f %s -ac %d -ar %d %s\n",
-               fmt, audio_dec_ctx->channels, audio_dec_ctx->sample_rate,
+               fmt, n_channels, audio_dec_ctx->sample_rate,
               audio_dst_filename);
    }

 end:
-    if (video_dec_ctx)
-        avcodec_close(video_dec_ctx);
-    if (audio_dec_ctx)
-        avcodec_close(audio_dec_ctx);
+    avcodec_close(video_dec_ctx);
+    avcodec_close(audio_dec_ctx);
    avformat_close_input(&fmt_ctx);
    if (video_dst_file)
        fclose(video_dst_file);
    if (audio_dst_file)
        fclose(audio_dst_file);
-    av_free(frame);
+    if (api_mode == API_MODE_OLD)
+        avcodec_free_frame(&frame);
+    else
+        av_frame_free(&frame);
    av_free(video_dst_data[0]);
-    av_free(audio_dst_data);

    return ret < 0;
 }
--- a/doc/examples/filter_audio.c
+++ b/doc/examples/filter_audio.c
@@ -0,0 +1,364 @@
+/*
+ * copyright (c) 2013 Andrew Kelley
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * libavfilter API usage example.
+ *
+ * @example filter_audio.c
+ * This example will generate a sine wave audio,
+ * pass it through a simple filter chain, and then compute the MD5 checksum of
+ * the output data.
+ *
+ * The filter chain it uses is:
+ * (input) -> abuffer -> volume -> aformat -> abuffersink -> (output)
+ *
+ * abuffer: This provides the endpoint where you can feed the decoded samples.
+ * volume: In this example we hardcode it to 0.90.
+ * aformat: This converts the samples to the samplefreq, channel layout,
+ *          and sample format required by the audio device.
+ * abuffersink: This provides the endpoint where you can read the samples after
+ *              they have passed through the filter chain.
+ */
+
+#include <inttypes.h>
+#include <math.h>
+#include <stdio.h>
+#include <stdlib.h>
+
+#include "libavutil/channel_layout.h"
+#include "libavutil/md5.h"
+#include "libavutil/opt.h"
+#include "libavutil/samplefmt.h"
+
+#include "libavfilter/avfilter.h"
+#include "libavfilter/buffersink.h"
+#include "libavfilter/buffersrc.h"
+
+#define INPUT_SAMPLERATE     48000
+#define INPUT_FORMAT         AV_SAMPLE_FMT_FLTP
+#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
+
+#define VOLUME_VAL 0.90
+
+static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
+                             AVFilterContext **sink)
+{
+    AVFilterGraph *filter_graph;
+    AVFilterContext *abuffer_ctx;
+    AVFilter        *abuffer;
+    AVFilterContext *volume_ctx;
+    AVFilter        *volume;
+    AVFilterContext *aformat_ctx;
+    AVFilter        *aformat;
+    AVFilterContext *abuffersink_ctx;
+    AVFilter        *abuffersink;
+
+    AVDictionary *options_dict = NULL;
+    uint8_t options_str[1024];
+    uint8_t ch_layout[64];
+
+    int err;
+
+    /* Create a new filtergraph, which will contain all the filters. */
+    filter_graph = avfilter_graph_alloc();
+    if (!filter_graph) {
+        fprintf(stderr, "Unable to create filter graph.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* Create the abuffer filter;
+     * it will be used for feeding the data into the graph. */
+    abuffer = avfilter_get_by_name("abuffer");
+    if (!abuffer) {
+        fprintf(stderr, "Could not find the abuffer filter.\n");
+        return AVERROR_FILTER_NOT_FOUND;
+    }
+
+    abuffer_ctx = avfilter_graph_alloc_filter(filter_graph, abuffer, "src");
+    if (!abuffer_ctx) {
+        fprintf(stderr, "Could not allocate the abuffer instance.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* Set the filter options through the AVOptions API. */
+    av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, INPUT_CHANNEL_LAYOUT);
+    av_opt_set    (abuffer_ctx, "channel_layout", ch_layout,                            AV_OPT_SEARCH_CHILDREN);
+    av_opt_set    (abuffer_ctx, "sample_fmt",     av_get_sample_fmt_name(INPUT_FORMAT), AV_OPT_SEARCH_CHILDREN);
+    av_opt_set_q  (abuffer_ctx, "time_base",      (AVRational){ 1, INPUT_SAMPLERATE },  AV_OPT_SEARCH_CHILDREN);
+    av_opt_set_int(abuffer_ctx, "sample_rate",    INPUT_SAMPLERATE,                     AV_OPT_SEARCH_CHILDREN);
+
+    /* Now initialize the filter; we pass NULL options, since we have already
+     * set all the options above. */
+    err = avfilter_init_str(abuffer_ctx, NULL);
+    if (err < 0) {
+        fprintf(stderr, "Could not initialize the abuffer filter.\n");
+        return err;
+    }
+
+    /* Create volume filter. */
+    volume = avfilter_get_by_name("volume");
+    if (!volume) {
+        fprintf(stderr, "Could not find the volume filter.\n");
+        return AVERROR_FILTER_NOT_FOUND;
+    }
+
+    volume_ctx = avfilter_graph_alloc_filter(filter_graph, volume, "volume");
+    if (!volume_ctx) {
+        fprintf(stderr, "Could not allocate the volume instance.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* A different way of passing the options is as key/value pairs in a
+     * dictionary. */
+    av_dict_set(&options_dict, "volume", AV_STRINGIFY(VOLUME_VAL), 0);
+    err = avfilter_init_dict(volume_ctx, &options_dict);
+    av_dict_free(&options_dict);
+    if (err < 0) {
+        fprintf(stderr, "Could not initialize the volume filter.\n");
+        return err;
+    }
+
+    /* Create the aformat filter;
+     * it ensures that the output is of the format we want. */
+    aformat = avfilter_get_by_name("aformat");
+    if (!aformat) {
+        fprintf(stderr, "Could not find the aformat filter.\n");
+        return AVERROR_FILTER_NOT_FOUND;
+    }
+
+    aformat_ctx = avfilter_graph_alloc_filter(filter_graph, aformat, "aformat");
+    if (!aformat_ctx) {
+        fprintf(stderr, "Could not allocate the aformat instance.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* A third way of passing the options is in a string of the form
+     * key1=value1:key2=value2.... */
+    snprintf(options_str, sizeof(options_str),
+             "sample_fmts=%s:sample_rates=%d:channel_layouts=0x%"PRIx64,
+             av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100,
+             (uint64_t)AV_CH_LAYOUT_STEREO);
+    err = avfilter_init_str(aformat_ctx, options_str);
+    if (err < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Could not initialize the aformat filter.\n");
+        return err;
+    }
+
+    /* Finally create the abuffersink filter;
+     * it will be used to get the filtered data out of the graph. */
+    abuffersink = avfilter_get_by_name("abuffersink");
+    if (!abuffersink) {
+        fprintf(stderr, "Could not find the abuffersink filter.\n");
+        return AVERROR_FILTER_NOT_FOUND;
+    }
+
+    abuffersink_ctx = avfilter_graph_alloc_filter(filter_graph, abuffersink, "sink");
+    if (!abuffersink_ctx) {
+        fprintf(stderr, "Could not allocate the abuffersink instance.\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /* This filter takes no options. */
+    err = avfilter_init_str(abuffersink_ctx, NULL);
+    if (err < 0) {
+        fprintf(stderr, "Could not initialize the abuffersink instance.\n");
+        return err;
+    }
+
+    /* Connect the filters;
+     * in this simple case the filters just form a linear chain. */
+    err = avfilter_link(abuffer_ctx, 0, volume_ctx, 0);
+    if (err >= 0)
+        err = avfilter_link(volume_ctx, 0, aformat_ctx, 0);
+    if (err >= 0)
+        err = avfilter_link(aformat_ctx, 0, abuffersink_ctx, 0);
+    if (err < 0) {
+        fprintf(stderr, "Error connecting filters\n");
+        return err;
+    }
+
+    /* Configure the graph. */
+    err = avfilter_graph_config(filter_graph, NULL);
+    if (err < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Error configuring the filter graph\n");
+        return err;
+    }
+
+    *graph = filter_graph;
+    *src   = abuffer_ctx;
+    *sink  = abuffersink_ctx;
+
+    return 0;
+}
+
+/* Do something useful with the filtered data: this simple
+ * example just prints the MD5 checksum of each plane to stdout. */
+static int process_output(struct AVMD5 *md5, AVFrame *frame)
+{
+    int planar     = av_sample_fmt_is_planar(frame->format);
+    int channels   = av_get_channel_layout_nb_channels(frame->channel_layout);
+    int planes     = planar ? channels : 1;
+    int bps        = av_get_bytes_per_sample(frame->format);
+    int plane_size = bps * frame->nb_samples * (planar ? 1 : channels);
+    int i, j;
+
+    for (i = 0; i < planes; i++) {
+        uint8_t checksum[16];
+
+        av_md5_init(md5);
+        av_md5_sum(checksum, frame->extended_data[i], plane_size);
+
+        fprintf(stdout, "plane %d: 0x", i);
+        for (j = 0; j < sizeof(checksum); j++)
+            fprintf(stdout, "%02X", checksum[j]);
+        fprintf(stdout, "\n");
+    }
+    fprintf(stdout, "\n");
+
+    return 0;
+}
+
+/* Construct a frame of audio data to be filtered;
+ * this simple example just synthesizes a sine wave. */
+static int get_input(AVFrame *frame, int frame_num)
+{
+    int err, i, j;
+
+#define FRAME_SIZE 1024
+
+    /* Set up the frame properties and allocate the buffer for the data. */
+    frame->sample_rate    = INPUT_SAMPLERATE;
+    frame->format         = INPUT_FORMAT;
+    frame->channel_layout = INPUT_CHANNEL_LAYOUT;
+    frame->nb_samples     = FRAME_SIZE;
+    frame->pts            = frame_num * FRAME_SIZE;
+
+    err = av_frame_get_buffer(frame, 0);
+    if (err < 0)
+        return err;
+
+    /* Fill the data for each channel. */
+    for (i = 0; i < 5; i++) {
+        float *data = (float*)frame->extended_data[i];
+
+        for (j = 0; j < frame->nb_samples; j++)
+            data[j] = sin(2 * M_PI * (frame_num + j) * (i + 1) / FRAME_SIZE);
+    }
+
+    return 0;
+}
+
+int main(int argc, char *argv[])
+{
+    struct AVMD5 *md5;
+    AVFilterGraph *graph;
+    AVFilterContext *src, *sink;
+    AVFrame *frame;
+    uint8_t errstr[1024];
+    float duration;
+    int err, nb_frames, i;
+
+    if (argc < 2) {
+        fprintf(stderr, "Usage: %s <duration>\n", argv[0]);
+        return 1;
+    }
+
+    duration  = atof(argv[1]);
+    nb_frames = duration * INPUT_SAMPLERATE / FRAME_SIZE;
+    if (nb_frames <= 0) {
+        fprintf(stderr, "Invalid duration: %s\n", argv[1]);
+        return 1;
+    }
+
+    avfilter_register_all();
+
+    /* Allocate the frame we will be using to store the data. */
+    frame  = av_frame_alloc();
+    if (!frame) {
+        fprintf(stderr, "Error allocating the frame\n");
+        return 1;
+    }
+
+    md5 = av_md5_alloc();
+    if (!md5) {
+        fprintf(stderr, "Error allocating the MD5 context\n");
+        return 1;
+    }
+
+    /* Set up the filtergraph. */
+    err = init_filter_graph(&graph, &src, &sink);
+    if (err < 0) {
+        fprintf(stderr, "Unable to init filter graph:");
+        goto fail;
+    }
+
+    /* the main filtering loop */
+    for (i = 0; i < nb_frames; i++) {
+        /* get an input frame to be filtered */
+        err = get_input(frame, i);
+        if (err < 0) {
+            fprintf(stderr, "Error generating input frame:");
+            goto fail;
+        }
+
+        /* Send the frame to the input of the filtergraph. */
+        err = av_buffersrc_add_frame(src, frame);
+        if (err < 0) {
+            av_frame_unref(frame);
+            fprintf(stderr, "Error submitting the frame to the filtergraph:");
+            goto fail;
+        }
+
+        /* Get all the filtered output that is available. */
+        while ((err = av_buffersink_get_frame(sink, frame)) >= 0) {
+            /* now do something with our filtered frame */
+            err = process_output(md5, frame);
+            if (err < 0) {
+                fprintf(stderr, "Error processing the filtered frame:");
+                goto fail;
+            }
+            av_frame_unref(frame);
+        }
+
+        if (err == AVERROR(EAGAIN)) {
+            /* Need to feed more frames in. */
+            continue;
+        } else if (err == AVERROR_EOF) {
+            /* Nothing more to do, finish. */
+            break;
+        } else if (err < 0) {
+            /* An error occurred. */
+            fprintf(stderr, "Error filtering the data:");
+            goto fail;
+        }
+    }
+
+    avfilter_graph_free(&graph);
+    av_frame_free(&frame);
+    av_freep(&md5);
+
+    return 0;
+
+fail:
+    av_strerror(err, errstr, sizeof(errstr));
+    fprintf(stderr, "%s\n", errstr);
+    return 1;
+}
--- a/doc/examples/filtering_audio.c
+++ b/doc/examples/filtering_audio.c
@@ -25,7 +25,7 @@
 /**
 * @file
 * API example for audio decoding and filtering
- * @example doc/examples/filtering_audio.c
+ * @example filtering_audio.c
 */

 #include <unistd.h>
@@ -38,8 +38,8 @@
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>

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

 static AVFormatContext *fmt_ctx;
 static AVCodecContext *dec_ctx;
@@ -85,18 +85,22 @@ static int open_input_file(const char *filename)
 static int init_filters(const char *filters_descr)
 {
    char args[512];
-    int ret;
+    int ret = 0;
    AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
    AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
    AVFilterInOut *outputs = avfilter_inout_alloc();
    AVFilterInOut *inputs  = avfilter_inout_alloc();
-    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 };
+    static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
+    static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
+    static const int out_sample_rates[] = { 8000, -1 };
    const AVFilterLink *outlink;
    AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;

    filter_graph = avfilter_graph_alloc();
+    if (!outputs || !inputs || !filter_graph) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }

    /* buffer audio source: the decoded frames from the decoder will be inserted here. */
    if (!dec_ctx->channel_layout)
@@ -109,7 +113,7 @@ static int init_filters(const char *filters_descr)
                                       args, NULL, filter_graph);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
-        return ret;
+        goto end;
    }

    /* buffer audio sink: to terminate the filter chain. */
@@ -117,28 +121,28 @@ static int init_filters(const char *filters_descr)
                                       NULL, NULL, filter_graph);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
-        return ret;
+        goto end;
    }

    ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
                              AV_OPT_SEARCH_CHILDREN);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
-        return ret;
+        goto end;
    }

    ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
                              AV_OPT_SEARCH_CHILDREN);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
-        return ret;
+        goto end;
    }

    ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
                              AV_OPT_SEARCH_CHILDREN);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
-        return ret;
+        goto end;
    }

    /* Endpoints for the filter graph. */
@@ -153,11 +157,11 @@ static int init_filters(const char *filters_descr)
    inputs->next       = NULL;

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

    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        return ret;
+        goto end;

    /* Print summary of the sink buffer
     * Note: args buffer is reused to store channel layout string */
@@ -168,7 +172,11 @@ static int init_filters(const char *filters_descr)
           (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
           args);

-    return 0;
+end:
+    avfilter_inout_free(&inputs);
+    avfilter_inout_free(&outputs);
+
+    return ret;
 }

 static void print_frame(const AVFrame *frame)
@@ -188,7 +196,7 @@ static void print_frame(const AVFrame *frame)
 int main(int argc, char **argv)
 {
    int ret;
-    AVPacket packet;
+    AVPacket packet0, packet;
    AVFrame *frame = av_frame_alloc();
    AVFrame *filt_frame = av_frame_alloc();
    int got_frame;
@@ -202,7 +210,6 @@ int main(int argc, char **argv)
        exit(1);
    }

-    avcodec_register_all();
    av_register_all();
    avfilter_register_all();

@@ -212,18 +219,24 @@ int main(int argc, char **argv)
        goto end;

    /* read all packets */
+    packet0.data = NULL;
+    packet.data = NULL;
    while (1) {
-        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
-            break;
+        if (!packet0.data) {
+            if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
+                break;
+            packet0 = packet;
+        }

        if (packet.stream_index == audio_stream_index) {
-            avcodec_get_frame_defaults(frame);
            got_frame = 0;
            ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
                continue;
            }
+            packet.size -= ret;
+            packet.data += ret;

            if (got_frame) {
                /* push the audio data from decoded frame into the filtergraph */
@@ -235,29 +248,31 @@ int main(int argc, char **argv)
                /* pull filtered audio from the filtergraph */
                while (1) {
                    ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
-                    if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                        break;
-                    if(ret < 0)
+                    if (ret < 0)
                        goto end;
                    print_frame(filt_frame);
                    av_frame_unref(filt_frame);
                }
            }
+
+            if (packet.size <= 0)
+                av_free_packet(&packet0);
+        } else {
+            /* discard non-wanted packets */
+            av_free_packet(&packet0);
        }
-        av_free_packet(&packet);
    }
 end:
    avfilter_graph_free(&filter_graph);
-    if (dec_ctx)
-        avcodec_close(dec_ctx);
+    avcodec_close(dec_ctx);
    avformat_close_input(&fmt_ctx);
    av_frame_free(&frame);
    av_frame_free(&filt_frame);

    if (ret < 0 && ret != AVERROR_EOF) {
-        char buf[1024];
-        av_strerror(ret, buf, sizeof(buf));
-        fprintf(stderr, "Error occurred: %s\n", buf);
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
        exit(1);
    }

--- a/doc/examples/filtering_video.c
+++ b/doc/examples/filtering_video.c
@@ -24,7 +24,7 @@
 /**
 * @file
 * API example for decoding and filtering
- * @example doc/examples/filtering_video.c
+ * @example filtering_video.c
 */

 #define _XOPEN_SOURCE 600 /* for usleep */
@@ -36,6 +36,7 @@
 #include <libavfilter/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
+#include <libavutil/opt.h>

 const char *filter_descr = "scale=78:24";

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

    /* init the video decoder */
    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -83,15 +85,18 @@ static int open_input_file(const char *filename)
 static int init_filters(const char *filters_descr)
 {
    char args[512];
-    int ret;
+    int ret = 0;
    AVFilter *buffersrc  = avfilter_get_by_name("buffer");
    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 };
-    AVBufferSinkParams *buffersink_params;

    filter_graph = avfilter_graph_alloc();
+    if (!outputs || !inputs || !filter_graph) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }

    /* buffer video source: the decoded frames from the decoder will be inserted here. */
    snprintf(args, sizeof(args),
@@ -104,18 +109,22 @@ static int init_filters(const char *filters_descr)
                                       args, NULL, filter_graph);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
-        return ret;
+        goto end;
    }

    /* buffer video sink: to terminate the filter chain. */
-    buffersink_params = av_buffersink_params_alloc();
-    buffersink_params->pixel_fmts = pix_fmts;
    ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
-                                       NULL, buffersink_params, filter_graph);
-    av_free(buffersink_params);
+                                       NULL, NULL, filter_graph);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
-        return ret;
+        goto end;
+    }
+
+    ret = av_opt_set_int_list(buffersink_ctx, "pix_fmts", pix_fmts,
+                              AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
+        goto end;
    }

    /* Endpoints for the filter graph. */
@@ -131,11 +140,16 @@ static int init_filters(const char *filters_descr)

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

    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        return ret;
-    return 0;
+        goto end;
+
+end:
+    avfilter_inout_free(&inputs);
+    avfilter_inout_free(&outputs);
+
+    return ret;
 }

 static void display_frame(const AVFrame *frame, AVRational time_base)
@@ -186,7 +200,6 @@ int main(int argc, char **argv)
        exit(1);
    }

-    avcodec_register_all();
    av_register_all();
    avfilter_register_all();

@@ -201,7 +214,6 @@ int main(int argc, char **argv)
            break;

        if (packet.stream_index == video_stream_index) {
-            avcodec_get_frame_defaults(frame);
            got_frame = 0;
            ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet);
            if (ret < 0) {
@@ -228,22 +240,20 @@ int main(int argc, char **argv)
                    display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
                    av_frame_unref(filt_frame);
                }
+                av_frame_unref(frame);
            }
        }
        av_free_packet(&packet);
    }
 end:
    avfilter_graph_free(&filter_graph);
-    if (dec_ctx)
-        avcodec_close(dec_ctx);
+    avcodec_close(dec_ctx);
    avformat_close_input(&fmt_ctx);
    av_frame_free(&frame);
    av_frame_free(&filt_frame);

    if (ret < 0 && ret != AVERROR_EOF) {
-        char buf[1024];
-        av_strerror(ret, buf, sizeof(buf));
-        fprintf(stderr, "Error occurred: %s\n", buf);
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
        exit(1);
    }

--- a/doc/examples/metadata.c
+++ b/doc/examples/metadata.c
@@ -23,7 +23,7 @@
 /**
 * @file
 * Shows how the metadata API can be used in application programs.
- * @example doc/examples/metadata.c
+ * @example metadata.c
 */

 #include <stdio.h>
--- a/doc/examples/muxing.c
+++ b/doc/examples/muxing.c
@@ -24,9 +24,9 @@
 * @file
 * libavformat API example.
 *
- * Output a media file in any supported libavformat format.
- * The default codecs are used.
- * @example doc/examples/muxing.c
+ * Output a media file in any supported libavformat format. The default
+ * codecs are used.
+ * @example muxing.c
 */

 #include <stdlib.h>
@@ -36,18 +36,43 @@

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

-/* 5 seconds stream duration */
-#define STREAM_DURATION   200.0
+static int audio_is_eof, video_is_eof;
+
+#define STREAM_DURATION   10.0
 #define STREAM_FRAME_RATE 25 /* 25 images/s */
-#define STREAM_NB_FRAMES  ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
 #define STREAM_PIX_FMT    AV_PIX_FMT_YUV420P /* default pix_fmt */

 static int sws_flags = SWS_BICUBIC;

+static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
+{
+    AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
+
+    printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
+           av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
+           av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
+           av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
+           pkt->stream_index);
+}
+
+static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
+{
+    /* rescale output packet timestamp values from codec to stream timebase */
+    pkt->pts = av_rescale_q_rnd(pkt->pts, *time_base, st->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
+    pkt->dts = av_rescale_q_rnd(pkt->dts, *time_base, st->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
+    pkt->duration = av_rescale_q(pkt->duration, *time_base, st->time_base);
+    pkt->stream_index = st->index;
+
+    /* Write the compressed frame to the media file. */
+    log_packet(fmt_ctx, pkt);
+    return av_interleaved_write_frame(fmt_ctx, pkt);
+}
+
 /* Add an output stream. */
 static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
                            enum AVCodecID codec_id)
@@ -73,7 +98,8 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,

    switch ((*codec)->type) {
    case AVMEDIA_TYPE_AUDIO:
-        c->sample_fmt  = AV_SAMPLE_FMT_FLTP;
+        c->sample_fmt  = (*codec)->sample_fmts ?
+            (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
        c->bit_rate    = 64000;
        c->sample_rate = 44100;
        c->channels    = 2;
@@ -122,6 +148,7 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,

 static float t, tincr, tincr2;

+AVFrame *audio_frame;
 static uint8_t **src_samples_data;
 static int       src_samples_linesize;
 static int       src_nb_samples;
@@ -130,6 +157,7 @@ static int max_dst_nb_samples;
 uint8_t **dst_samples_data;
 int       dst_samples_linesize;
 int       dst_samples_size;
+int samples_count;

 struct SwrContext *swr_ctx = NULL;

@@ -140,6 +168,13 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)

    c = st->codec;

+    /* allocate and init a re-usable frame */
+    audio_frame = av_frame_alloc();
+    if (!audio_frame) {
+        fprintf(stderr, "Could not allocate audio frame\n");
+        exit(1);
+    }
+
    /* open it */
    ret = avcodec_open2(c, codec, NULL);
    if (ret < 0) {
@@ -157,12 +192,17 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
        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);
+                                             src_nb_samples, AV_SAMPLE_FMT_S16, 0);
    if (ret < 0) {
        fprintf(stderr, "Could not allocate source samples\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;
+
    /* create resampler context */
    if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
        swr_ctx = swr_alloc();
@@ -184,17 +224,15 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
            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);
+        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);
+        }
+    } else {
+        dst_samples_data = src_samples_data;
    }
    dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples,
                                                  c->sample_fmt, 0);
@@ -217,77 +255,83 @@ static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
    }
 }

-static void write_audio_frame(AVFormatContext *oc, AVStream *st)
+static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush)
 {
    AVCodecContext *c;
    AVPacket pkt = { 0 }; // data and size must be 0;
-    AVFrame *frame = avcodec_alloc_frame();
    int got_packet, ret, dst_nb_samples;

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

-    get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
+    if (!flush) {
+        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)
+        /* 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);
-            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);
+            }
+        } else {
+            dst_nb_samples = src_nb_samples;
        }

-        /* 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;
+        audio_frame->nb_samples = dst_nb_samples;
+        audio_frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
+        avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt,
+                                 dst_samples_data[0], dst_samples_size, 0);
+        samples_count += dst_nb_samples;
    }

-    frame->nb_samples = dst_nb_samples;
-    avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
-                             dst_samples_data[0], dst_samples_size, 0);
-
-    ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
+    ret = avcodec_encode_audio2(c, &pkt, flush ? NULL : audio_frame, &got_packet);
    if (ret < 0) {
        fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
        exit(1);
    }

-    if (!got_packet)
+    if (!got_packet) {
+        if (flush)
+            audio_is_eof = 1;
        return;
+    }

-    pkt.stream_index = st->index;
-
-    /* Write the compressed frame to the media file. */
-    ret = av_interleaved_write_frame(oc, &pkt);
-    if (ret != 0) {
+    ret = write_frame(oc, &c->time_base, st, &pkt);
+    if (ret < 0) {
        fprintf(stderr, "Error while writing audio frame: %s\n",
                av_err2str(ret));
        exit(1);
    }
-    avcodec_free_frame(&frame);
 }

 static void close_audio(AVFormatContext *oc, AVStream *st)
 {
    avcodec_close(st->codec);
+    if (dst_samples_data != src_samples_data) {
+        av_free(dst_samples_data[0]);
+        av_free(dst_samples_data);
+    }
    av_free(src_samples_data[0]);
-    av_free(dst_samples_data[0]);
+    av_free(src_samples_data);
+    av_frame_free(&audio_frame);
 }

 /**************************************************************/
@@ -310,11 +354,14 @@ static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
    }

    /* allocate and init a re-usable frame */
-    frame = avcodec_alloc_frame();
+    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate video frame\n");
        exit(1);
    }
+    frame->format = c->pix_fmt;
+    frame->width = c->width;
+    frame->height = c->height;

    /* Allocate the encoded raw picture. */
    ret = avpicture_alloc(&dst_picture, c->pix_fmt, c->width, c->height);
@@ -361,17 +408,13 @@ static void fill_yuv_image(AVPicture *pict, int frame_index,
    }
 }

-static void write_video_frame(AVFormatContext *oc, AVStream *st)
+static void write_video_frame(AVFormatContext *oc, AVStream *st, int flush)
 {
    int ret;
    static struct SwsContext *sws_ctx;
    AVCodecContext *c = st->codec;

-    if (frame_count >= STREAM_NB_FRAMES) {
-        /* No more frames to compress. The codec has a latency of a few
-         * frames if using B-frames, so we get the last frames by
-         * passing the same picture again. */
-    } else {
+    if (!flush) {
        if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
            /* as we only generate a YUV420P picture, we must convert it
             * to the codec pixel format if needed */
@@ -394,7 +437,7 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)
        }
    }

-    if (oc->oformat->flags & AVFMT_RAWPICTURE) {
+    if (oc->oformat->flags & AVFMT_RAWPICTURE && !flush) {
        /* Raw video case - directly store the picture in the packet */
        AVPacket pkt;
        av_init_packet(&pkt);
@@ -411,23 +454,24 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)
        av_init_packet(&pkt);

        /* encode the image */
-        ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
+        frame->pts = frame_count;
+        ret = avcodec_encode_video2(c, &pkt, flush ? NULL : 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 (!ret && got_packet && pkt.size) {
-            pkt.stream_index = st->index;
-
-            /* Write the compressed frame to the media file. */
-            ret = av_interleaved_write_frame(oc, &pkt);
+        if (got_packet) {
+            ret = write_frame(oc, &c->time_base, st, &pkt);
        } else {
+            if (flush)
+                video_is_eof = 1;
            ret = 0;
        }
    }
-    if (ret != 0) {
+
+    if (ret < 0) {
        fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
        exit(1);
    }
@@ -439,7 +483,7 @@ static void close_video(AVFormatContext *oc, AVStream *st)
    avcodec_close(st->codec);
    av_free(src_picture.data[0]);
    av_free(dst_picture.data[0]);
-    av_free(frame);
+    av_frame_free(&frame);
 }

 /**************************************************************/
@@ -453,7 +497,7 @@ int main(int argc, char **argv)
    AVStream *audio_st, *video_st;
    AVCodec *audio_codec, *video_codec;
    double audio_time, video_time;
-    int ret;
+    int flush, ret;

    /* Initialize libavcodec, and register all codecs and formats. */
    av_register_all();
@@ -477,9 +521,9 @@ int main(int argc, char **argv)
        printf("Could not deduce output format from file extension: using MPEG.\n");
        avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
    }
-    if (!oc) {
+    if (!oc)
        return 1;
-    }
+
    fmt = oc->oformat;

    /* Add the audio and video streams using the default format codecs
@@ -487,12 +531,10 @@ int main(int argc, char **argv)
    video_st = NULL;
    audio_st = NULL;

-    if (fmt->video_codec != AV_CODEC_ID_NONE) {
+    if (fmt->video_codec != AV_CODEC_ID_NONE)
        video_st = add_stream(oc, &video_codec, fmt->video_codec);
-    }
-    if (fmt->audio_codec != AV_CODEC_ID_NONE) {
+    if (fmt->audio_codec != AV_CODEC_ID_NONE)
        audio_st = add_stream(oc, &audio_codec, fmt->audio_codec);
-    }

    /* Now that all the parameters are set, we can open the audio and
     * video codecs and allocate the necessary encode buffers. */
@@ -521,23 +563,23 @@ int main(int argc, char **argv)
        return 1;
    }

-    if (frame)
-        frame->pts = 0;
-    for (;;) {
+    flush = 0;
+    while ((video_st && !video_is_eof) || (audio_st && !audio_is_eof)) {
        /* Compute current audio and video time. */
-        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;
+        audio_time = (audio_st && !audio_is_eof) ? audio_st->pts.val * av_q2d(audio_st->time_base) : INFINITY;
+        video_time = (video_st && !video_is_eof) ? video_st->pts.val * av_q2d(video_st->time_base) : INFINITY;

-        if ((!audio_st || audio_time >= STREAM_DURATION) &&
-            (!video_st || video_time >= STREAM_DURATION))
-            break;
+        if (!flush &&
+            (!audio_st || audio_time >= STREAM_DURATION) &&
+            (!video_st || video_time >= STREAM_DURATION)) {
+            flush = 1;
+        }

        /* write interleaved audio and video frames */
-        if (!video_st || (video_st && audio_st && audio_time < video_time)) {
-            write_audio_frame(oc, audio_st);
-        } else {
-            write_video_frame(oc, video_st);
-            frame->pts += av_rescale_q(1, video_st->codec->time_base, video_st->time_base);
+        if (audio_st && !audio_is_eof && audio_time <= video_time) {
+            write_audio_frame(oc, audio_st, flush);
+        } else if (video_st && !video_is_eof && video_time < audio_time) {
+            write_video_frame(oc, video_st, flush);
        }
    }

--- a/doc/examples/remuxing.c
+++ b/doc/examples/remuxing.c
@@ -0,0 +1,164 @@
+/*
+ * Copyright (c) 2013 Stefano Sabatini
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libavformat/libavcodec demuxing and muxing API example.
+ *
+ * Remux streams from one container format to another.
+ * @example remuxing.c
+ */
+
+#include <libavutil/timestamp.h>
+#include <libavformat/avformat.h>
+
+static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, const char *tag)
+{
+    AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
+
+    printf("%s: pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
+           tag,
+           av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
+           av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
+           av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
+           pkt->stream_index);
+}
+
+int main(int argc, char **argv)
+{
+    AVOutputFormat *ofmt = NULL;
+    AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
+    AVPacket pkt;
+    const char *in_filename, *out_filename;
+    int ret, i;
+
+    if (argc < 3) {
+        printf("usage: %s input output\n"
+               "API example program to remux a media file with libavformat and libavcodec.\n"
+               "The output format is guessed according to the file extension.\n"
+               "\n", argv[0]);
+        return 1;
+    }
+
+    in_filename  = argv[1];
+    out_filename = argv[2];
+
+    av_register_all();
+
+    if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
+        fprintf(stderr, "Could not open input file '%s'", in_filename);
+        goto end;
+    }
+
+    if ((ret = avformat_find_stream_info(ifmt_ctx, 0)) < 0) {
+        fprintf(stderr, "Failed to retrieve input stream information");
+        goto end;
+    }
+
+    av_dump_format(ifmt_ctx, 0, in_filename, 0);
+
+    avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, out_filename);
+    if (!ofmt_ctx) {
+        fprintf(stderr, "Could not create output context\n");
+        ret = AVERROR_UNKNOWN;
+        goto end;
+    }
+
+    ofmt = ofmt_ctx->oformat;
+
+    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
+        AVStream *in_stream = ifmt_ctx->streams[i];
+        AVStream *out_stream = avformat_new_stream(ofmt_ctx, in_stream->codec->codec);
+        if (!out_stream) {
+            fprintf(stderr, "Failed allocating output stream\n");
+            ret = AVERROR_UNKNOWN;
+            goto end;
+        }
+
+        ret = avcodec_copy_context(out_stream->codec, in_stream->codec);
+        if (ret < 0) {
+            fprintf(stderr, "Failed to copy context from input to output stream codec context\n");
+            goto end;
+        }
+        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
+            out_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
+    }
+    av_dump_format(ofmt_ctx, 0, out_filename, 1);
+
+    if (!(ofmt->flags & AVFMT_NOFILE)) {
+        ret = avio_open(&ofmt_ctx->pb, out_filename, AVIO_FLAG_WRITE);
+        if (ret < 0) {
+            fprintf(stderr, "Could not open output file '%s'", out_filename);
+            goto end;
+        }
+    }
+
+    ret = avformat_write_header(ofmt_ctx, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Error occurred when opening output file\n");
+        goto end;
+    }
+
+    while (1) {
+        AVStream *in_stream, *out_stream;
+
+        ret = av_read_frame(ifmt_ctx, &pkt);
+        if (ret < 0)
+            break;
+
+        in_stream  = ifmt_ctx->streams[pkt.stream_index];
+        out_stream = ofmt_ctx->streams[pkt.stream_index];
+
+        log_packet(ifmt_ctx, &pkt, "in");
+
+        /* copy packet */
+        pkt.pts = av_rescale_q_rnd(pkt.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
+        pkt.dts = av_rescale_q_rnd(pkt.dts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
+        pkt.duration = av_rescale_q(pkt.duration, in_stream->time_base, out_stream->time_base);
+        pkt.pos = -1;
+        log_packet(ofmt_ctx, &pkt, "out");
+
+        ret = av_interleaved_write_frame(ofmt_ctx, &pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error muxing packet\n");
+            break;
+        }
+        av_free_packet(&pkt);
+    }
+
+    av_write_trailer(ofmt_ctx);
+end:
+
+    avformat_close_input(&ifmt_ctx);
+
+    /* close output */
+    if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
+        avio_close(ofmt_ctx->pb);
+    avformat_free_context(ofmt_ctx);
+
+    if (ret < 0 && ret != AVERROR_EOF) {
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        return 1;
+    }
+
+    return 0;
+}
--- a/doc/examples/resampling_audio.c
+++ b/doc/examples/resampling_audio.c
@@ -21,7 +21,7 @@
 */

 /**
- * @example doc/examples/resampling_audio.c
+ * @example resampling_audio.c
 * libswresample API use example.
 */

@@ -62,7 +62,7 @@ static int get_format_from_sample_fmt(const char **fmt,
 /**
 * Fill dst buffer with nb_samples, generated starting from t.
 */
-void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
+static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
 {
    int i, j;
    double tincr = 1.0 / sample_rate, *dstp = dst;
@@ -184,6 +184,10 @@ int main(int argc, char **argv)
        }
        dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
                                                 ret, dst_sample_fmt, 1);
+        if (dst_bufsize < 0) {
+            fprintf(stderr, "Could not get sample buffer size\n");
+            goto end;
+        }
        printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
        fwrite(dst_data[0], 1, dst_bufsize, dst_file);
    } while (t < 10);
--- a/doc/examples/scaling_video.c
+++ b/doc/examples/scaling_video.c
@@ -23,7 +23,7 @@
 /**
 * @file
 * libswscale API use example.
- * @example doc/examples/scaling_video.c
+ * @example scaling_video.c
 */

 #include <libavutil/imgutils.h>
--- a/doc/examples/transcode_aac.c
+++ b/doc/examples/transcode_aac.c
@@ -0,0 +1,755 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * simple audio converter
+ *
+ * @example transcode_aac.c
+ * Convert an input audio file to AAC in an MP4 container using FFmpeg.
+ * @author Andreas Unterweger (dustsigns@gmail.com)
+ */
+
+#include <stdio.h>
+
+#include "libavformat/avformat.h"
+#include "libavformat/avio.h"
+
+#include "libavcodec/avcodec.h"
+
+#include "libavutil/audio_fifo.h"
+#include "libavutil/avassert.h"
+#include "libavutil/avstring.h"
+#include "libavutil/frame.h"
+#include "libavutil/opt.h"
+
+#include "libswresample/swresample.h"
+
+/** The output bit rate in kbit/s */
+#define OUTPUT_BIT_RATE 48000
+/** The number of output channels */
+#define OUTPUT_CHANNELS 2
+/** The audio sample output format */
+#define OUTPUT_SAMPLE_FORMAT AV_SAMPLE_FMT_S16
+
+/**
+ * Convert an error code into a text message.
+ * @param error Error code to be converted
+ * @return Corresponding error text (not thread-safe)
+ */
+static char *const get_error_text(const int error)
+{
+    static char error_buffer[255];
+    av_strerror(error, error_buffer, sizeof(error_buffer));
+    return error_buffer;
+}
+
+/** Open an input file and the required decoder. */
+static int open_input_file(const char *filename,
+                           AVFormatContext **input_format_context,
+                           AVCodecContext **input_codec_context)
+{
+    AVCodec *input_codec;
+    int error;
+
+    /** Open the input file to read from it. */
+    if ((error = avformat_open_input(input_format_context, filename, NULL,
+                                     NULL)) < 0) {
+        fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
+                filename, get_error_text(error));
+        *input_format_context = NULL;
+        return error;
+    }
+
+    /** Get information on the input file (number of streams etc.). */
+    if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
+        fprintf(stderr, "Could not open find stream info (error '%s')\n",
+                get_error_text(error));
+        avformat_close_input(input_format_context);
+        return error;
+    }
+
+    /** Make sure that there is only one stream in the input file. */
+    if ((*input_format_context)->nb_streams != 1) {
+        fprintf(stderr, "Expected one audio input stream, but found %d\n",
+                (*input_format_context)->nb_streams);
+        avformat_close_input(input_format_context);
+        return AVERROR_EXIT;
+    }
+
+    /** Find a decoder for the audio stream. */
+    if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codec->codec_id))) {
+        fprintf(stderr, "Could not find input codec\n");
+        avformat_close_input(input_format_context);
+        return AVERROR_EXIT;
+    }
+
+    /** Open the decoder for the audio stream to use it later. */
+    if ((error = avcodec_open2((*input_format_context)->streams[0]->codec,
+                               input_codec, NULL)) < 0) {
+        fprintf(stderr, "Could not open input codec (error '%s')\n",
+                get_error_text(error));
+        avformat_close_input(input_format_context);
+        return error;
+    }
+
+    /** Save the decoder context for easier access later. */
+    *input_codec_context = (*input_format_context)->streams[0]->codec;
+
+    return 0;
+}
+
+/**
+ * Open an output file and the required encoder.
+ * Also set some basic encoder parameters.
+ * Some of these parameters are based on the input file's parameters.
+ */
+static int open_output_file(const char *filename,
+                            AVCodecContext *input_codec_context,
+                            AVFormatContext **output_format_context,
+                            AVCodecContext **output_codec_context)
+{
+    AVIOContext *output_io_context = NULL;
+    AVStream *stream               = NULL;
+    AVCodec *output_codec          = NULL;
+    int error;
+
+    /** Open the output file to write to it. */
+    if ((error = avio_open(&output_io_context, filename,
+                           AVIO_FLAG_WRITE)) < 0) {
+        fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
+                filename, get_error_text(error));
+        return error;
+    }
+
+    /** Create a new format context for the output container format. */
+    if (!(*output_format_context = avformat_alloc_context())) {
+        fprintf(stderr, "Could not allocate output format context\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /** Associate the output file (pointer) with the container format context. */
+    (*output_format_context)->pb = output_io_context;
+
+    /** Guess the desired container format based on the file extension. */
+    if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
+                                                              NULL))) {
+        fprintf(stderr, "Could not find output file format\n");
+        goto cleanup;
+    }
+
+    av_strlcpy((*output_format_context)->filename, filename,
+               sizeof((*output_format_context)->filename));
+
+    /** Find the encoder to be used by its name. */
+    if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
+        fprintf(stderr, "Could not find an AAC encoder.\n");
+        goto cleanup;
+    }
+
+    /** Create a new audio stream in the output file container. */
+    if (!(stream = avformat_new_stream(*output_format_context, output_codec))) {
+        fprintf(stderr, "Could not create new stream\n");
+        error = AVERROR(ENOMEM);
+        goto cleanup;
+    }
+
+    /** Save the encoder context for easiert access later. */
+    *output_codec_context = stream->codec;
+
+    /**
+     * Set the basic encoder parameters.
+     * The input file's sample rate is used to avoid a sample rate conversion.
+     */
+    (*output_codec_context)->channels       = OUTPUT_CHANNELS;
+    (*output_codec_context)->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
+    (*output_codec_context)->sample_rate    = input_codec_context->sample_rate;
+    (*output_codec_context)->sample_fmt     = AV_SAMPLE_FMT_S16;
+    (*output_codec_context)->bit_rate       = OUTPUT_BIT_RATE;
+
+    /**
+     * Some container formats (like MP4) require global headers to be present
+     * Mark the encoder so that it behaves accordingly.
+     */
+    if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
+        (*output_codec_context)->flags |= CODEC_FLAG_GLOBAL_HEADER;
+
+    /** Open the encoder for the audio stream to use it later. */
+    if ((error = avcodec_open2(*output_codec_context, output_codec, NULL)) < 0) {
+        fprintf(stderr, "Could not open output codec (error '%s')\n",
+                get_error_text(error));
+        goto cleanup;
+    }
+
+    return 0;
+
+cleanup:
+    avio_close((*output_format_context)->pb);
+    avformat_free_context(*output_format_context);
+    *output_format_context = NULL;
+    return error < 0 ? error : AVERROR_EXIT;
+}
+
+/** Initialize one data packet for reading or writing. */
+static void init_packet(AVPacket *packet)
+{
+    av_init_packet(packet);
+    /** Set the packet data and size so that it is recognized as being empty. */
+    packet->data = NULL;
+    packet->size = 0;
+}
+
+/** Initialize one audio frame for reading from the input file */
+static int init_input_frame(AVFrame **frame)
+{
+    if (!(*frame = av_frame_alloc())) {
+        fprintf(stderr, "Could not allocate input frame\n");
+        return AVERROR(ENOMEM);
+    }
+    return 0;
+}
+
+/**
+ * Initialize the audio resampler based on the input and output codec settings.
+ * If the input and output sample formats differ, a conversion is required
+ * libswresample takes care of this, but requires initialization.
+ */
+static int init_resampler(AVCodecContext *input_codec_context,
+                          AVCodecContext *output_codec_context,
+                          SwrContext **resample_context)
+{
+        int error;
+
+        /**
+         * Create a resampler context for the conversion.
+         * Set the conversion parameters.
+         * Default channel layouts based on the number of channels
+         * are assumed for simplicity (they are sometimes not detected
+         * properly by the demuxer and/or decoder).
+         */
+        *resample_context = swr_alloc_set_opts(NULL,
+                                              av_get_default_channel_layout(output_codec_context->channels),
+                                              output_codec_context->sample_fmt,
+                                              output_codec_context->sample_rate,
+                                              av_get_default_channel_layout(input_codec_context->channels),
+                                              input_codec_context->sample_fmt,
+                                              input_codec_context->sample_rate,
+                                              0, NULL);
+        if (!*resample_context) {
+            fprintf(stderr, "Could not allocate resample context\n");
+            return AVERROR(ENOMEM);
+        }
+        /**
+        * Perform a sanity check so that the number of converted samples is
+        * not greater than the number of samples to be converted.
+        * If the sample rates differ, this case has to be handled differently
+        */
+        av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
+
+        /** Open the resampler with the specified parameters. */
+        if ((error = swr_init(*resample_context)) < 0) {
+            fprintf(stderr, "Could not open resample context\n");
+            swr_free(resample_context);
+            return error;
+        }
+    return 0;
+}
+
+/** Initialize a FIFO buffer for the audio samples to be encoded. */
+static int init_fifo(AVAudioFifo **fifo)
+{
+    /** Create the FIFO buffer based on the specified output sample format. */
+    if (!(*fifo = av_audio_fifo_alloc(OUTPUT_SAMPLE_FORMAT, OUTPUT_CHANNELS, 1))) {
+        fprintf(stderr, "Could not allocate FIFO\n");
+        return AVERROR(ENOMEM);
+    }
+    return 0;
+}
+
+/** Write the header of the output file container. */
+static int write_output_file_header(AVFormatContext *output_format_context)
+{
+    int error;
+    if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
+        fprintf(stderr, "Could not write output file header (error '%s')\n",
+                get_error_text(error));
+        return error;
+    }
+    return 0;
+}
+
+/** Decode one audio frame from the input file. */
+static int decode_audio_frame(AVFrame *frame,
+                              AVFormatContext *input_format_context,
+                              AVCodecContext *input_codec_context,
+                              int *data_present, int *finished)
+{
+    /** Packet used for temporary storage. */
+    AVPacket input_packet;
+    int error;
+    init_packet(&input_packet);
+
+    /** Read one audio frame from the input file into a temporary packet. */
+    if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
+        /** If we are the the end of the file, flush the decoder below. */
+        if (error == AVERROR_EOF)
+            *finished = 1;
+        else {
+            fprintf(stderr, "Could not read frame (error '%s')\n",
+                    get_error_text(error));
+            return error;
+        }
+    }
+
+    /**
+     * Decode the audio frame stored in the temporary packet.
+     * The input audio stream decoder is used to do this.
+     * If we are at the end of the file, pass an empty packet to the decoder
+     * to flush it.
+     */
+    if ((error = avcodec_decode_audio4(input_codec_context, frame,
+                                       data_present, &input_packet)) < 0) {
+        fprintf(stderr, "Could not decode frame (error '%s')\n",
+                get_error_text(error));
+        av_free_packet(&input_packet);
+        return error;
+    }
+
+    /**
+     * If the decoder has not been flushed completely, we are not finished,
+     * so that this function has to be called again.
+     */
+    if (*finished && *data_present)
+        *finished = 0;
+    av_free_packet(&input_packet);
+    return 0;
+}
+
+/**
+ * Initialize a temporary storage for the specified number of audio samples.
+ * The conversion requires temporary storage due to the different format.
+ * The number of audio samples to be allocated is specified in frame_size.
+ */
+static int init_converted_samples(uint8_t ***converted_input_samples,
+                                  AVCodecContext *output_codec_context,
+                                  int frame_size)
+{
+    int error;
+
+    /**
+     * Allocate as many pointers as there are audio channels.
+     * Each pointer will later point to the audio samples of the corresponding
+     * channels (although it may be NULL for interleaved formats).
+     */
+    if (!(*converted_input_samples = calloc(output_codec_context->channels,
+                                            sizeof(**converted_input_samples)))) {
+        fprintf(stderr, "Could not allocate converted input sample pointers\n");
+        return AVERROR(ENOMEM);
+    }
+
+    /**
+     * Allocate memory for the samples of all channels in one consecutive
+     * block for convenience.
+     */
+    if ((error = av_samples_alloc(*converted_input_samples, NULL,
+                                  output_codec_context->channels,
+                                  frame_size,
+                                  output_codec_context->sample_fmt, 0)) < 0) {
+        fprintf(stderr,
+                "Could not allocate converted input samples (error '%s')\n",
+                get_error_text(error));
+        av_freep(&(*converted_input_samples)[0]);
+        free(*converted_input_samples);
+        return error;
+    }
+    return 0;
+}
+
+/**
+ * Convert the input audio samples into the output sample format.
+ * The conversion happens on a per-frame basis, the size of which is specified
+ * by frame_size.
+ */
+static int convert_samples(const uint8_t **input_data,
+                           uint8_t **converted_data, const int frame_size,
+                           SwrContext *resample_context)
+{
+    int error;
+
+    /** Convert the samples using the resampler. */
+    if ((error = swr_convert(resample_context,
+                             converted_data, frame_size,
+                             input_data    , frame_size)) < 0) {
+        fprintf(stderr, "Could not convert input samples (error '%s')\n",
+                get_error_text(error));
+        return error;
+    }
+
+    return 0;
+}
+
+/** Add converted input audio samples to the FIFO buffer for later processing. */
+static int add_samples_to_fifo(AVAudioFifo *fifo,
+                               uint8_t **converted_input_samples,
+                               const int frame_size)
+{
+    int error;
+
+    /**
+     * Make the FIFO as large as it needs to be to hold both,
+     * the old and the new samples.
+     */
+    if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
+        fprintf(stderr, "Could not reallocate FIFO\n");
+        return error;
+    }
+
+    /** Store the new samples in the FIFO buffer. */
+    if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
+                            frame_size) < frame_size) {
+        fprintf(stderr, "Could not write data to FIFO\n");
+        return AVERROR_EXIT;
+    }
+    return 0;
+}
+
+/**
+ * Read one audio frame from the input file, decodes, converts and stores
+ * it in the FIFO buffer.
+ */
+static int read_decode_convert_and_store(AVAudioFifo *fifo,
+                                         AVFormatContext *input_format_context,
+                                         AVCodecContext *input_codec_context,
+                                         AVCodecContext *output_codec_context,
+                                         SwrContext *resampler_context,
+                                         int *finished)
+{
+    /** Temporary storage of the input samples of the frame read from the file. */
+    AVFrame *input_frame = NULL;
+    /** Temporary storage for the converted input samples. */
+    uint8_t **converted_input_samples = NULL;
+    int data_present;
+    int ret = AVERROR_EXIT;
+
+    /** Initialize temporary storage for one input frame. */
+    if (init_input_frame(&input_frame))
+        goto cleanup;
+    /** Decode one frame worth of audio samples. */
+    if (decode_audio_frame(input_frame, input_format_context,
+                           input_codec_context, &data_present, finished))
+        goto cleanup;
+    /**
+     * If we are at the end of the file and there are no more samples
+     * in the decoder which are delayed, we are actually finished.
+     * This must not be treated as an error.
+     */
+    if (*finished && !data_present) {
+        ret = 0;
+        goto cleanup;
+    }
+    /** If there is decoded data, convert and store it */
+    if (data_present) {
+        /** Initialize the temporary storage for the converted input samples. */
+        if (init_converted_samples(&converted_input_samples, output_codec_context,
+                                   input_frame->nb_samples))
+            goto cleanup;
+
+        /**
+         * Convert the input samples to the desired output sample format.
+         * This requires a temporary storage provided by converted_input_samples.
+         */
+        if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
+                            input_frame->nb_samples, resampler_context))
+            goto cleanup;
+
+        /** Add the converted input samples to the FIFO buffer for later processing. */
+        if (add_samples_to_fifo(fifo, converted_input_samples,
+                                input_frame->nb_samples))
+            goto cleanup;
+        ret = 0;
+    }
+    ret = 0;
+
+cleanup:
+    if (converted_input_samples) {
+        av_freep(&converted_input_samples[0]);
+        free(converted_input_samples);
+    }
+    av_frame_free(&input_frame);
+
+    return ret;
+}
+
+/**
+ * Initialize one input frame for writing to the output file.
+ * The frame will be exactly frame_size samples large.
+ */
+static int init_output_frame(AVFrame **frame,
+                             AVCodecContext *output_codec_context,
+                             int frame_size)
+{
+    int error;
+
+    /** Create a new frame to store the audio samples. */
+    if (!(*frame = av_frame_alloc())) {
+        fprintf(stderr, "Could not allocate output frame\n");
+        return AVERROR_EXIT;
+    }
+
+    /**
+     * Set the frame's parameters, especially its size and format.
+     * av_frame_get_buffer needs this to allocate memory for the
+     * audio samples of the frame.
+     * Default channel layouts based on the number of channels
+     * are assumed for simplicity.
+     */
+    (*frame)->nb_samples     = frame_size;
+    (*frame)->channel_layout = output_codec_context->channel_layout;
+    (*frame)->format         = output_codec_context->sample_fmt;
+    (*frame)->sample_rate    = output_codec_context->sample_rate;
+
+    /**
+     * Allocate the samples of the created frame. This call will make
+     * sure that the audio frame can hold as many samples as specified.
+     */
+    if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
+        fprintf(stderr, "Could allocate output frame samples (error '%s')\n",
+                get_error_text(error));
+        av_frame_free(frame);
+        return error;
+    }
+
+    return 0;
+}
+
+/** Encode one frame worth of audio to the output file. */
+static int encode_audio_frame(AVFrame *frame,
+                              AVFormatContext *output_format_context,
+                              AVCodecContext *output_codec_context,
+                              int *data_present)
+{
+    /** Packet used for temporary storage. */
+    AVPacket output_packet;
+    int error;
+    init_packet(&output_packet);
+
+    /**
+     * Encode the audio frame and store it in the temporary packet.
+     * The output audio stream encoder is used to do this.
+     */
+    if ((error = avcodec_encode_audio2(output_codec_context, &output_packet,
+                                       frame, data_present)) < 0) {
+        fprintf(stderr, "Could not encode frame (error '%s')\n",
+                get_error_text(error));
+        av_free_packet(&output_packet);
+        return error;
+    }
+
+    /** Write one audio frame from the temporary packet to the output file. */
+    if (*data_present) {
+        if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
+            fprintf(stderr, "Could not write frame (error '%s')\n",
+                    get_error_text(error));
+            av_free_packet(&output_packet);
+            return error;
+        }
+
+        av_free_packet(&output_packet);
+    }
+
+    return 0;
+}
+
+/**
+ * Load one audio frame from the FIFO buffer, encode and write it to the
+ * output file.
+ */
+static int load_encode_and_write(AVAudioFifo *fifo,
+                                 AVFormatContext *output_format_context,
+                                 AVCodecContext *output_codec_context)
+{
+    /** Temporary storage of the output samples of the frame written to the file. */
+    AVFrame *output_frame;
+    /**
+     * Use the maximum number of possible samples per frame.
+     * If there is less than the maximum possible frame size in the FIFO
+     * buffer use this number. Otherwise, use the maximum possible frame size
+     */
+    const int frame_size = FFMIN(av_audio_fifo_size(fifo),
+                                 output_codec_context->frame_size);
+    int data_written;
+
+    /** Initialize temporary storage for one output frame. */
+    if (init_output_frame(&output_frame, output_codec_context, frame_size))
+        return AVERROR_EXIT;
+
+    /**
+     * Read as many samples from the FIFO buffer as required to fill the frame.
+     * The samples are stored in the frame temporarily.
+     */
+    if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
+        fprintf(stderr, "Could not read data from FIFO\n");
+        av_frame_free(&output_frame);
+        return AVERROR_EXIT;
+    }
+
+    /** Encode one frame worth of audio samples. */
+    if (encode_audio_frame(output_frame, output_format_context,
+                           output_codec_context, &data_written)) {
+        av_frame_free(&output_frame);
+        return AVERROR_EXIT;
+    }
+    av_frame_free(&output_frame);
+    return 0;
+}
+
+/** Write the trailer of the output file container. */
+static int write_output_file_trailer(AVFormatContext *output_format_context)
+{
+    int error;
+    if ((error = av_write_trailer(output_format_context)) < 0) {
+        fprintf(stderr, "Could not write output file trailer (error '%s')\n",
+                get_error_text(error));
+        return error;
+    }
+    return 0;
+}
+
+/** Convert an audio file to an AAC file in an MP4 container. */
+int main(int argc, char **argv)
+{
+    AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
+    AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
+    SwrContext *resample_context = NULL;
+    AVAudioFifo *fifo = NULL;
+    int ret = AVERROR_EXIT;
+
+    if (argc < 3) {
+        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
+        exit(1);
+    }
+
+    /** Register all codecs and formats so that they can be used. */
+    av_register_all();
+    /** Open the input file for reading. */
+    if (open_input_file(argv[1], &input_format_context,
+                        &input_codec_context))
+        goto cleanup;
+    /** Open the output file for writing. */
+    if (open_output_file(argv[2], input_codec_context,
+                         &output_format_context, &output_codec_context))
+        goto cleanup;
+    /** Initialize the resampler to be able to convert audio sample formats. */
+    if (init_resampler(input_codec_context, output_codec_context,
+                       &resample_context))
+        goto cleanup;
+    /** Initialize the FIFO buffer to store audio samples to be encoded. */
+    if (init_fifo(&fifo))
+        goto cleanup;
+    /** Write the header of the output file container. */
+    if (write_output_file_header(output_format_context))
+        goto cleanup;
+
+    /**
+     * Loop as long as we have input samples to read or output samples
+     * to write; abort as soon as we have neither.
+     */
+    while (1) {
+        /** Use the encoder's desired frame size for processing. */
+        const int output_frame_size = output_codec_context->frame_size;
+        int finished                = 0;
+
+        /**
+         * Make sure that there is one frame worth of samples in the FIFO
+         * buffer so that the encoder can do its work.
+         * Since the decoder's and the encoder's frame size may differ, we
+         * need to FIFO buffer to store as many frames worth of input samples
+         * that they make up at least one frame worth of output samples.
+         */
+        while (av_audio_fifo_size(fifo) < output_frame_size) {
+            /**
+             * Decode one frame worth of audio samples, convert it to the
+             * output sample format and put it into the FIFO buffer.
+             */
+            if (read_decode_convert_and_store(fifo, input_format_context,
+                                              input_codec_context,
+                                              output_codec_context,
+                                              resample_context, &finished))
+                goto cleanup;
+
+            /**
+             * If we are at the end of the input file, we continue
+             * encoding the remaining audio samples to the output file.
+             */
+            if (finished)
+                break;
+        }
+
+        /**
+         * If we have enough samples for the encoder, we encode them.
+         * At the end of the file, we pass the remaining samples to
+         * the encoder.
+         */
+        while (av_audio_fifo_size(fifo) >= output_frame_size ||
+               (finished && av_audio_fifo_size(fifo) > 0))
+            /**
+             * Take one frame worth of audio samples from the FIFO buffer,
+             * encode it and write it to the output file.
+             */
+            if (load_encode_and_write(fifo, output_format_context,
+                                      output_codec_context))
+                goto cleanup;
+
+        /**
+         * If we are at the end of the input file and have encoded
+         * all remaining samples, we can exit this loop and finish.
+         */
+        if (finished) {
+            int data_written;
+            /** Flush the encoder as it may have delayed frames. */
+            do {
+                if (encode_audio_frame(NULL, output_format_context,
+                                       output_codec_context, &data_written))
+                    goto cleanup;
+            } while (data_written);
+            break;
+        }
+    }
+
+    /** Write the trailer of the output file container. */
+    if (write_output_file_trailer(output_format_context))
+        goto cleanup;
+    ret = 0;
+
+cleanup:
+    if (fifo)
+        av_audio_fifo_free(fifo);
+    swr_free(&resample_context);
+    if (output_codec_context)
+        avcodec_close(output_codec_context);
+    if (output_format_context) {
+        avio_close(output_format_context->pb);
+        avformat_free_context(output_format_context);
+    }
+    if (input_codec_context)
+        avcodec_close(input_codec_context);
+    if (input_format_context)
+        avformat_close_input(&input_format_context);
+
+    return ret;
+}
--- a/doc/faq.texi
+++ b/doc/faq.texi
@@ -105,7 +105,7 @@ For example, img1.jpg, img2.jpg, img3.jpg,...
 Then you may run:

@example
-  ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
+ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
@end example

 Notice that @samp{%d} is replaced by the image number.
@@ -118,7 +118,7 @@ the sequence. This is useful if your sequence does not start with
 example will start with @file{img100.jpg}:

@example
-  ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
+ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
@end example

 If you have large number of pictures to rename, you can use the
@@ -128,7 +128,7 @@ that match @code{*jpg} to the @file{/tmp} directory in the sequence of
@file{img001.jpg}, @file{img002.jpg} and so on.

@example
-  x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
+x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
@end example

 If you want to sequence them by oldest modified first, substitute
@@ -137,7 +137,7 @@ If you want to sequence them by oldest modified first, substitute
 Then run:

@example
-  ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
+ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
@end example

 The same logic is used for any image format that ffmpeg reads.
@@ -145,7 +145,7 @@ The same logic is used for any image format that ffmpeg reads.
 You can also use @command{cat} to pipe images to ffmpeg:

@example
-  cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
+cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
@end example

@section How do I encode movie to single pictures?
@@ -153,7 +153,7 @@ You can also use @command{cat} to pipe images to ffmpeg:
 Use:

@example
-  ffmpeg -i movie.mpg movie%d.jpg
+ffmpeg -i movie.mpg movie%d.jpg
@end example

 The @file{movie.mpg} used as input will be converted to
@@ -169,7 +169,7 @@ to force the encoding.

 Applying that to the previous example:
@example
-  ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
+ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
@end example

 Beware that there is no "jpeg" codec. Use "mjpeg" instead.
@@ -227,15 +227,15 @@ then you may use any file that DirectShow can read as input.

 Just create an "input.avs" text file with this single line ...
@example
-  DirectShowSource("C:\path to your file\yourfile.asf")
+DirectShowSource("C:\path to your file\yourfile.asf")
@end example
 ... and then feed that text file to ffmpeg:
@example
-  ffmpeg -i input.avs
+ffmpeg -i input.avs
@end example

-For ANY other help on Avisynth, please visit the
-@uref{http://www.avisynth.org/, Avisynth homepage}.
+For ANY other help on AviSynth, please visit the
+@uref{http://www.avisynth.org/, AviSynth homepage}.

@section How can I join video files?

@@ -368,26 +368,6 @@ ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
 rm temp[12].[av] all.[av]
@end example

-@section -profile option fails when encoding H.264 video with AAC audio
-
-@command{ffmpeg} prints an error like
-
-@example
-Undefined constant or missing '(' in 'baseline'
-Unable to parse option value "baseline"
-Error setting option profile to value baseline.
-@end example
-
-Short answer: write @option{-profile:v} instead of @option{-profile}.
-
-Long answer: this happens because the @option{-profile} option can apply to both
-video and audio.  Specifically the AAC encoder also defines some profiles, none
-of which are named @var{baseline}.
-
-The solution is to apply the @option{-profile} option to the video stream only
-by using @url{http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1, Stream specifiers}.
-Appending @code{:v} to it will do exactly that.
-
@section Using @option{-f lavfi}, audio becomes mono for no apparent reason.

 Use @option{-dumpgraph -} to find out exactly where the channel layout is
@@ -475,9 +455,10 @@ read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.

@section Why are the ffmpeg programs devoid of debugging symbols?

-The build process creates ffmpeg_g, ffplay_g, etc. which contain full debug
-information. Those binaries are stripped to create ffmpeg, ffplay, etc. If
-you need the debug information, use the *_g versions.
+The build process creates @command{ffmpeg_g}, @command{ffplay_g}, etc. which
+contain full debug information. Those binaries are stripped to create
+@command{ffmpeg}, @command{ffplay}, etc. If you need the debug information, use
+the *_g versions.

@section I do not like the LGPL, can I contribute code under the GPL instead?

@@ -497,7 +478,7 @@ An easy way to get the full list of required libraries in dependency order
 is to use @code{pkg-config}.

@example
-  c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
+c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
@end example

 See @file{doc/example/Makefile} and @file{doc/example/pc-uninstalled} for
@@ -521,10 +502,6 @@ to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS
 You have to create a custom AVIOContext using @code{avio_alloc_context},
 see @file{libavformat/aviobuf.c} in FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer or MPlayer2 sources.

-@section Where can I find libav* headers for Pascal/Delphi?
-
-see @url{http://www.iversenit.dk/dev/ffmpeg-headers/}
-
@section Where is the documentation about ffv1, msmpeg4, asv1, 4xm?

 see @url{http://www.ffmpeg.org/~michael/}
@@ -537,11 +514,12 @@ In this specific case please look at RFC 4629 to see how it should be done.

@section AVStream.r_frame_rate is wrong, it is much larger than the frame rate.

-r_frame_rate is NOT the average frame rate, it is the smallest frame rate
+@code{r_frame_rate} is NOT the average frame rate, it is the smallest frame rate
 that can accurately represent all timestamps. So no, it is not
 wrong if it is larger than the average!
-For example, if you have mixed 25 and 30 fps content, then r_frame_rate
-will be 150.
+For example, if you have mixed 25 and 30 fps content, then @code{r_frame_rate}
+will be 150 (it is the least common multiple).
+If you are looking for the average frame rate, see @code{AVStream.avg_frame_rate}.

@section Why is @code{make fate} not running all tests?

--- a/doc/fate.texi
+++ b/doc/fate.texi
@@ -153,20 +153,20 @@ the synchronisation of the samples directory.

@table @option
@item fate-rsync
-    Download/synchronize sample files to the configured samples directory.
+Download/synchronize sample files to the configured samples directory.

@item fate-list
-    Will list all fate/regression test targets.
+Will list all fate/regression test targets.

@item fate
-    Run the FATE test suite (requires the fate-suite dataset).
+Run the FATE test suite (requires the fate-suite dataset).
@end table

@section Makefile variables

@table @option
@item V
-    Verbosity level, can be set to 0, 1 or 2.
+Verbosity level, can be set to 0, 1 or 2.
    @itemize
        @item 0: show just the test arguments
        @item 1: show just the command used in the test
@@ -174,22 +174,26 @@ the synchronisation of the samples directory.
    @end itemize

@item SAMPLES
-    Specify or override the path to the FATE samples at make time, it has a
-    meaning only while running the regression tests.
+Specify or override the path to the FATE samples at make time, it has a
+meaning only while running the regression tests.

@item THREADS
-    Specify how many threads to use while running regression tests, it is
-    quite useful to detect thread-related regressions.
+Specify how many threads to use while running regression tests, it is
+quite useful to detect thread-related regressions.
+
@item THREAD_TYPE
-    Specify which threading strategy test, either @var{slice} or @var{frame},
-    by default @var{slice+frame}
+Specify which threading strategy test, either @var{slice} or @var{frame},
+by default @var{slice+frame}
+
@item CPUFLAGS
-    Specify CPU flags.
+Specify CPU flags.
+
@item TARGET_EXEC
-    Specify or override the wrapper used to run the tests.
-    The @var{TARGET_EXEC} option provides a way to run FATE wrapped in
-    @command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
-    through @command{ssh}.
+Specify or override the wrapper used to run the tests.
+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
--- a/doc/ffmpeg-resampler.texi
+++ b/doc/ffmpeg-resampler.texi
@@ -12,9 +12,9 @@
@chapter Description
@c man begin DESCRIPTION

-The FFmpeg resampler provides an high-level interface to the
+The FFmpeg resampler provides a high-level interface to the
 libswresample library audio resampling utilities. In particular it
-allows to perform audio resampling, audio channel layout rematrixing,
+allows one to perform audio resampling, audio channel layout rematrixing,
 and convert audio format and packing layout.

@c man end DESCRIPTION
--- a/doc/ffmpeg-scaler.texi
+++ b/doc/ffmpeg-scaler.texi
@@ -12,8 +12,8 @@
@chapter Description
@c man begin DESCRIPTION

-The FFmpeg rescaler provides an high-level interface to the libswscale
-library image conversion utilities. In particular it allows to perform
+The FFmpeg rescaler provides a high-level interface to the libswscale
+library image conversion utilities. In particular it allows one to perform
 image rescaling and pixel format conversion.

@c man end DESCRIPTION
--- a/doc/ffmpeg.texi
+++ b/doc/ffmpeg.texi
@@ -80,11 +80,22 @@ The transcoding process in @command{ffmpeg} for each output can be described by
 the following diagram:

@example
- _______              ______________               _________              ______________            ________
-|       |            |              |             |         |            |              |          |        |
-| input |  demuxer   | encoded data |   decoder   | decoded |  encoder   | encoded data |  muxer   | output |
-| file  | ---------> | packets      |  ---------> | frames  | ---------> | packets      | -------> | file   |
-|_______|            |______________|             |_________|            |______________|          |________|
+ _______              ______________
+|       |            |              |
+| input |  demuxer   | encoded data |   decoder
+| file  | ---------> | packets      | -----+
+|_______|            |______________|      |
+                                           v
+                                       _________
+                                      |         |
+                                      | decoded |
+                                      | frames  |
+ ________             ______________  |_________|
+|        |           |              |      |
+| output | <-------- | encoded data | <----+
+| file   |   muxer   | packets      |   encoder
+|________|           |______________|
+

@end example

@@ -112,11 +123,11 @@ the same type. In the above diagram they can be represented by simply inserting
 an additional step between decoding and encoding:

@example
- _________                        __________              ______________
-|         |                      |          |            |              |
-| decoded |  simple filtergraph  | filtered |  encoder   | encoded data |
-| frames  | -------------------> | frames   | ---------> | packets      |
-|_________|                      |__________|            |______________|
+ _________               __________              ______________
+|         |  simple     |          |            |              |
+| decoded |  fltrgrph   | filtered |  encoder   | encoded data |
+| frames  | ----------> | frames   | ---------> | packets      |
+|_________|             |__________|            |______________|

@end example

@@ -125,10 +136,10 @@ Simple filtergraphs are configured with the per-stream @option{-filter} option
 A simple filtergraph for video can look for example like this:

@example
- _______        _____________        _______        _____        ________
-|       |      |             |      |       |      |     |      |        |
-| input | ---> | deinterlace | ---> | scale | ---> | fps | ---> | output |
-|_______|      |_____________|      |_______|      |_____|      |________|
+ _______        _____________        _______        ________
+|       |      |             |      |       |      |        |
+| input | ---> | deinterlace | ---> | scale | ---> | output |
+|_______|      |_____________|      |_______|      |________|

@end example

@@ -214,7 +225,7 @@ described.
@chapter Options
@c man begin OPTIONS

-@include avtools-common-opts.texi
+@include fftools-common-opts.texi

@section Main options

@@ -272,30 +283,33 @@ Set the file size limit, expressed in bytes.

@item -ss @var{position} (@emph{input/output})
 When used as an input option (before @code{-i}), seeks in this input file to
-@var{position}. When used as an output option (before an output filename),
-decodes but discards input until the timestamps reach @var{position}. This is
-slower, but more accurate.
+@var{position}. Note the in most formats it is not possible to seek exactly, so
+@command{ffmpeg} will seek to the closest seek point before @var{position}.
+When transcoding and @option{-accurate_seek} is enabled (the default), this
+extra segment between the seek point and @var{position} will be decoded and
+discarded. When doing stream copy or when @option{-noaccurate_seek} is used, it
+will be preserved.
+
+When used as an output option (before an output filename), decodes but discards
+input until the timestamps reach @var{position}.

@var{position} may be either in seconds or in @code{hh:mm:ss[.xxx]} form.

@item -itsoffset @var{offset} (@emph{input})
-Set the input time offset in seconds.
-@code{[-]hh:mm:ss[.xxx]} syntax is also supported.
-The offset is added to the timestamps of the input files.
-Specifying a positive offset means that the corresponding
-streams are delayed by @var{offset} seconds.
+Set the input time offset.

-@item -timestamp @var{time} (@emph{output})
+@var{offset} must be a time duration specification,
+see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+
+The offset is added to the timestamps of the input files. Specifying
+a positive offset means that the corresponding streams are delayed by
+the time duration specified in @var{offset}.
+
+@item -timestamp @var{date} (@emph{output})
 Set the recording timestamp in the container.
-The syntax for @var{time} is:
-@example
-now|([(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...])|(HHMMSS[.m...]))[Z|z])
-@end example
-If the value is "now" it takes the current time.
-Time is local time unless 'Z' or '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.
+
+@var{date} must be a time duration specification,
+see @ref{date syntax,,the Date section in the ffmpeg-utils(1) manual,ffmpeg-utils}.

@item -metadata[:metadata_specifier] @var{key}=@var{value} (@emph{output,per-metadata})
 Set a metadata key/value pair.
@@ -342,8 +356,13 @@ Stop writing to the stream after @var{framecount} frames.

@item -q[:@var{stream_specifier}] @var{q} (@emph{output,per-stream})
@itemx -qscale[:@var{stream_specifier}] @var{q} (@emph{output,per-stream})
-Use fixed quality scale (VBR). The meaning of @var{q} is
+Use fixed quality scale (VBR). The meaning of @var{q}/@var{qscale} is
 codec-dependent.
+If @var{qscale} is used without a @var{stream_specifier} then it applies only
+to the video stream, this is to maintain compatibility with previous behavior
+and as specifying the same codec specific value to 2 different codecs that is
+audio and video generally is not what is intended when no stream_specifier is
+used.

@anchor{filter_option}
@item -filter[:@var{stream_specifier}] @var{filtergraph} (@emph{output,per-stream})
@@ -497,9 +516,6 @@ prefix is ``ffmpeg2pass''. The complete file name will be
@file{PREFIX-N.log}, where N is a number specific to the output
 stream

-@item -vlang @var{code}
-Set the ISO 639 language code (3 letters) of the current video stream.
-
@item -vf @var{filtergraph} (@emph{output})
 Create the filtergraph specified by @var{filtergraph} and use it to
 filter the stream.
@@ -610,6 +626,42 @@ would be more efficient.
@item -copyinkf[:@var{stream_specifier}] (@emph{output,per-stream})
 When doing stream copy, copy also non-key frames found at the
 beginning.
+
+@item -hwaccel[:@var{stream_specifier}] @var{hwaccel} (@emph{input,per-stream})
+Use hardware acceleration to decode the matching stream(s). The allowed values
+of @var{hwaccel} are:
+@table @option
+@item none
+Do not use any hardware acceleration (the default).
+
+@item auto
+Automatically select the hardware acceleration method.
+
+@item vdpau
+Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration.
+@end table
+
+This option has no effect if the selected hwaccel is not available or not
+supported by the chosen decoder.
+
+Note that most acceleration methods are intended for playback and will not be
+faster than software decoding on modern CPUs. Additionally, @command{ffmpeg}
+will usually need to copy the decoded frames from the GPU memory into the system
+memory, resulting in further performance loss. This option is thus mainly
+useful for testing.
+
+@item -hwaccel_device[:@var{stream_specifier}] @var{hwaccel_device} (@emph{input,per-stream})
+Select a device to use for hardware acceleration.
+
+This option only makes sense when the @option{-hwaccel} option is also
+specified. Its exact meaning depends on the specific hardware acceleration
+method chosen.
+
+@table @option
+@item vdpau
+For VDPAU, this option specifies the X11 display/screen to use. If this option
+is not specified, the value of the @var{DISPLAY} environment variable is used
+@end table
@end table

@section Audio Options
@@ -662,8 +714,6 @@ stereo but not 6 channels as 5.1. The default is to always try to guess. Use
@section Subtitle options:

@table @option
-@item -slang @var{code}
-Set the ISO 639 language code (3 letters) of the current subtitle stream.
@item -scodec @var{codec} (@emph{input/output})
 Set the subtitle codec. This is an alias for @code{-codec:s}.
@item -sn (@emph{output})
@@ -871,13 +921,12 @@ Dump each input packet to stderr.
 When dumping packets, also dump the payload.
@item -re (@emph{input})
 Read input at native frame rate. Mainly used to simulate a grab device.
+or live input stream (e.g. when reading from a file). Should not be used
+with actual grab devices or live input streams (where it can cause packet
+loss).
 By default @command{ffmpeg} attempts to read the input(s) as fast as possible.
 This option will slow down the reading of the input(s) to the native frame rate
-of the input(s). It is useful for real-time output (e.g. live streaming). If
-your input(s) is coming from some other live streaming source (through HTTP or
-UDP for example) the server might already be in real-time, thus the option will
-likely not be required. On the other hand, this is meaningful if your input(s)
-is a file you are trying to push in real-time.
+of the input(s). It is useful for real-time output (e.g. live streaming).
@item -loop_input
 Loop over the input stream. Currently it works only for image
 streams. This option is used for automatic FFserver testing.
@@ -993,7 +1042,7 @@ ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264
 ffmpeg -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt
@end example

-@item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{per-stream})
+@item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{input/output,per-stream})
 Force a tag/fourcc for matching streams.

@item -timecode @var{hh}:@var{mm}:@var{ss}SEP@var{ff}
@@ -1060,13 +1109,21 @@ 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 -accurate_seek (@emph{input})
+This option enables or disables accurate seeking in input files with the
+@option{-ss} option. It is enabled by default, so seeking is accurate when
+transcoding. Use @option{-noaccurate_seek} to disable it, which may be useful
+e.g. when copying some streams and transcoding the others.
+
@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.
+Overrides the input specifications from @command{ffserver}. Using this
+option you can map any input stream to @command{ffserver} and control
+many aspects of the encoding from @command{ffmpeg}. Without this
+option @command{ffmpeg} will transmit to @command{ffserver} what is
+requested by @command{ffserver}.
+
 The option is intended for cases where features are needed that cannot be
-specified to ffserver but can be to ffmpeg.
+specified to @command{ffserver} but can be to @command{ffmpeg}.

@end table

@@ -1125,7 +1182,7 @@ then it will search for the file @file{libvpx-1080p.ffpreset}.

@itemize
@item
-For streaming at very low bitrate application, use a low frame rate
+For streaming at very low bitrates, use a low frame rate
 and a small GOP size. This is especially true for RealVideo where
 the Linux player does not seem to be very fast, so it can miss
 frames. An example is:
@@ -1204,14 +1261,14 @@ standard mixer.
 Grab the X11 display with ffmpeg via

@example
-ffmpeg -f x11grab -s cif -r 25 -i :0.0 /tmp/out.mpg
+ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0 /tmp/out.mpg
@end example

 0.0 is display.screen number of your X11 server, same as
 the DISPLAY environment variable.

@example
-ffmpeg -f x11grab -s cif -r 25 -i :0.0+10,20 /tmp/out.mpg
+ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0+10,20 /tmp/out.mpg
@end example

 0.0 is display.screen number of your X11 server, same as the DISPLAY environment
@@ -1348,11 +1405,11 @@ ffmpeg -f image2 -pattern_type glob -i 'foo-*.jpeg' -r 12 -s WxH foo.avi
 You can put many streams of the same type in the output:

@example
-ffmpeg -i test1.avi -i test2.avi -map 0:3 -map 0:2 -map 0:1 -map 0:0 -c copy test12.nut
+ffmpeg -i test1.avi -i test2.avi -map 1:1 -map 1:0 -map 0:1 -map 0:0 -c copy -y test12.nut
@end example

-The resulting output file @file{test12.avi} will contain first four streams from
-the input file in reverse order.
+The resulting output file @file{test12.nut} will contain the first four streams
+from the input files in reverse order.

@item
 To force CBR video output:
--- a/doc/ffplay.texi
+++ b/doc/ffplay.texi
@@ -24,7 +24,7 @@ various FFmpeg APIs.
@chapter Options
@c man begin OPTIONS

-@include avtools-common-opts.texi
+@include fftools-common-opts.texi

@section Main options

@@ -174,17 +174,26 @@ Toggle full screen.
 Pause.

@item a
-Cycle audio channel.
+Cycle audio channel in the curret program.

@item v
 Cycle video channel.

@item t
-Cycle subtitle channel.
+Cycle subtitle channel in the current program.
+
+@item c
+Cycle program.

@item w
 Show audio waves.

+@item s
+Step to the next frame.
+
+Pause if the stream is not already paused, step to the next video
+frame, and pause.
+
@item left/right
 Seek backward/forward 10 seconds.

@@ -192,6 +201,8 @@ Seek backward/forward 10 seconds.
 Seek backward/forward 1 minute.

@item page down/page up
+Seek to the previous/next chapter.
+or if there are no chapters
 Seek backward/forward 10 minutes.

@item mouse click
@@ -203,6 +214,7 @@ Seek to percentage in file corresponding to fraction of width.

@include config.texi
@ifset config-all
+@set config-readonly
@ifset config-avutil
@include utils.texi
@end ifset
--- a/doc/ffprobe.texi
+++ b/doc/ffprobe.texi
@@ -44,14 +44,15 @@ name (which may be shared by other sections), and an unique
 name. See the output of @option{sections}.

 Metadata tags stored in the container or in the streams are recognized
-and printed in the corresponding "FORMAT" or "STREAM" section.
+and printed in the corresponding "FORMAT", "STREAM" or "PROGRAM_STREAM"
+section.

@c man end

@chapter Options
@c man begin OPTIONS

-@include avtools-common-opts.texi
+@include fftools-common-opts.texi

@section Main options

@@ -112,7 +113,7 @@ ffprobe -show_packets -select_streams v:1 INPUT
@end example

@item -show_data
-Show payload data, as an hexadecimal and ASCII dump. Coupled with
+Show payload data, as a hexadecimal and ASCII dump. Coupled with
@option{-show_packets}, it will dump the packets' data. Coupled with
@option{-show_streams}, it will dump the codec extradata.

@@ -196,11 +197,11 @@ The information for each single packet is printed within a dedicated
 section with name "PACKET".

@item -show_frames
-Show information about each frame contained in the input multimedia
-stream.
+Show information about each frame and subtitle contained in the input
+multimedia stream.

 The information for each single frame is printed within a dedicated
-section with name "FRAME".
+section with name "FRAME" or "SUBTITLE".

@item -show_streams
 Show information about each media stream contained in the input
@@ -209,6 +210,13 @@ multimedia stream.
 Each media stream information is printed within a dedicated section
 with name "STREAM".

+@item -show_programs
+Show information about programs and their streams contained in the input
+multimedia stream.
+
+Each media stream information is printed within a dedicated section
+with name "PROGRAM_STREAM".
+
@item -show_chapters
 Show information about chapters stored in the format.

@@ -222,6 +230,70 @@ corresponding stream section.
 Count the number of packets per stream and report it in the
 corresponding stream section.

+@item -read_intervals @var{read_intervals}
+
+Read only the specified intervals. @var{read_intervals} must be a
+sequence of interval specifications separated by ",".
+@command{ffprobe} will seek to the interval starting point, and will
+continue reading from that.
+
+Each interval is specified by two optional parts, separated by "%".
+
+The first part specifies the interval start position. It is
+interpreted as an abolute position, or as a relative offset from the
+current position if it is preceded by the "+" character. If this first
+part is not specified, no seeking will be performed when reading this
+interval.
+
+The second part specifies the interval end position. It is interpreted
+as an absolute position, or as a relative offset from the current
+position if it is preceded by the "+" character. If the offset
+specification starts with "#", it is interpreted as the number of
+packets to read (not including the flushing packets) from the interval
+start. If no second part is specified, the program will read until the
+end of the input.
+
+Note that seeking is not accurate, thus the actual interval start
+point may be different from the specified position. Also, when an
+interval duration is specified, the absolute end time will be computed
+by adding the duration to the interval start point found by seeking
+the file, rather than to the specified start value.
+
+The formal syntax is given by:
+@example
+@var{INTERVAL}  ::= [@var{START}|+@var{START_OFFSET}][%[@var{END}|+@var{END_OFFSET}]]
+@var{INTERVALS} ::= @var{INTERVAL}[,@var{INTERVALS}]
+@end example
+
+A few examples follow.
+@itemize
+@item
+Seek to time 10, read packets until 20 seconds after the found seek
+point, then seek to position @code{01:30} (1 minute and thirty
+seconds) and read packets until position @code{01:45}.
+@example
+10%+20,01:30%01:45
+@end example
+
+@item
+Read only 42 packets after seeking to position @code{01:23}:
+@example
+01:23%+#42
+@end example
+
+@item
+Read only the first 20 seconds from the start:
+@example
+%+20
+@end example
+
+@item
+Read from the start until position @code{02:30}:
+@example
+%02:30
+@end example
+@end itemize
+
@item -show_private_data, -private
 Show private data, that is data depending on the format of the
 particular shown element.
@@ -265,6 +337,39 @@ A writer may accept one or more arguments, which specify the options
 to adopt. The options are specified as a list of @var{key}=@var{value}
 pairs, separated by ":".

+All writers support the following options:
+
+@table @option
+@item string_validation, sv
+Set string validation mode.
+
+The following values are accepted.
+@table @samp
+@item fail
+The writer will fail immediately in case an invalid string (UTF-8)
+sequence or code point is found in the input. This is especially
+useful to validate input metadata.
+
+@item ignore
+Any validation error will be ignored. This will result in possibly
+broken output, especially with the json or xml writer.
+
+@item replace
+The writer will substitute invalid UTF-8 sequences or code points with
+the string specified with the @option{string_validation_replacement}.
+@end table
+
+Default value is @samp{replace}.
+
+@item string_validation_replacement, svr
+Set replacement string to use in case @option{string_validation} is
+set to @samp{replace}.
+
+In case the option is not specified, the writer will assume the empty
+string, that is it will remove the invalid sequences from the input
+strings.
+@end table
+
 A description of the currently available writers follows.

@section default
@@ -279,8 +384,8 @@ keyN=valN
 [/SECTION]
@end example

-Metadata tags are printed as a line in the corresponding FORMAT or
-STREAM section, and are prefixed by the string "TAG:".
+Metadata tags are printed as a line in the corresponding FORMAT, STREAM or
+PROGRAM_STREAM section, and are prefixed by the string "TAG:".

 A description of the accepted options follows.

@@ -494,6 +599,7 @@ DV, GXF and AVI timecodes are available in format metadata

@include config.texi
@ifset config-all
+@set config-readonly
@ifset config-avutil
@include utils.texi
@end ifset
--- 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="programs" type="ffprobe:programsType" 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" />
@@ -27,7 +28,10 @@

    <xsd:complexType name="framesType">
        <xsd:sequence>
-            <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
+            <xsd:choice minOccurs="0" maxOccurs="unbounded">
+                <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
+                <xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
+            </xsd:choice>
        </xsd:sequence>
    </xsd:complexType>

@@ -57,6 +61,8 @@
      <xsd:attribute name="pkt_pts_time"  type="xsd:float"/>
      <xsd:attribute name="pkt_dts"       type="xsd:long" />
      <xsd:attribute name="pkt_dts_time"  type="xsd:float"/>
+      <xsd:attribute name="best_effort_timestamp"      type="xsd:long" />
+      <xsd:attribute name="best_effort_timestamp_time" type="xsd:float" />
      <xsd:attribute name="pkt_duration"  type="xsd:long" />
      <xsd:attribute name="pkt_duration_time" type="xsd:float"/>
      <xsd:attribute name="pkt_pos"       type="xsd:long" />
@@ -81,12 +87,28 @@
      <xsd:attribute name="repeat_pict"            type="xsd:int"   />
    </xsd:complexType>

+    <xsd:complexType name="subtitleType">
+      <xsd:attribute name="media_type"         type="xsd:string" fixed="subtitle" use="required"/>
+      <xsd:attribute name="pts"                type="xsd:long" />
+      <xsd:attribute name="pts_time"           type="xsd:float"/>
+      <xsd:attribute name="format"             type="xsd:int"  />
+      <xsd:attribute name="start_display_time" type="xsd:int"  />
+      <xsd:attribute name="end_display_time"   type="xsd:int"  />
+      <xsd:attribute name="num_rects"          type="xsd:int"  />
+    </xsd:complexType>
+
    <xsd:complexType name="streamsType">
        <xsd:sequence>
            <xsd:element name="stream" type="ffprobe:streamType" minOccurs="0" maxOccurs="unbounded"/>
        </xsd:sequence>
    </xsd:complexType>

+    <xsd:complexType name="programsType">
+        <xsd:sequence>
+            <xsd:element name="program" type="ffprobe:programType" minOccurs="0" maxOccurs="unbounded"/>
+        </xsd:sequence>
+    </xsd:complexType>
+
    <xsd:complexType name="streamDispositionType">
      <xsd:attribute name="default"          type="xsd:int" use="required" />
      <xsd:attribute name="dub"              type="xsd:int" use="required" />
@@ -103,8 +125,8 @@

    <xsd:complexType name="streamType">
      <xsd:sequence>
-        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
        <xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
+        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
      </xsd:sequence>

      <xsd:attribute name="index"            type="xsd:int" use="required"/>
@@ -131,6 +153,7 @@
      <xsd:attribute name="sample_fmt"       type="xsd:string"/>
      <xsd:attribute name="sample_rate"      type="xsd:int"/>
      <xsd:attribute name="channels"         type="xsd:int"/>
+      <xsd:attribute name="channel_layout"   type="xsd:string"/>
      <xsd:attribute name="bits_per_sample"  type="xsd:int"/>

      <xsd:attribute name="id"               type="xsd:string"/>
@@ -147,6 +170,23 @@
      <xsd:attribute name="nb_read_packets"  type="xsd:int"/>
    </xsd:complexType>

+    <xsd:complexType name="programType">
+      <xsd:sequence>
+        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
+        <xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1"/>
+      </xsd:sequence>
+
+      <xsd:attribute name="program_id"           type="xsd:int"    use="required"/>
+      <xsd:attribute name="program_num"          type="xsd:int"    use="required"/>
+      <xsd:attribute name="nb_streams"           type="xsd:int"    use="required"/>
+      <xsd:attribute name="start_time"           type="xsd:float"/>
+      <xsd:attribute name="start_pts"            type="xsd:long"/>
+      <xsd:attribute name="end_time"             type="xsd:float"/>
+      <xsd:attribute name="end_pts"              type="xsd:long"/>
+      <xsd:attribute name="pmt_pid"              type="xsd:int"    use="required"/>
+      <xsd:attribute name="pcr_pid"              type="xsd:int"    use="required"/>
+    </xsd:complexType>
+
    <xsd:complexType name="formatType">
      <xsd:sequence>
        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
@@ -154,12 +194,14 @@

      <xsd:attribute name="filename"         type="xsd:string" use="required"/>
      <xsd:attribute name="nb_streams"       type="xsd:int"    use="required"/>
+      <xsd:attribute name="nb_programs"      type="xsd:int"    use="required"/>
      <xsd:attribute name="format_name"      type="xsd:string" use="required"/>
      <xsd:attribute name="format_long_name" type="xsd:string"/>
      <xsd:attribute name="start_time"       type="xsd:float"/>
      <xsd:attribute name="duration"         type="xsd:float"/>
      <xsd:attribute name="size"             type="xsd:long"/>
      <xsd:attribute name="bit_rate"         type="xsd:long"/>
+      <xsd:attribute name="probe_score"      type="xsd:int"/>
    </xsd:complexType>

    <xsd:complexType name="tagType">
--- a/doc/ffserver.conf
+++ b/doc/ffserver.conf
@@ -235,7 +235,7 @@ StartSendOnKey

 #<Stream test.ogg>
 #Feed feed1.ffm
-#Title "Stream title"
+#Metadata title "Stream title"
 #AudioBitRate 64
 #AudioChannels 2
 #AudioSampleRate 44100
@@ -280,10 +280,10 @@ StartSendOnKey
 #<Stream file.asf>
 #File "/usr/local/httpd/htdocs/test.asf"
 #NoAudio
-#Author "Me"
-#Copyright "Super MegaCorp"
-#Title "Test stream from disk"
-#Comment "Test comment"
+#Metadata author "Me"
+#Metadata copyright "Super MegaCorp"
+#Metadata title "Test stream from disk"
+#Metadata comment "Test comment"
 #</Stream>


--- a/doc/ffserver.texi
+++ b/doc/ffserver.texi
@@ -16,11 +16,14 @@ ffserver [@var{options}]
@chapter Description
@c man begin DESCRIPTION

-@command{ffserver} is a streaming server for both audio and video. It
-supports several live feeds, streaming from files and time shifting on
-live feeds (you can seek to positions in the past on each live feed,
-provided you specify a big enough feed storage in
-@file{ffserver.conf}).
+@command{ffserver} is a streaming server for both audio and video.
+It supports several live feeds, streaming from files and time shifting
+on live feeds. You can seek to positions in the past on each live
+feed, provided you specify a big enough feed storage.
+
+@command{ffserver} is configured through a configuration file, which
+is read at startup. If not explicitly specified, it will read from
+@file{/etc/ffserver.conf}.

@command{ffserver} receives prerecorded files or FFM streams from some
@command{ffmpeg} instance as input, then streams them over
@@ -39,10 +42,126 @@ For each feed you can have different output streams in various
 formats, each one specified by a @code{<Stream>} section in the
 configuration file.

+@chapter Detailed description
+
+@command{ffserver} works by forwarding streams encoded by
+@command{ffmpeg}, or pre-recorded streams which are read from disk.
+
+Precisely, @command{ffserver} acts as an HTTP server, accepting POST
+requests from @command{ffmpeg} to acquire the stream to publish, and
+serving RTSP clients or HTTP clients GET requests with the stream
+media content.
+
+A feed is an @ref{FFM} stream created by @command{ffmpeg}, and sent to
+a port where @command{ffserver} is listening.
+
+Each feed is identified by a unique name, corresponding to the name
+of the resource published on @command{ffserver}, and is configured by
+a dedicated @code{Feed} section in the configuration file.
+
+The feed publish URL is given by:
+@example
+http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name}
+@end example
+
+where @var{ffserver_ip_address} is the IP address of the machine where
+@command{ffserver} is installed, @var{http_port} is the port number of
+the HTTP server (configured through the @option{Port} option), and
+@var{feed_name} is the name of the corresponding feed defined in the
+configuration file.
+
+Each feed is associated to a file which is stored on disk. This stored
+file is used to allow to send pre-recorded data to a player as fast as
+possible when new content is added in real-time to the stream.
+
+A "live-stream" or "stream" is a resource published by
+@command{ffserver}, and made accessible through the HTTP protocol to
+clients.
+
+A stream can be connected to a feed, or to a file. In the first case,
+the published stream is forwarded from the corresponding feed
+generated by a running instance of @command{ffmpeg}, in the second
+case the stream is read from a pre-recorded file.
+
+Each stream is identified by a unique name, corresponding to the name
+of the resource served by @command{ffserver}, and is configured by
+a dedicated @code{Stream} section in the configuration file.
+
+The stream access HTTP URL is given by:
+@example
+http://@var{ffserver_ip_address}:@var{http_port}/@var{stream_name}[@var{options}]
+@end example
+
+The stream access RTSP URL is given by:
+@example
+http://@var{ffserver_ip_address}:@var{rtsp_port}/@var{stream_name}[@var{options}]
+@end example
+
+@var{stream_name} is the name of the corresponding stream defined in
+the configuration file. @var{options} is a list of options specified
+after the URL which affects how the stream is served by
+@command{ffserver}. @var{http_port} and @var{rtsp_port} are the HTTP
+and RTSP ports configured with the options @var{Port} and
+@var{RTSPPort} respectively.
+
+In case the stream is associated to a feed, the encoding parameters
+must be configured in the stream configuration. They are sent to
+@command{ffmpeg} when setting up the encoding. This allows
+@command{ffserver} to define the encoding parameters used by
+the @command{ffmpeg} encoders.
+
+The @command{ffmpeg} @option{override_ffserver} commandline option
+allows one to override the encoding parameters set by the server.
+
+Multiple streams can be connected to the same feed.
+
+For example, you can have a situation described by the following
+graph:
+@example
+               _________       __________
+              |         |     |          |
+ffmpeg 1 -----| feed 1  |-----| stream 1 |
+    \         |_________|\    |__________|
+     \                    \
+      \                    \   __________
+       \                    \ |          |
+        \                    \| stream 2 |
+         \                    |__________|
+          \
+           \   _________       __________
+            \ |         |     |          |
+             \| feed 2  |-----| stream 3 |
+              |_________|     |__________|
+
+               _________       __________
+              |         |     |          |
+ffmpeg 2 -----| feed 3  |-----| stream 4 |
+              |_________|     |__________|
+
+               _________       __________
+              |         |     |          |
+              | file 1  |-----| stream 5 |
+              |_________|     |__________|
+@end example
+
+@anchor{FFM}
+@section FFM, FFM2 formats
+
+FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
+video and audio streams and encoding options, and can store a moving time segment
+of an infinite movie or a whole movie.
+
+FFM is version specific, and there is limited compatibility of FFM files
+generated by one version of ffmpeg/ffserver and another version of
+ffmpeg/ffserver. It may work but it is not guaranteed to work.
+
+FFM2 is extensible while maintaining compatibility and should work between
+differing versions of tools. FFM2 is the default.
+
@section Status stream

-ffserver supports an HTTP interface which exposes the current status
-of the server.
+@command{ffserver} supports an HTTP interface which exposes the
+current status of the server.

 Simply point your browser to the address of the special status stream
 specified in the configuration file.
@@ -61,27 +180,8 @@ ACL allow 192.168.0.0 192.168.255.255
 then the server will post a page with the status information when
 the special stream @file{status.html} is requested.

-@section What can this do?
-
-When properly configured and running, you can capture video and audio in real
-time from a suitable capture card, and stream it out over the Internet to
-either Windows Media Player or RealAudio player (with some restrictions).
-
-It can also stream from files, though that is currently broken. Very often, a
-web server can be used to serve up the files just as well.
-
-It can stream prerecorded video from .ffm files, though it is somewhat tricky
-to make it work correctly.
-
@section How do I make it work?

-First, build the kit. It *really* helps to have installed LAME first. Then when
-you run the ffserver ./configure, make sure that you have the
-@code{--enable-libmp3lame} flag turned on.
-
-LAME is important as it allows for streaming audio to Windows Media Player.
-Don't ask why the other audio types do not work.
-
 As a simple test, just run the following two command lines where INPUTFILE
 is some file which you can decode with ffmpeg:

@@ -109,35 +209,6 @@ You should edit the ffserver.conf file to suit your needs (in terms of
 frame rates etc). Then install ffserver and ffmpeg, write a script to start
 them up, and off you go.

-@section Troubleshooting
-
-@subsection I don't hear any audio, but video is fine.
-
-Maybe you didn't install LAME, or got your ./configure statement wrong. Check
-the ffmpeg output to see if a line referring to MP3 is present. If not, then
-your configuration was incorrect. If it is, then maybe your wiring is not
-set up correctly. Maybe the sound card is not getting data from the right
-input source. Maybe you have a really awful audio interface (like I do)
-that only captures in stereo and also requires that one channel be flipped.
-If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before
-starting ffmpeg.
-
-@subsection The audio and video lose sync after a while.
-
-Yes, they do.
-
-@subsection After a long while, the video update rate goes way down in WMP.
-
-Yes, it does. Who knows why?
-
-@subsection WMP 6.4 behaves differently to WMP 7.
-
-Yes, it does. Any thoughts on this would be gratefully received. These
-differences extend to embedding WMP into a web page. [There are two
-object IDs that you can use: The old one, which does not play well, and
-the new one, which does (both tested on the same system). However,
-I suspect that the new one is not available unless you have installed WMP 7].
-
@section What else can it do?

 You can replay video from .ffm files that was recorded earlier.
@@ -177,9 +248,6 @@ specify a time. In addition, ffserver will skip frames until a key_frame
 is found. This further reduces the startup delay by not transferring data
 that will be discarded.

-* You may want to adjust the MaxBandwidth in the ffserver.conf to limit
-the amount of bandwidth consumed by live streams.
-
@section Why does the ?buffer / Preroll stop working after a time?

 It turns out that (on my machine at least) the number of frames successfully
@@ -213,37 +281,549 @@ You use this by adding the ?date= to the end of the URL for the stream.
 For example:   @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
@c man end

-@section What is FFM, FFM2
-
-FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
-video and audio streams and encoding options, and can store a moving time segment
-of an infinite movie or a whole movie.
-
-FFM is version specific, and there is limited compatibility of FFM files
-generated by one version of ffmpeg/ffserver and another version of
-ffmpeg/ffserver. It may work but it is not guaranteed to work.
-
-FFM2 is extensible while maintaining compatibility and should work between
-differing versions of tools. FFM2 is the default.
-
@chapter Options
@c man begin OPTIONS

-@include avtools-common-opts.texi
+@include fftools-common-opts.texi

@section Main options

@table @option
@item -f @var{configfile}
-Use @file{configfile} instead of @file{/etc/ffserver.conf}.
+Read configuration file @file{configfile}. If not specified it will
+read by default from @file{/etc/ffserver.conf}.
+
@item -n
-Enable no-launch mode. This option disables all the Launch directives
-within the various <Stream> sections. Since ffserver will not launch
-any ffmpeg instances, you will have to launch them manually.
+Enable no-launch mode. This option disables all the @code{Launch}
+directives within the various @code{<Feed>} sections. Since
+@command{ffserver} will not launch any @command{ffmpeg} instances, you
+will have to launch them manually.
+
@item -d
-Enable debug mode. This option increases log verbosity, directs log
-messages to stdout.
+Enable debug mode. This option increases log verbosity, and directs
+log messages to stdout. When specified, the @option{CustomLog} option
+is ignored.
@end table
+
+@chapter Configuration file syntax
+
+@command{ffserver} reads a configuration file containing global
+options and settings for each stream and feed.
+
+The configuration file consists of global options and dedicated
+sections, which must be introduced by "<@var{SECTION_NAME}
+@var{ARGS}>" on a separate line and must be terminated by a line in
+the form "</@var{SECTION_NAME}>". @var{ARGS} is optional.
+
+Currently the following sections are recognized: @samp{Feed},
+@samp{Stream}, @samp{Redirect}.
+
+A line starting with @code{#} is ignored and treated as a comment.
+
+Name of options and sections are case-insensitive.
+
+@section ACL syntax
+An ACL (Access Control List) specifies the address which are allowed
+to access a given stream, or to write a given feed.
+
+It accepts the folling forms
+@itemize
+@item
+Allow/deny access to @var{address}.
+@example
+ACL ALLOW <address>
+ACL DENY <address>
+@end example
+
+@item
+Allow/deny access to ranges of addresses from @var{first_address} to
+@var{last_address}.
+@example
+ACL ALLOW <first_address> <last_address>
+ACL DENY <first_address> <last_address>
+@end example
+@end itemize
+
+You can repeat the ACL allow/deny as often as you like. It is on a per
+stream basis. The first match defines the action. If there are no matches,
+then the default is the inverse of the last ACL statement.
+
+Thus 'ACL allow localhost' only allows access from localhost.
+'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
+allow everybody else.
+
+@section Global options
+@table @option
+@item Port @var{port_number}
+@item RTSPPort @var{port_number}
+
+Set TCP port number on which the HTTP/RTSP server is listening. You
+must select a different port from your standard HTTP web server if it
+is running on the same computer.
+
+If not specified, no corresponding server will be created.
+
+@item BindAddress @var{ip_address}
+@item RTSPBindAddress @var{ip_address}
+Set address on which the HTTP/RTSP server is bound. Only useful if you
+have several network interfaces.
+
+@item MaxHTTPConnections @var{n}
+Set number of simultaneous HTTP connections that can be handled. It
+has to be defined @emph{before} the @option{MaxClients} parameter,
+since it defines the @option{MaxClients} maximum limit.
+
+Default value is 2000.
+
+@item MaxClients @var{n}
+Set number of simultaneous requests that can be handled. Since
+@command{ffserver} is very fast, it is more likely that you will want
+to leave this high and use @option{MaxBandwidth}.
+
+Default value is 5.
+
+@item MaxBandwidth @var{kbps}
+Set the maximum amount of kbit/sec that you are prepared to consume
+when streaming to clients.
+
+Default value is 1000.
+
+@item CustomLog @var{filename}
+Set access log file (uses standard Apache log file format). '-' is the
+standard output.
+
+If not specified @command{ffserver} will produce no log.
+
+In case the commandline option @option{-d} is specified this option is
+ignored, and the log is written to standard output.
+
+@item NoDaemon
+Set no-daemon mode. This option is currently ignored since now
+@command{ffserver} will always work in no-daemon mode, and is
+deprecated.
+@end table
+
+@section Feed section
+
+A Feed section defines a feed provided to @command{ffserver}.
+
+Each live feed contains one video and/or audio sequence coming from an
+@command{ffmpeg} encoder or another @command{ffserver}. This sequence
+may be encoded simultaneously with several codecs at several
+resolutions.
+
+A feed instance specification is introduced by a line in the form:
+@example
+<Feed FEED_FILENAME>
+@end example
+
+where @var{FEED_FILENAME} specifies the unique name of the FFM stream.
+
+The following options are recognized within a Feed section.
+
+@table @option
+@item File @var{filename}
+@item ReadOnlyFile @var{filename}
+Set the path where the feed file is stored on disk.
+
+If not specified, the @file{/tmp/FEED.ffm} is assumed, where
+@var{FEED} is the feed name.
+
+If @option{ReadOnlyFile} is used the file is marked as read-only and
+it will not be deleted or updated.
+
+@item Truncate
+Truncate the feed file, rather than appending to it. By default
+@command{ffserver} will append data to the file, until the maximum
+file size value is reached (see @option{FileMaxSize} option).
+
+@item FileMaxSize @var{size}
+Set maximum size of the feed file in bytes. 0 means unlimited. The
+postfixes @code{K} (2^10), @code{M} (2^20), and @code{G} (2^30) are
+recognized.
+
+Default value is 5M.
+
+@item Launch @var{args}
+Launch an @command{ffmpeg} command when creating @command{ffserver}.
+
+@var{args} must be a sequence of arguments to be provided to an
+@command{ffmpeg} instance. The first provided argument is ignored, and
+it is replaced by a path with the same dirname of the @command{ffserver}
+instance, followed by the remaining argument and terminated with a
+path corresponding to the feed.
+
+When the launched process exits, @command{ffserver} will launch
+another program instance.
+
+In case you need a more complex @command{ffmpeg} configuration,
+e.g. if you need to generate multiple FFM feeds with a single
+@command{ffmpeg} instance, you should launch @command{ffmpeg} by hand.
+
+This option is ignored in case the commandline option @option{-n} is
+specified.
+
+@item ACL @var{spec}
+Specify the list of IP address which are allowed or denied to write
+the feed. Multiple ACL options can be specified.
+@end table
+
+@section Stream section
+
+A Stream section defines a stream provided by @command{ffserver}, and
+identified by a single name.
+
+The stream is sent when answering a request containing the stream
+name.
+
+A stream section must be introduced by the line:
+@example
+<Stream STREAM_NAME>
+@end example
+
+where @var{STREAM_NAME} specifies the unique name of the stream.
+
+The following options are recognized within a Stream section.
+
+Encoding options are marked with the @emph{encoding} tag, and they are
+used to set the encoding parameters, and are mapped to libavcodec
+encoding options. Not all encoding options are supported, in
+particular it is not possible to set encoder private options. In order
+to override the encoding options specified by @command{ffserver}, you
+can use the @command{ffmpeg} @option{override_ffserver} commandline
+option.
+
+Only one of the @option{Feed} and @option{File} options should be set.
+
+@table @option
+@item Feed @var{feed_name}
+Set the input feed. @var{feed_name} must correspond to an existing
+feed defined in a @code{Feed} section.
+
+When this option is set, encoding options are used to setup the
+encoding operated by the remote @command{ffmpeg} process.
+
+@item File @var{filename}
+Set the filename of the pre-recorded input file to stream.
+
+When this option is set, encoding options are ignored and the input
+file content is re-streamed as is.
+
+@item Format @var{format_name}
+Set the format of the output stream.
+
+Must be the name of a format recognized by FFmpeg. If set to
+@samp{status}, it is treated as a status stream.
+
+@item InputFormat @var{format_name}
+Set input format. If not specified, it is automatically guessed.
+
+@item Preroll @var{n}
+Set this to the number of seconds backwards in time to start. Note that
+most players will buffer 5-10 seconds of video, and also you need to allow
+for a keyframe to appear in the data stream.
+
+Default value is 0.
+
+@item StartSendOnKey
+Do not send stream until it gets the first key frame. By default
+@command{ffserver} will send data immediately.
+
+@item MaxTime @var{n}
+Set the number of seconds to run. This value set the maximum duration
+of the stream a client will be able to receive.
+
+A value of 0 means that no limit is set on the stream duration.
+
+@item ACL @var{spec}
+Set ACL for the stream.
+
+@item DynamicACL @var{spec}
+
+@item RTSPOption @var{option}
+
+@item MulticastAddress @var{address}
+
+@item MulticastPort @var{port}
+
+@item MulticastTTL @var{integer}
+
+@item NoLoop
+
+@item FaviconURL @var{url}
+Set favicon (favourite icon) for the server status page. It is ignored
+for regular streams.
+
+@item Author @var{value}
+@item Comment @var{value}
+@item Copyright @var{value}
+@item Title @var{value}
+Set metadata corresponding to the option. All these options are
+deprecated in favor of @option{Metadata}.
+
+@item Metadata @var{key} @var{value}
+Set metadata value on the output stream.
+
+@item NoAudio
+@item NoVideo
+Suppress audio/video.
+
+@item AudioCodec @var{codec_name} (@emph{encoding,audio})
+Set audio codec.
+
+@item AudioBitRate @var{rate} (@emph{encoding,audio})
+Set bitrate for the audio stream in kbits per second.
+
+@item AudioChannels @var{n} (@emph{encoding,audio})
+Set number of audio channels.
+
+@item AudioSampleRate @var{n} (@emph{encoding,audio})
+Set sampling frequency for audio. When using low bitrates, you should
+lower this frequency to 22050 or 11025. The supported frequencies
+depend on the selected audio codec.
+
+@item AVOptionAudio @var{option} @var{value} (@emph{encoding,audio})
+Set generic option for audio stream.
+
+@item AVPresetAudio @var{preset} (@emph{encoding,audio})
+Set preset for audio stream.
+
+@item VideoCodec @var{codec_name} (@emph{encoding,video})
+Set video codec.
+
+@item VideoBitRate @var{n} (@emph{encoding,video})
+Set bitrate for the video stream in kbits per second.
+
+@item VideoBitRateRange @var{range} (@emph{encoding,video})
+Set video bitrate range.
+
+A range must be specified in the form @var{minrate}-@var{maxrate}, and
+specifies the @option{minrate} and @option{maxrate} encoding options
+expressed in kbits per second.
+
+@item VideoBitRateRangeTolerance @var{n} (@emph{encoding,video})
+Set video bitrate tolerance in kbits per second.
+
+@item PixelFormat @var{pixel_format} (@emph{encoding,video})
+Set video pixel format.
+
+@item Debug @var{integer} (@emph{encoding,video})
+Set video @option{debug} encoding option.
+
+@item Strict @var{integer} (@emph{encoding,video})
+Set video @option{strict} encoding option.
+
+@item VideoBufferSize @var{n} (@emph{encoding,video})
+Set ratecontrol buffer size, expressed in KB.
+
+@item VideoFrameRate @var{n} (@emph{encoding,video})
+Set number of video frames per second.
+
+@item VideoSize (@emph{encoding,video})
+Set size of the video frame, must be an abbreviation or in the form
+@var{W}x@var{H}.  See @ref{video size syntax,,the Video size section
+in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+
+Default value is @code{160x128}.
+
+@item VideoIntraOnly (@emph{encoding,video})
+Transmit only intra frames (useful for low bitrates, but kills frame rate).
+
+@item VideoGopSize @var{n} (@emph{encoding,video})
+If non-intra only, an intra frame is transmitted every VideoGopSize
+frames. Video synchronization can only begin at an intra frame.
+
+@item VideoTag @var{tag} (@emph{encoding,video})
+Set video tag.
+
+@item VideoHighQuality (@emph{encoding,video})
+@item Video4MotionVector (@emph{encoding,video})
+
+@item BitExact (@emph{encoding,video})
+Set bitexact encoding flag.
+
+@item IdctSimple (@emph{encoding,video})
+Set simple IDCT algorithm.
+
+@item Qscale @var{n} (@emph{encoding,video})
+Enable constant quality encoding, and set video qscale (quantization
+scale) value, expressed in @var{n} QP units.
+
+@item VideoQMin @var{n} (@emph{encoding,video})
+@item VideoQMax @var{n} (@emph{encoding,video})
+Set video qmin/qmax.
+
+@item VideoQDiff @var{integer} (@emph{encoding,video})
+Set video @option{qdiff} encoding option.
+
+@item LumiMask @var{float} (@emph{encoding,video})
+@item DarkMask @var{float} (@emph{encoding,video})
+Set @option{lumi_mask}/@option{dark_mask} encoding options.
+
+@item AVOptionVideo @var{option} @var{value} (@emph{encoding,video})
+Set generic option for video stream.
+
+@item AVPresetVideo @var{preset} (@emph{encoding,video})
+Set preset for video stream.
+
+@var{preset} must be the path of a preset file.
+@end table
+
+@subsection Server status stream
+
+A server status stream is a special stream which is used to show
+statistics about the @command{ffserver} operations.
+
+It must be specified setting the option @option{Format} to
+@samp{status}.
+
+@section Redirect section
+
+A redirect section specifies where to redirect the requested URL to
+another page.
+
+A redirect section must be introduced by the line:
+@example
+<Redirect NAME>
+@end example
+
+where @var{NAME} is the name of the page which should be redirected.
+
+It only accepts the option @option{URL}, which specify the redirection
+URL.
+
+@chapter Stream examples
+
+@itemize
+@item
+Multipart JPEG
+@example
+<Stream test.mjpg>
+Feed feed1.ffm
+Format mpjpeg
+VideoFrameRate 2
+VideoIntraOnly
+NoAudio
+Strict -1
+</Stream>
+@end example
+
+@item
+Single JPEG
+@example
+<Stream test.jpg>
+Feed feed1.ffm
+Format jpeg
+VideoFrameRate 2
+VideoIntraOnly
+VideoSize 352x240
+NoAudio
+Strict -1
+</Stream>
+@end example
+
+@item
+Flash
+@example
+<Stream test.swf>
+Feed feed1.ffm
+Format swf
+VideoFrameRate 2
+VideoIntraOnly
+NoAudio
+</Stream>
+@end example
+
+@item
+ASF compatible
+@example
+<Stream test.asf>
+Feed feed1.ffm
+Format asf
+VideoFrameRate 15
+VideoSize 352x240
+VideoBitRate 256
+VideoBufferSize 40
+VideoGopSize 30
+AudioBitRate 64
+StartSendOnKey
+</Stream>
+@end example
+
+@item
+MP3 audio
+@example
+<Stream test.mp3>
+Feed feed1.ffm
+Format mp2
+AudioCodec mp3
+AudioBitRate 64
+AudioChannels 1
+AudioSampleRate 44100
+NoVideo
+</Stream>
+@end example
+
+@item
+Ogg Vorbis audio
+@example
+<Stream test.ogg>
+Feed feed1.ffm
+Metadata title "Stream title"
+AudioBitRate 64
+AudioChannels 2
+AudioSampleRate 44100
+NoVideo
+</Stream>
+@end example
+
+@item
+Real with audio only at 32 kbits
+@example
+<Stream test.ra>
+Feed feed1.ffm
+Format rm
+AudioBitRate 32
+NoVideo
+</Stream>
+@end example
+
+@item
+Real with audio and video at 64 kbits
+@example
+<Stream test.rm>
+Feed feed1.ffm
+Format rm
+AudioBitRate 32
+VideoBitRate 128
+VideoFrameRate 25
+VideoGopSize 25
+</Stream>
+@end example
+
+@item
+For stream coming from a file: you only need to set the input filename
+and optionally a new format.
+
+@example
+<Stream file.rm>
+File "/usr/local/httpd/htdocs/tlive.rm"
+NoAudio
+</Stream>
+@end example
+
+@example
+<Stream file.asf>
+File "/usr/local/httpd/htdocs/test.asf"
+NoAudio
+Metadata author "Me"
+Metadata copyright "Super MegaCorp"
+Metadata title "Test stream from disk"
+Metadata comment "Test comment"
+</Stream>
+@end example
+@end itemize
+
@c man end

@include config.texi
--- a/doc/fftools-common-opts.texi
+++ b/doc/fftools-common-opts.texi
@@ -59,10 +59,18 @@ Show license.

@item -h, -?, -help, --help [@var{arg}]
 Show help. An optional parameter may be specified to print help about a specific
-item.
+item. If no argument is specified, only basic (non advanced) tool
+options are shown.

 Possible values of @var{arg} are:
@table @option
+@item long
+Print advanced tool options in addition to the basic tool options.
+
+@item full
+Print complete list of options, including shared and private options
+for encoders, decoders, demuxers, muxers, filters, etc.
+
@item decoder=@var{decoder_name}
 Print detailed information about the decoder named @var{decoder_name}. Use the
@option{-decoders} option to get a list of all decoders.
@@ -82,7 +90,6 @@ Print detailed information about the muxer named @var{muxer_name}. Use the
@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
@@ -121,6 +128,9 @@ Show available sample formats.
@item -layouts
 Show channel names and standard channel layouts.

+@item -colors
+Show recognized color names.
+
@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
@@ -184,6 +194,13 @@ to a plain @code{%}
 Errors in parsing the environment variable are not fatal, and will not
 appear in the report.

+@item -hide_banner
+Suppress printing banner.
+
+All FFmpeg tools will normally show a copyright notice, build options
+and library versions. This option can be used to suppress printing
+this information.
+
@item -cpuflags flags (@emph{global})
 Allows setting and clearing cpu flags. This option is intended
 for testing. Do not use it unless you know what you're doing.
@@ -240,6 +257,10 @@ Possible flags for this option are:
@end table
@end table

+@item -opencl_bench
+Benchmark all available OpenCL devices and show the results. This option
+is only available when FFmpeg has been compiled with @code{--enable-opencl}.
+
@item -opencl_options options (@emph{global})
 Set OpenCL environment options. This option is only available when
 FFmpeg has been compiled with @code{--enable-opencl}.
@@ -271,11 +292,12 @@ muxer:
 ffmpeg -i input.flac -id3v2_version 3 out.mp3
@end example

-All codec AVOptions are obviously per-stream, so the chapter on stream
-specifiers applies to them
+All codec AVOptions are per-stream, and thus a stream specifier
+should be attached to them.

-Note @option{-nooption} syntax cannot be used for boolean AVOptions,
-use @option{-option 0}/@option{-option 1}.
+Note: the @option{-nooption} syntax cannot be used for boolean
+AVOptions, use @option{-option 0}/@option{-option 1}.

-Note2 old undocumented way of specifying per-stream AVOptions by prepending
-v/a/s to the options name is now obsolete and will be removed soon.
+Note: the old undocumented way of specifying per-stream AVOptions by
+prepending v/a/s to the options name is now obsolete and will be
+removed soon.
--- a/doc/filter_design.txt
+++ b/doc/filter_design.txt
@@ -166,7 +166,7 @@ Buffer references ownership and permissions
      WRITE permission.

    * Filters that read their input to produce a new frame on output (like
-      scale) need the READ permission on input and and must request a buffer
+      scale) need the READ permission on input and must request a buffer
      with the WRITE permission.

    * Filters that intend to keep a reference after the filtering process
--- a/doc/filters.texi
+++ b/doc/filters.texi
--- a/doc/formats.texi
+++ b/doc/formats.texi
@@ -57,6 +57,10 @@ Enable RTP MP4A-LATM payload.
 Reduce the latency introduced by optional buffering
@end table

+@item seek2any @var{integer} (@emph{input})
+Allow seeking to non-keyframes on demuxer level when supported if set to 1.
+Default is 0.
+
@item analyzeduration @var{integer} (@emph{input})
 Specify how many microseconds are analyzed to probe the input. A
 higher value will allow to detect more accurate information, but will
@@ -121,26 +125,88 @@ Consider things that a sane encoder should not do as an error.
 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.
+
+Possible values:
+@table @samp
+@item make_non_negative
+Shift timestamps to make them non-negative.
+Also note that this affects only leading negative timestamps, and not
+non-monotonic negative timestamps.
+@item make_zero
+Shift timestamps so that the first timestamp is 0.
+@item auto (default)
+Enables shifting when required by the target format.
+@item disabled
+Disables shifting of timestamp.
+@end table

 When shifting is enabled, all output timestamps are shifted by the
 same amount. Audio, video, and subtitles desynching and relative
 timestamp differences are preserved compared to how they would have
 been without shifting.

-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 of bytes to skip before reading header and frames if set to 1.
+Default is 0.
+
+@item correct_ts_overflow @var{integer} (@emph{input})
+Correct single timestamp overflows if set to 1. Default is 1.

@item flush_packets @var{integer} (@emph{output})
 Flush the underlying I/O stream after each packet. Default 1 enables it, and
 has the effect of reducing the latency; 0 disables it and may slightly
 increase performance in some cases.
+
+@item output_ts_offset @var{offset} (@emph{output})
+Set the output time offset.
+
+@var{offset} must be a time duration specification,
+see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+
+The offset is added by the muxer to the output timestamps.
+
+Specifying a positive offset means that the corresponding streams are
+delayed bt the time duration specified in @var{offset}. Default value
+is @code{0} (meaning that no offset is applied).
@end table

@c man end FORMAT OPTIONS

+@anchor{Format stream specifiers}
+@section Format stream specifiers
+
+Format stream specifiers allow selection of one or more streams that
+match specific properties.
+
+Possible forms of stream specifiers are:
+@table @option
+@item @var{stream_index}
+Matches the stream with this index.
+
+@item @var{stream_type}[:@var{stream_index}]
+@var{stream_type} is one of following: 'v' for video, 'a' for audio,
+'s' for subtitle, 'd' for data, and 't' for attachments. If
+@var{stream_index} is given, then it matches the 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 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 a format-specific ID.
+@end table
+
+The exact semantics of stream specifiers is defined by the
+@code{avformat_match_stream_specifier()} function declared in the
+@file{libavformat/avformat.h} header.
+
+@ifclear config-writeonly
@include demuxers.texi
+@end ifclear
+@ifclear config-readonly
@include muxers.texi
+@end ifclear
@include metadata.texi
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -94,7 +94,7 @@ Then pass @code{--enable-libtwolame} to configure to enable it.

@section libvpx

-FFmpeg can make use of the libvpx library for VP8 encoding.
+FFmpeg can make use of the libvpx library for VP8/VP9 encoding.

 Go to @url{http://www.webmproject.org/} and follow the instructions for
 installing the library. Then pass @code{--enable-libvpx} to configure to
@@ -122,6 +122,20 @@ x264 is under the GNU Public License Version 2 or later
 details), you must upgrade FFmpeg's license to GPL in order to use it.
@end float

+@section x265
+
+FFmpeg can make use of the x265 library for HEVC encoding.
+
+Go to @url{http://x265.org/developers.html} and follow the instructions
+for installing the library. Then pass @code{--enable-libx265} to configure
+to enable it.
+
+@float note
+x265 is under the GNU Public License Version 2 or later
+(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for
+details), you must upgrade FFmpeg's license to GPL in order to use it.
+@end float
+
@section libilbc

 iLBC is a narrowband speech codec that has been made freely available
@@ -133,6 +147,41 @@ Go to @url{https://github.com/dekkers/libilbc} and follow the instructions for
 installing the library. Then pass @code{--enable-libilbc} to configure to
 enable it.

+@section libzvbi
+
+libzvbi is a VBI decoding library which can be used by FFmpeg to decode DVB
+teletext pages and DVB teletext subtitles.
+
+Go to @url{http://sourceforge.net/projects/zapping/} and follow the instructions for
+installing the library. Then pass @code{--enable-libzvbi} to configure to
+enable it.
+
+@float NOTE
+libzvbi is licensed under the GNU General Public License Version 2 or later
+(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for details),
+you must upgrade FFmpeg's license to GPL in order to use it.
+@end float
+
+@section AviSynth
+
+FFmpeg can read AviSynth scripts as input. To enable support, pass
+@code{--enable-avisynth} to configure.  The correct headers are
+included in compat/avisynth/, which allows the user to enable support
+without needing to search for these headers themselves.
+
+For Windows, supported AviSynth variants are
+@url{http://avisynth.nl, AviSynth 2.5 or 2.6} for 32-bit builds and
+@url{http://avs-plus.net, AviSynth+ 0.1} for 32-bit and 64-bit builds.
+
+For Linux and OS X, the supported AviSynth variant is
+@url{https://github.com/avxsynth/avxsynth, AvxSynth}.
+
+@float NOTE
+AviSynth and AvxSynth are loaded dynamically.  Distributors can build FFmpeg
+with @code{--enable-avisynth}, and the binaries will work regardless of the
+end user having AviSynth or AvxSynth installed - they'll only need to be
+installed to use AviSynth scripts (obviously).
+@end float


@chapter Supported File Formats, Codecs or Features
@@ -168,7 +217,7 @@ library:
@item AST                       @tab X @tab X
    @tab Audio format used on the Nintendo Wii.
@item AVI                       @tab X @tab X
-@item AVISynth                  @tab   @tab X
+@item AviSynth                  @tab   @tab X
@item AVR                       @tab   @tab X
    @tab Audio format used on Mac.
@item AVS                       @tab   @tab X
@@ -235,6 +284,8 @@ library:
@item GXF                       @tab X @tab X
    @tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley
         playout servers.
+@item HNM @tab   @tab X
+    @tab Only version 4 supported, used in some games from Cryo Interactive
@item iCEDraw File              @tab   @tab X
@item ICO                       @tab X @tab X
    @tab Microsoft Windows ICO
@@ -285,6 +336,8 @@ library:
    @tab also known as DVB Transport Stream
@item MPEG-4                    @tab X @tab X
    @tab MPEG-4 is a variant of QuickTime.
+@item Mirillis FIC video        @tab   @tab X
+    @tab No cursor rendering.
@item MIME multipart JPEG       @tab X @tab
@item MSN TCP webcam            @tab   @tab X
    @tab Used by MSN Messenger webcam streams.
@@ -324,6 +377,7 @@ library:
@item raw H.261                 @tab X @tab X
@item raw H.263                 @tab X @tab X
@item raw H.264                 @tab X @tab X
+@item raw HEVC                  @tab X @tab X
@item raw Ingenient MJPEG       @tab   @tab X
@item raw MJPEG                 @tab X @tab X
@item raw MLP                   @tab   @tab X
@@ -474,6 +528,8 @@ following image formats are supported:
    @tab YUV, JPEG and some extension is not supported yet.
@item Truevision Targa  @tab X @tab X
    @tab Targa (.TGA) image format
+@item WebP         @tab E @tab X
+    @tab WebP image format, encoding supported through external library libwebp
@item XBM  @tab X @tab X
    @tab X BitMap image format
@item XFace @tab X @tab X
@@ -591,7 +647,9 @@ following image formats are supported:
@item H.263+ / H.263-1998 / H.263 version 2  @tab  X  @tab  X
@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10  @tab  E  @tab  X
    @tab encoding supported through external library libx264
-@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 (VDPAU acceleration)  @tab  E  @tab  X
+@item HEVC                   @tab  X  @tab  X
+    @tab encoding supported through the external library libx265
+@item HNM version 4          @tab     @tab  X
@item HuffYUV                @tab  X  @tab  X
@item HuffYUV FFmpeg variant @tab  X  @tab  X
@item IBM Ultimotion         @tab     @tab  X
@@ -643,8 +701,6 @@ following image formats are supported:
@item Mobotix MxPEG video    @tab     @tab  X
@item Motion Pixels video    @tab     @tab  X
@item MPEG-1 video           @tab  X  @tab  X
-@item MPEG-1/2 video XvMC (X-Video Motion Compensation)  @tab     @tab  X
-@item MPEG-1/2 video (VDPAU acceleration)  @tab     @tab  X
@item MPEG-2 video           @tab  X  @tab  X
@item MPEG-4 part 2          @tab  X  @tab  X
    @tab libxvidcore can be used alternatively for encoding.
@@ -662,6 +718,8 @@ following image formats are supported:
    @tab fourcc: VP60,VP61,VP62
@item VP8                    @tab  E  @tab  X
    @tab fourcc: VP80, encoding supported through external library libvpx
+@item VP9                    @tab  E  @tab  X
+    @tab encoding supported through external library libvpx
@item Pinnacle TARGA CineWave YUV16 @tab     @tab  X
    @tab fourcc: Y216
@item Prores                 @tab     @tab  X
@@ -811,8 +869,9 @@ following image formats are supported:
@item Amazing Studio PAF Audio @tab     @tab  X
@item Apple lossless audio   @tab  X  @tab  X
    @tab QuickTime fourcc 'alac'
-@item Atrac 1                @tab     @tab  X
-@item Atrac 3                @tab     @tab  X
+@item ATRAC1                 @tab     @tab  X
+@item ATRAC3                 @tab     @tab  X
+@item ATRAC3+                @tab     @tab  X
@item Bink Audio             @tab     @tab  X
    @tab Used in Bink and Smacker files in many games.
@item CELT                   @tab     @tab  E
@@ -922,8 +981,8 @@ 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  E  @tab  X
-    @tab supported through external library libwavpack
+@item Voxware MetaSound      @tab     @tab  X
+@item WavPack                @tab  X  @tab  X
@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
@@ -946,6 +1005,7 @@ performance on systems without hardware floating point support).
@item 3GPP Timed Text  @tab   @tab   @tab X @tab X
@item AQTitle          @tab   @tab X @tab   @tab X
@item DVB              @tab X @tab X @tab X @tab X
+@item DVB teletext     @tab   @tab X @tab   @tab E
@item DVD              @tab X @tab X @tab X @tab X
@item JACOsub          @tab X @tab X @tab   @tab X
@item MicroDVD         @tab X @tab X @tab   @tab X
@@ -968,11 +1028,14 @@ performance on systems without hardware floating point support).

@code{X} means that the feature is supported.

+@code{E} means that support is provided through an external library.
+
@section Network Protocols

@multitable @columnfractions .4 .1
@item Name         @tab Support
@item file         @tab X
+@item FTP          @tab X
@item Gopher       @tab X
@item HLS          @tab X
@item HTTP         @tab X
@@ -988,6 +1051,7 @@ performance on systems without hardware floating point support).
@item RTMPTS       @tab X
@item RTP          @tab X
@item SCTP         @tab X
+@item SFTP         @tab E
@item TCP          @tab X
@item TLS          @tab X
@item UDP          @tab X
@@ -1007,13 +1071,14 @@ performance on systems without hardware floating point support).
@item caca              @tab        @tab X
@item DV1394            @tab X      @tab
@item Lavfi virtual device @tab X   @tab
-@item Linux framebuffer @tab X      @tab
+@item Linux framebuffer @tab X      @tab X
@item JACK              @tab X      @tab
@item LIBCDIO           @tab X
@item LIBDC1394         @tab X      @tab
@item OpenAL            @tab X
+@item OpenGL            @tab        @tab X
@item OSS               @tab X      @tab X
-@item Pulseaudio        @tab X      @tab
+@item PulseAudio        @tab X      @tab X
@item SDL               @tab        @tab X
@item Video4Linux2      @tab X      @tab X
@item VfW capture       @tab X      @tab
--- a/doc/git-howto.texi
+++ b/doc/git-howto.texi
@@ -299,7 +299,7 @@ the current branch history.
 git commit --amend
@end example

-allows to amend the last commit details quickly.
+allows one to amend the last commit details quickly.

@example
 git rebase -i origin/master
--- a/doc/git-howto.txt
+++ b/doc/git-howto.txt
@@ -1,273 +0,0 @@
-
-About Git write access:
-~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
-
-Before everything else, you should know how to use GIT properly.
-Luckily Git comes with excellent documentation.
-
-  git --help
-  man git
-
-shows you the available subcommands,
-
-  git <command> --help
-  man git-<command>
-
-shows information about the subcommand <command>.
-
-The most comprehensive manual is the website Git Reference
-
-http://gitref.org/
-
-For more information about the Git project, visit
-
-http://git-scm.com/
-
-Consult these resources whenever you have problems, they are quite exhaustive.
-
-You do not need a special username or password.
-All you need is to provide a ssh public key to the Git server admin.
-
-What follows now is a basic introduction to Git and some FFmpeg-specific
-guidelines. Read it at least once, if you are granted commit privileges to the
-FFmpeg project you are expected to be familiar with these rules.
-
-
-
-I. BASICS:
-==========
-
-0. Get GIT:
-
-  Most distributions have a git package, if not
-  You can get git from http://git-scm.com/
-
-
-1. Cloning the source tree:
-
-    git clone git://source.ffmpeg.org/ffmpeg <target>
-
-  This will put the FFmpeg sources into the directory <target>.
-
-    git clone git@source.ffmpeg.org:ffmpeg <target>
-
-  This will put the FFmpeg sources into the directory <target> and let
-  you push back your changes to the remote repository.
-
-
-2. Updating the source tree to the latest revision:
-
-    git pull (--ff-only)
-
-  pulls in the latest changes from the tracked branch. The tracked branch
-  can be remote. By default the master branch tracks the branch master in
-  the remote origin.
-  Caveat: Since merge commits are forbidden at least for the initial
-          months of git --ff-only or --rebase (see below) are recommended.
-          --ff-only will fail and not create merge commits if your branch
-          has diverged (has a different history) from the tracked branch.
-
-2.a Rebasing your local branches:
-
-    git pull --rebase
-
-  fetches the changes from the main repository and replays your local commits
-  over it. This is required to keep all your local changes at the top of
-  FFmpeg's master tree. The master tree will reject pushes with merge commits.
-
-
-3. Adding/removing files/directories:
-
-    git add [-A] <filename/dirname>
-    git rm [-r] <filename/dirname>
-
-  GIT needs to get notified of all changes you make to your working
-  directory that makes files appear or disappear.
-  Line moves across files are automatically tracked.
-
-
-4. Showing modifications:
-
-    git diff <filename(s)>
-
-  will show all local modifications in your working directory as unified diff.
-
-
-5. Inspecting the changelog:
-
-    git log <filename(s)>
-
-  You may also use the graphical tools like gitview or gitk or the web
-  interface available at http://source.ffmpeg.org
-
-6. Checking source tree status:
-
-    git status
-
-  detects all the changes you made and lists what actions will be taken in case
-  of a commit (additions, modifications, deletions, etc.).
-
-
-7. Committing:
-
-    git diff --check
-
-  to double check your changes before committing them to avoid trouble later
-  on. All experienced developers do this on each and every commit, no matter
-  how small.
-  Every one of them has been saved from looking like a fool by this many times.
-  It's very easy for stray debug output or cosmetic modifications to slip in,
-  please avoid problems through this extra level of scrutiny.
-
-  For cosmetics-only commits you should get (almost) empty output from
-
-    git diff -w -b <filename(s)>
-
-  Also check the output of
-
-    git status
-
-  to make sure you don't have untracked files or deletions.
-
-    git add [-i|-p|-A] <filenames/dirnames>
-
-  Make sure you have told git your name and email address, e.g. by running
-    git config --global user.name "My Name"
-    git config --global user.email my@email.invalid
-  (--global to set the global configuration for all your git checkouts).
-
-  Git will select the changes to the files for commit. Optionally you can use
-  the interactive or the patch mode to select hunk by hunk what should be
-  added to the commit.
-
-    git commit
-
-  Git will commit the selected changes to your current local branch.
-
-  You will be prompted for a log message in an editor, which is either
-  set in your personal configuration file through
-
-    git config core.editor
-
-  or set by one of the following environment variables:
-  GIT_EDITOR, VISUAL or EDITOR.
-
-  Log messages should be concise but descriptive. Explain why you made a change,
-  what you did will be obvious from the changes themselves most of the time.
-  Saying just "bug fix" or "10l" is bad. Remember that people of varying skill
-  levels look at and educate themselves while reading through your code. Don't
-  include filenames in log messages, Git provides that information.
-
-  Possibly make the commit message have a terse, descriptive first line, an
-  empty line and then a full description. The first line will be used to name
-  the patch by git format-patch.
-
-
-8. Renaming/moving/copying files or contents of files:
-
-  Git automatically tracks such changes, making those normal commits.
-
-    mv/cp path/file otherpath/otherfile
-
-    git add [-A] .
-
-    git commit
-
-  Do not move, rename or copy files of which you are not the maintainer without
-  discussing it on the mailing list first!
-
-9. Reverting broken commits
-
-    git revert <commit>
-
-  git revert will generate a revert commit. This will not make the faulty
-  commit disappear from the history.
-
-    git reset <commit>
-
-  git reset will uncommit the changes till <commit> rewriting the current
-  branch history.
-
-    git commit --amend
-
-  allows to amend the last commit details quickly.
-
-    git rebase -i origin/master
-
-  will replay local commits over the main repository allowing to edit,
-  merge or remove some of them in the process.
-
-  Note that the reset, commit --amend and rebase rewrite history, so you
-  should use them ONLY on your local or topic branches.
-
-  The main repository will reject those changes.
-
-10. Preparing a patchset.
-
-    git format-patch <commit> [-o directory]
-
-  will generate a set of patches for each commit between <commit> and
-  current HEAD. E.g.
-
-    git format-patch origin/master
-
-  will generate patches for all commits on current branch which are not
-  present in upstream.
-  A useful shortcut is also
-
-    git format-patch -n
-
-  which will generate patches from last n commits.
-  By default the patches are created in the current directory.
-
-11. Sending patches for review
-
-    git send-email <commit list|directory>
-
-  will send the patches created by git format-patch or directly generates
-  them. All the email fields can be configured in the global/local
-  configuration or overridden by command line.
-  Note that this tool must often be installed separately (e.g. git-email
-  package on Debian-based distros).
-
-12. Pushing changes to remote trees
-
-    git push
-
-  Will push the changes to the default remote (origin).
-  Git will prevent you from pushing changes if the local and remote trees are
-  out of sync. Refer to 2 and 2.a to sync the local tree.
-
-    git remote add <name> <url>
-
-  Will add additional remote with a name reference, it is useful if you want
-  to push your local branch for review on a remote host.
-
-    git push <remote> <refspec>
-
-  Will push the changes to the remote repository. Omitting refspec makes git
-  push update all the remote branches matching the local ones.
-
-13. Finding a specific svn revision
-
-  Since version 1.7.1 git supports ':/foo' syntax for specifying commits
-  based on a regular expression. see man gitrevisions
-
-    git show :/'as revision 23456'
-
-  will show the svn changeset r23456. With older git versions searching in
-  the git log output is the easiest option (especially if a pager with
-  search capabilities is used).
-  This commit can be checked out with
-
-    git checkout -b svn_23456 :/'as revision 23456'
-
-  or for git < 1.7.1 with
-
-    git checkout -b svn_23456 $SHA1
-
-  where $SHA1 is the commit SHA1 from the 'git log' output.
-
-
-Contact the project admins <root at ffmpeg dot org> if you have technical
-problems with the GIT server.
--- a/doc/indevs.texi
+++ b/doc/indevs.texi
@@ -409,7 +409,7 @@ OpenAL is part of Core Audio, the official Mac OS X Audio interface.
 See @url{http://developer.apple.com/technologies/mac/audio-and-video.html}
@end table

-This device allows to capture from an audio input device handled
+This device allows one to capture from an audio input device handled
 through OpenAL.

 You need to specify the name of the device to capture in the provided
@@ -485,87 +485,52 @@ For more information about OSS see:

@section pulse

-pulseaudio input device.
+PulseAudio input device.

-To enable this input device during configuration you need libpulse-simple
-installed in your system.
+To enable this output device you need to configure FFmpeg with @code{--enable-libpulse}.

 The filename to provide to the input device is a source device or the
 string "default"

-To list the pulse source devices and their properties you can invoke
+To list the PulseAudio source devices and their properties you can invoke
 the command @command{pactl list sources}.

+More information about PulseAudio can be found on @url{http://www.pulseaudio.org}.
+
+@subsection Options
+@table @option
+@item server
+Connect to a specific PulseAudio server, specified by an IP address.
+Default server is used when not provided.
+
+@item name
+Specify the application name PulseAudio will use when showing active clients,
+by default it is the @code{LIBAVFORMAT_IDENT} string.
+
+@item stream_name
+Specify the stream name PulseAudio will use when showing active streams,
+by default it is "record".
+
+@item sample_rate
+Specify the samplerate in Hz, by default 48kHz is used.
+
+@item channels
+Specify the channels in use, by default 2 (stereo) is set.
+
+@item frame_size
+Specify the number of bytes per frame, by default it is set to 1024.
+
+@item fragment_size
+Specify the minimal buffering fragment in PulseAudio, it will affect the
+audio latency. By default it is unset.
+@end table
+
+@subsection Examples
+Record a stream from default device:
@example
 ffmpeg -f pulse -i default /tmp/pulse.wav
@end example

-@subsection @var{server} AVOption
-
-The syntax is:
-@example
-server @var{server name}
-@end example
-
-Connects to a specific server.
-
-@subsection @var{name} AVOption
-
-The syntax is:
-@example
-name @var{application name}
-@end example
-
-Specify the application name pulse will use when showing active clients,
-by default it is the LIBAVFORMAT_IDENT string
-
-@subsection @var{stream_name} AVOption
-
-The syntax is:
-@example
-stream_name @var{stream name}
-@end example
-
-Specify the stream name pulse will use when showing active streams,
-by default it is "record"
-
-@subsection @var{sample_rate} AVOption
-
-The syntax is:
-@example
-sample_rate @var{samplerate}
-@end example
-
-Specify the samplerate in Hz, by default 48kHz is used.
-
-@subsection @var{channels} AVOption
-
-The syntax is:
-@example
-channels @var{N}
-@end example
-
-Specify the channels in use, by default 2 (stereo) is set.
-
-@subsection @var{frame_size} AVOption
-
-The syntax is:
-@example
-frame_size @var{bytes}
-@end example
-
-Specify the number of byte per frame, by default it is set to 1024.
-
-@subsection @var{fragment_size} AVOption
-
-The syntax is:
-@example
-fragment_size @var{bytes}
-@end example
-
-Specify the minimal buffering fragment in pulseaudio, it will affect the
-audio latency. By default it is unset.
-
@section sndio

 sndio input device.
@@ -652,7 +617,7 @@ Select the pixel format (only valid for raw video input).

@item input_format
 Set the preferred pixel format (for raw video) or a codec name.
-This option allows to select the input format, when several are
+This option allows one to select the input format, when several are
 available.

@item framerate
@@ -713,7 +678,7 @@ other filename will be interpreted as device number 0.

 X11 video input device.

-This device allows to capture a region of an X11 display.
+This device allows one to capture a region of an X11 display.

 The filename passed as input has the syntax:
@example
@@ -736,12 +701,12 @@ properties of your X11 display (e.g. grep for "name" or "dimensions").

 For example to grab from @file{:0.0} using @command{ffmpeg}:
@example
-ffmpeg -f x11grab -r 25 -s cif -i :0.0 out.mpg
+ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0 out.mpg
@end example

 Grab at position @code{10,20}:
@example
-ffmpeg -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg
+ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
@end example

@subsection Options
@@ -762,12 +727,12 @@ zero) to the edge of region.

 For example:
@example
-ffmpeg -f x11grab -follow_mouse centered -r 25 -s cif -i :0.0 out.mpg
+ffmpeg -f x11grab -follow_mouse centered -framerate 25 -video_size cif -i :0.0 out.mpg
@end example

 To follow only when the mouse pointer reaches within 100 pixels to edge:
@example
-ffmpeg -f x11grab -follow_mouse 100 -r 25 -s cif -i :0.0 out.mpg
+ffmpeg -f x11grab -follow_mouse 100 -framerate 25 -video_size cif -i :0.0 out.mpg
@end example

@item framerate
@@ -783,12 +748,12 @@ know what is being grabbed if only a portion of the screen is grabbed.

 For example:
@example
-ffmpeg -f x11grab -show_region 1 -r 25 -s cif -i :0.0+10,20 out.mpg
+ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
@end example

 With @var{follow_mouse}:
@example
-ffmpeg -f x11grab -follow_mouse centered -show_region 1 -r 25 -s cif -i :0.0 out.mpg
+ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size cif -i :0.0 out.mpg
@end example

@item video_size
--- a/doc/issue_tracker.txt
+++ b/doc/issue_tracker.txt
@@ -1,4 +1,4 @@
-FFmpeg's bug/patch/feature request tracker manual
+FFmpeg's bug/feature request tracker manual
 =================================================

 NOTE: This is a draft.
@@ -11,7 +11,7 @@ existing issues can be done through a web interface.

 Issues can be different kinds of things we want to keep track of
 but that do not belong into the source tree itself. This includes
-bug reports, patches, feature requests and license violations. We
+bug reports, feature requests and license violations. We
 might add more items to this list in the future, so feel free to
 propose a new `type of issue' on the ffmpeg-devel mailing list if
 you feel it is worth tracking.
@@ -28,6 +28,9 @@ http(s)://trac.ffmpeg.org

 Type:
 -----
+art
+    Artwork such as photos, music, banners, and logos.
+
 bug / defect
    An error, flaw, mistake, failure, or fault in FFmpeg or libav* that
    prevents it from behaving as intended.
@@ -41,20 +44,18 @@ feature request / enhancement
 license violation
    ticket to keep track of (L)GPL violations of ffmpeg by others

-patch
-    A patch as generated by diff which conforms to the patch submission and
-    development policy.
-
+sponsoring request
+    Developer requests for hardware, software, specifications, money,
+    refunds, etc.

 Priority:
 ---------
 critical
-    Bugs and patches which deal with data loss and security issues.
+    Bugs about data loss and security issues.
    No feature request can be critical.

 important
-    Bugs which make FFmpeg unusable for a significant number of users, and
-    patches fixing them.
+    Bugs which make FFmpeg unusable for a significant number of users.
    Examples here might be completely broken MPEG-4 decoding or a build issue
    on Linux.
    While broken 4xm decoding or a broken OS/2 build would not be important,
@@ -68,7 +69,7 @@ normal


 minor
-    Bugs and patches about things like spelling errors, "mp2" instead of
+    Bugs about things like spelling errors, "mp2" instead of
    "mp3" being shown and such.
    Feature requests about things few people want or which do not make a big
    difference.
@@ -103,13 +104,13 @@ This state implicates that the bug either has been reproduced or that
 reproduction is not needed as the bug is already understood.


-Type/Status/Substatus:
+Type/Status:
 ----------
-*/new/new
-    Initial state of new bugs, patches and feature requests submitted by
+*/new
+    Initial state of new bugs and feature requests submitted by
    users.

-*/open/open
+*/open
    Issues which have been briefly looked at and which did not look outright
    invalid.
    This implicates that no real more detailed state applies yet. Conversely,
@@ -117,9 +118,7 @@ Type/Status/Substatus:
    looked at.

 */closed/duplicate
-    Bugs, patches or feature requests which are duplicates.
-    Note that patches dealing with the same thing in a different way are not
-    duplicates.
+    Bugs or feature requests which are duplicates.
    Note, if you mark something as duplicate, do not forget setting the
    superseder so bug reports are properly linked.

@@ -134,7 +133,7 @@ Type/Status/Substatus:
 bug/closed/fixed
    Bugs which have to the best of our knowledge been fixed.

-bug/closed/wont_fix
+bug/closed/wontfix
    Bugs which we will not fix. Possible reasons include legality, high
    complexity for the sake of supporting obscure corner cases, speed loss
    for similarly esoteric purposes, et cetera.
@@ -148,33 +147,15 @@ bug/closed/works_for_me
    reproduction failed - that is the code seems to work correctly to the
    best of our knowledge.

-patch/open/approved
-    Patches which have been reviewed and approved by a developer.
-    Such patches can be applied anytime by any other developer after some
-    reasonable testing (compile + regression tests + does the patch do
-    what the author claimed).
-
-patch/open/needs_changes
-    Patches which have been reviewed and need changes to be accepted.
-
-patch/closed/applied
-    Patches which have been applied.
-
-patch/closed/rejected
-    Patches which have been rejected.
-
-feature_request/closed/implemented
+feature_request/closed/fixed
    Feature requests which have been implemented.

-feature_request/closed/wont_implement
+feature_request/closed/wontfix
    Feature requests which will not be implemented. The reasons here could
    be legal, philosophical or others.

-Note, please do not use type-status-substatus combinations other than the
-above without asking on ffmpeg-dev first!
-
 Note2, if you provide the requested info do not forget to remove the
-needs_more_info substatus.
+needs_more_info resolution.

 Component:
 ----------
--- a/doc/libavutil.texi
+++ b/doc/libavutil.texi
@@ -16,7 +16,25 @@ The libavutil library is a utility library to aid portable
 multimedia programming. It contains safe portable string functions,
 random number generators, data structures, additional mathematics
 functions, cryptography and multimedia related functionality (like
-enumerations for pixel and sample formats).
+enumerations for pixel and sample formats). It is not a library for
+code needed by both libavcodec and libavformat.
+
+The goals for this library is to be:
+
+@table @strong
+@item Modular
+It should have few interdependencies and the possibility of disabling individual
+parts during @command{./configure}.
+
+@item Small
+Both sources and objects should be small.
+
+@item Efficient
+It should have low CPU and memory usage.
+
+@item Useful
+It should avoid useless features that almost no one needs.
+@end table

@c man end DESCRIPTION

--- a/doc/libswresample.texi
+++ b/doc/libswresample.texi
@@ -20,7 +20,7 @@ Specifically, this library performs the following conversions:
@itemize
@item
@emph{Resampling}: is the process of changing the audio rate, for
-example from an high sample rate of 44100Hz to 8000Hz. Audio
+example from a high sample rate of 44100Hz to 8000Hz. Audio
 conversion from high to low sample rate is a lossy process. Several
 resampling options and algorithms are available.

--- a/doc/mips.txt
+++ b/doc/mips.txt
@@ -47,6 +47,11 @@ Files that have MIPS copyright notice in them:
 * libavutil/mips/
      float_dsp_mips.c
      libm_mips.h
+* libavcodec/
+      fft_fixed_32.c
+      fft_init_table.c
+      fft_table.h
+      mdct_fixed_32.c
 * libavcodec/mips/
      aaccoder_mips.c
      aacpsy_mips.h
--- a/doc/muxers.texi
+++ b/doc/muxers.texi
@@ -23,6 +23,8 @@ A description of some of the currently available muxers follows.

 Audio Interchange File Format muxer.

+@subsection Options
+
 It accepts the following options:

@table @option
@@ -49,6 +51,10 @@ The output of the muxer consists of a single line of the form:
 CRC=0x@var{CRC}, where @var{CRC} is a hexadecimal number 0-padded to
 8 digits containing the CRC for all the decoded input frames.

+See also the @ref{framecrc} muxer.
+
+@subsection Examples
+
 For example to compute the CRC of the input, and store it in the file
@file{out.crc}:
@example
@@ -68,8 +74,6 @@ and the input video converted to MPEG-2 video, use the command:
 ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc -
@end example

-See also the @ref{framecrc} muxer.
-
@anchor{framecrc}
@section framecrc

@@ -89,6 +93,8 @@ packet of the form:
@var{CRC} is a hexadecimal number 0-padded to 8 digits containing the
 CRC of the packet.

+@subsection Examples
+
 For example to compute the CRC of the audio and video frames in
@file{INPUT}, converted to raw audio and video packets, and store it
 in the file @file{out.crc}:
@@ -132,6 +138,8 @@ packet of the form:
@var{MD5} is a hexadecimal number representing the computed MD5 hash
 for the packet.

+@subsection Examples
+
 For example to compute the MD5 of the audio and video frames in
@file{INPUT}, converted to raw audio and video packets, and store it
 in the file @file{out.md5}:
@@ -146,30 +154,89 @@ ffmpeg -i INPUT -f framemd5 -

 See also the @ref{md5} muxer.

+@anchor{gif}
+@section gif
+
+Animated GIF muxer.
+
+It accepts the following options:
+
+@table @option
+@item loop
+Set the number of times to loop the output. Use @code{-1} for no loop, @code{0}
+for looping indefinitely (default).
+
+@item final_delay
+Force the delay (expressed in centiseconds) after the last frame. Each frame
+ends with a delay until the next frame. The default is @code{-1}, which is a
+special value to tell the muxer to re-use the previous delay. In case of a
+loop, you might want to customize this value to mark a pause for instance.
+@end table
+
+For example, to encode a gif looping 10 times, with a 5 seconds delay between
+the loops:
+@example
+ffmpeg -i INPUT -loop 10 -final_delay 500 out.gif
+@end example
+
+Note 1: if you wish to extract the frames in separate GIF files, you need to
+force the @ref{image2} muxer:
+@example
+ffmpeg -i INPUT -c:v gif -f image2 "out%d.gif"
+@end example
+
+Note 2: the GIF format has a very small time base: the delay between two frames
+can not be smaller than one centi second.
+
@anchor{hls}
@section hls

 Apple HTTP Live Streaming muxer that segments MPEG-TS according to
-the HTTP Live Streaming specification.
+the HTTP Live Streaming (HLS) specification.

 It creates a playlist file and numbered segment files. The output
 filename specifies the playlist filename; the segment filenames
 receive the same basename as the playlist, a sequential number and
 a .ts extension.

+For example, to convert an input file with @command{ffmpeg}:
@example
 ffmpeg -i in.nut out.m3u8
@end example

+See also the @ref{segment} muxer, which provides a more generic and
+flexible implementation of a segmenter, and can be used to perform HLS
+segmentation.
+
+@subsection Options
+
+This muxer supports the following options:
+
@table @option
-@item -hls_time @var{seconds}
-Set the segment length in seconds.
-@item -hls_list_size @var{size}
-Set the maximum number of playlist entries.
-@item -hls_wrap @var{wrap}
-Set the number after which index wraps.
-@item -start_number @var{number}
-Start the sequence from @var{number}.
+@item hls_time @var{seconds}
+Set the segment length in seconds. Default value is 2.
+
+@item hls_list_size @var{size}
+Set the maximum number of playlist entries. If set to 0 the list file
+will contain all the segments. Default value is 5.
+
+@item hls_wrap @var{wrap}
+Set the number after which the segment filename number (the number
+specified in each segment file) wraps. If set to 0 the number will be
+never wrapped. Default value is 0.
+
+This option is useful to avoid to fill the disk with many segment
+files, and limits the maximum number of segment files written to disk
+to @var{wrap}.
+
+@item start_number @var{number}
+Start the playlist sequence number from @var{number}. Default value is
+0.
+
+Note that the playlist sequence number must be unique for each segment
+and it is not to be confused with the segment filename sequence number
+which can be cyclic, for example if the @option{wrap} option is
+specified.
@end table

@anchor{ico}
@@ -235,6 +302,8 @@ The pattern "img%%-%d.jpg" will specify a sequence of filenames of the
 form @file{img%-1.jpg}, @file{img%-2.jpg}, ..., @file{img%-10.jpg},
 etc.

+@subsection Examples
+
 The following example shows how to use @command{ffmpeg} for creating a
 sequence of files @file{img-001.jpeg}, @file{img-002.jpeg}, ...,
 taking one image every second from the input video:
@@ -257,16 +326,32 @@ Note also that the pattern must not necessarily contain "%d" or
 ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg
@end example

+The @option{strftime} option allows you to expand the filename with
+date and time information. Check the documentation of
+the @code{strftime()} function for the syntax.
+
+For example to generate image files from the @code{strftime()}
+"%Y-%m-%d_%H-%M-%S" pattern, the following @command{ffmpeg} command
+can be used:
+@example
+ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg"
+@end example
+
+@subsection Options
+
@table @option
-@item start_number @var{number}
-Start the sequence from @var{number}. Default value is 1. Must be a
-positive number.
+@item start_number
+Start the sequence from the specified number. Default value is 1. Must
+be a non-negative number.

-@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.
+@item update
+If set to 1, the filename will always be interpreted as just a
+filename, not a pattern, and the corresponding file will be continuously
+overwritten with new images. Default value is 0.

+@item strftime
+If set to 1, expand the filename with date and time information from
+@code{strftime()}. Default value is 0.
@end table

 The image muxer supports the .Y.U.V image file format. This format is
@@ -281,25 +366,27 @@ Matroska container muxer.

 This muxer implements the matroska and webm container specs.

+@subsection Metadata
+
 The recognized metadata settings in this muxer are:

@table @option
+@item title
+Set title name provided to a single track.

-@item title=@var{title name}
-Name provided to a single track
-@end table
+@item language
+Specify the language of the track in the Matroska languages form.

-@table @option
+The language can be either the 3 letters bibliographic ISO-639-2 (ISO
+639-2/B) form (like "fre" for French), or a language code mixed with a
+country code for specialities in languages (like "fre-ca" for Canadian
+French).

-@item language=@var{language name}
-Specifies the language of the track in the Matroska languages form
-@end table
+@item stereo_mode
+Set stereo 3D video layout of two views in a single video track.

-@table @option
-
-@item stereo_mode=@var{mode}
-Stereo 3D video layout of two views in a single video track
-@table @option
+The following values are recognized:
+@table @samp
@item mono
 video is not stereo
@item left_right
@@ -338,10 +425,11 @@ For example a 3D WebM clip can be created using the following command line:
 ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm
@end example

+@subsection Options
+
 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
@@ -356,7 +444,6 @@ 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}
@@ -386,7 +473,9 @@ ffmpeg -i INPUT -f md5 -

 See also the @ref{framemd5} muxer.

-@section MOV/MP4/ISMV
+@section mov, mp4, ismv
+
+MOV/MP4/ISMV (Smooth Streaming) muxer.

 The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4
 file has all the metadata about all packets stored in one location
@@ -402,6 +491,8 @@ very long files (since writing normal MOV/MP4 files stores info about
 every single packet in memory until the file is closed). The downside
 is that it is less compatible with other applications.

+@subsection Options
+
 Fragmentation is enabled by setting one of the AVOptions that define
 how to cut the file into fragments:

@@ -451,13 +542,15 @@ pair for each track, making it easier to separate tracks.

 This option is implicitly set when writing ismv (Smooth Streaming) files.
@item -movflags faststart
-Run a second pass moving the moov atom on top of the file. This
-operation can take a while, and will not work in various situations such
+Run a second pass moving the index (moov atom) to the beginning of the file.
+This operation can take a while, and will not work in various situations such
 as fragmented output, thus it is not enabled by default.
@item -movflags rtphint
 Add RTP hinting tracks to the output file.
@end table

+@subsection Example
+
 Smooth Streaming content can be pushed in real time to a publishing
 point on IIS with this muxer. Example:
@example
@@ -468,12 +561,15 @@ ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe

 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.
+@code{id3v2_version} option controls which one is used. Setting
+@code{id3v2_version} to 0 will disable the ID3v2 header completely. The legacy
+ID3v1 tag is not written by default, but may be enabled with the
+@code{write_id3v1} option.

-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 may also write a Xing frame at the beginning, which contains the
+number of frames in the file. It is useful for computing duration of VBR files.
+The Xing frame is written if the output stream is seekable and if the
+@code{write_xing} option is set to 1 (the default).

 The muxer supports writing ID3v2 attached pictures (APIC frames). The pictures
 are supplied to the muxer in form of a video stream with a single packet. There
@@ -500,12 +596,24 @@ 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

+Write a "clean" MP3 without any extra features:
+@example
+ffmpeg -i input.wav -write_xing 0 -id3v2_version 0 out.mp3
+@end example
+
@section mpegts

 MPEG transport stream muxer.

 This muxer implements ISO 13818-1 and part of ETSI EN 300 468.

+The recognized metadata settings in mpegts muxer are @code{service_provider}
+and @code{service_name}. If they are not set the default for
+@code{service_provider} is "FFmpeg" and the default for
+@code{service_name} is "Service01".
+
+@subsection Options
+
 The muxer options are:

@table @option
@@ -522,12 +630,43 @@ Set the service_id (default 0x0001) also known as program in DVB.
 Set the first PID for PMT (default 0x1000, max 0x1f00).
@item -mpegts_start_pid @var{number}
 Set the first PID for data packets (default 0x0100, max 0x0f00).
+@item -mpegts_m2ts_mode @var{number}
+Enable m2ts mode if set to 1. Default value is -1 which disables m2ts mode.
+@item -muxrate @var{number}
+Set muxrate.
+@item -pes_payload_size @var{number}
+Set minimum PES packet payload in bytes.
+@item -mpegts_flags @var{flags}
+Set flags (see below).
+@item -mpegts_copyts @var{number}
+Preserve original timestamps, if value is set to 1. Default value is -1, which
+results in shifting timestamps so that they start from 0.
+@item -tables_version @var{number}
+Set PAT, PMT and SDT version (default 0, valid values are from 0 to 31, inclusively).
+This option allows updating stream structure so that standard consumer may
+detect the change. To do so, reopen output AVFormatContext (in case of API
+usage) or restart ffmpeg instance, cyclically changing tables_version value:
+@example
+ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
+ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
+...
+ffmpeg -i source3.ts -codec copy -f mpegts -tables_version 31 udp://1.1.1.1:1111
+ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
+ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
+...
+@end example
@end table

-The recognized metadata settings in mpegts muxer are @code{service_provider}
-and @code{service_name}. If they are not set the default for
-@code{service_provider} is "FFmpeg" and the default for
-@code{service_name} is "Service01".
+Option mpegts_flags may take a set of such flags:
+
+@table @option
+@item resend_headers
+Reemit PAT/PMT before writing the next packet.
+@item latm
+Use LATM packetization for AAC.
+@end table
+
+@subsection Example

@example
 ffmpeg -i file.mpg -c copy \
@@ -578,11 +717,12 @@ situations, giving a small seek granularity at the cost of additional container
 overhead.
@end table

+@anchor{segment}
@section segment, stream_segment, ssegment

 Basic stream segmenter.

-The segmenter muxer outputs streams to a number of separate files of nearly
+This muxer outputs streams to a number of separate files of nearly
 fixed duration. Output filename pattern can be set in a fashion similar to
@ref{image2}.

@@ -604,14 +744,21 @@ The segment muxer works best with a single constant frame rate video.

 Optionally it can generate a list of the created segments, by setting
 the option @var{segment_list}. The list type is specified by the
-@var{segment_list_type} option.
+@var{segment_list_type} option. The entry filenames in the segment
+list are set by default to the basename of the corresponding segment
+files.
+
+See also the @ref{hls} muxer, which provides a more specific
+implementation for HLS segmentation.
+
+@subsection Options

 The segment muxer supports the following options:

@table @option
@item reference_stream @var{specifier}
 Set the reference stream, as specified by the string @var{specifier}.
-If @var{specifier} is set to @code{auto}, the reference is choosen
+If @var{specifier} is set to @code{auto}, the reference is chosen
 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 @code{auto}.
@@ -636,13 +783,15 @@ Allow caching (only affects M3U8 list files).
 Allow live-friendly file generation.
@end table

-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_entry_prefix @var{prefix}
+Set @var{prefix} to prepend to the name of each entry filename. By
+default no prefix is applied.
+
@item segment_list_type @var{type}
 Specify the format for the segment list file.

@@ -714,7 +863,7 @@ In particular may be used in combination with the @file{ffmpeg} option
@var{force_key_frames} may not be set accurately because of rounding
 issues, with the consequence that a key frame time may result set just
 before the specified time. For constant frame rate videos a value of
-1/2*@var{frame_rate} should address the worst case mismatch between
+1/(2*@var{frame_rate}) should address the worst case mismatch between
 the specified time and the time set by @var{force_key_frames}.

@item segment_times @var{times}
@@ -741,6 +890,10 @@ Reset timestamps at the begin of each segment, so that each segment
 will start with near-zero timestamps. It is meant to ease the playback
 of the generated segments. May not work with some combinations of
 muxers/codecs. It is set to @code{0} by default.
+
+@item initial_offset @var{offset}
+Specify timestamp offset to apply to the output packet timestamps. The
+argument must be a time duration specification, and defaults to 0.
@end table

@subsection Examples
@@ -811,22 +964,71 @@ to feed the same packets to several muxers directly.
 The slave outputs are specified in the file name given to the muxer,
 separated by '|'. If any of the slave name contains the '|' separator,
 leading or trailing spaces or any special character, it must be
-escaped (see the ``Quoting and escaping'' section in the ffmpeg-utils
-manual).
+escaped (see @ref{quoting_and_escaping,,the "Quoting and escaping"
+section in the ffmpeg-utils(1) manual,ffmpeg-utils}).

-Options can be specified for each slave by prepending them as a list of
+Muxer options can be specified for each slave by prepending them as a list of
@var{key}=@var{value} pairs separated by ':', between square brackets. If
 the options values contain a special character or the ':' separator, they
 must be escaped; note that this is a second level escaping.

-Example: encode something and both archive it in a WebM file and stream it
-as MPEG-TS over UDP (the streams need to be explicitly mapped):
+The following special options are also recognized:
+@table @option
+@item f
+Specify the format name. Useful if it cannot be guessed from the
+output name suffix.

+@item bsfs[/@var{spec}]
+Specify a list of bitstream filters to apply to the specified
+output.
+
+It is possible to specify to which streams a given bitstream filter
+applies, by appending a stream specifier to the option separated by
+@code{/}. @var{spec} must be a stream specifier (see @ref{Format
+stream specifiers}).  If the stream specifier is not specified, the
+bistream filters will be applied to all streams in the output.
+
+Several bitstream filters can be specified, separated by ",".
+
+@item select
+Select the streams that should be mapped to the slave output,
+specified by a stream specifier. If not specified, this defaults to
+all the input streams.
+@end table
+
+@subsection Examples
+
+@itemize
+@item
+Encode something and both archive it in a WebM file and stream it
+as MPEG-TS over UDP (the streams need to be explicitly mapped):
@example
 ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
  "archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"
@end example

+@item
+Use @command{ffmpeg} to encode the input, and send the output
+to three different destinations. The @code{dump_extra} bitstream
+filter is used to add extradata information to all the output video
+keyframes packets, as requested by the MPEG-TS format. The select
+option is applied to @file{out.aac} in order to make it contain only
+audio packets.
+@example
+ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental
+       -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
+@end example
+
+@item
+As below, but select only stream @code{a:1} for the audio output. Note
+that a second level escaping must be performed, as ":" is a special
+character used to separate options.
+@example
+ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental
+       -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"
+@end example
+@end itemize
+
 Note: some codecs may need different options depending on the output format;
 the auto-detection of this can not work with the tee muxer. The main example
 is the @option{global_header} flag.
--- a/doc/outdevs.texi
+++ b/doc/outdevs.texi
@@ -1,7 +1,7 @@
@chapter Output Devices
@c man begin OUTPUT DEVICES

-Output devices are configured elements in FFmpeg which allow to write
+Output devices are configured elements in FFmpeg that can write
 multimedia data to an output device attached to your system.

 When you configure your FFmpeg build, all the supported output devices
@@ -22,11 +22,27 @@ A description of the currently available output devices follows.

 ALSA (Advanced Linux Sound Architecture) output device.

+@subsection Examples
+
+@itemize
+@item
+Play a file on default ALSA device:
+@example
+ffmpeg -i INPUT -f alsa default
+@end example
+
+@item
+Play a file on soundcard 1, audio device 7:
+@example
+ffmpeg -i INPUT -f alsa hw:1,7
+@end example
+@end itemize
+
@section caca

 CACA output device.

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

@@ -104,15 +120,189 @@ ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors -
@end example
@end itemize

+@section decklink
+
+The decklink output device provides playback capabilities for Blackmagic
+DeckLink devices.
+
+To enable this output device, you need the Blackmagic DeckLink SDK and you
+need to configure with the appropriate @code{--extra-cflags}
+and @code{--extra-ldflags}.
+On Windows, you need to run the IDL files through @command{widl}.
+
+DeckLink is very picky about the formats it supports. Pixel format is always
+uyvy422, framerate and video size must be determined for your device with
+@command{-list_formats 1}. Audio sample rate is always 48 kHz.
+
+@subsection Options
+
+@table @option
+
+@item list_devices
+If set to @option{true}, print a list of devices and exit.
+Defaults to @option{false}.
+
+@item list_formats
+If set to @option{true}, print a list of supported formats and exit.
+Defaults to @option{false}.
+
+@item preroll
+Amount of time to preroll video in seconds.
+Defaults to @option{0.5}.
+
+@end table
+
+@subsection Examples
+
+@itemize
+
+@item
+List output devices:
+@example
+ffmpeg -i test.avi -f decklink -list_devices 1 dummy
+@end example
+
+@item
+List supported formats:
+@example
+ffmpeg -i test.avi -f decklink -list_formats 1 'DeckLink Mini Monitor'
+@end example
+
+@item
+Play video clip:
+@example
+ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 'DeckLink Mini Monitor'
+@end example
+
+@item
+Play video clip with non-standard framerate or video size:
+@example
+ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLink Mini Monitor'
+@end example
+
+@end itemize
+
+@section fbdev
+
+Linux framebuffer output device.
+
+The Linux framebuffer is a graphic hardware-independent abstraction
+layer to show graphics on a computer monitor, typically on the
+console. It is accessed through a file device node, usually
+@file{/dev/fb0}.
+
+For more detailed information read the file
+@file{Documentation/fb/framebuffer.txt} included in the Linux source tree.
+
+@subsection Options
+@table @option
+
+@item xoffset
+@item yoffset
+Set x/y coordinate of top left corner. Default is 0.
+@end table
+
+@subsection Examples
+Play a file on framebuffer device @file{/dev/fb0}.
+Required pixel format depends on current framebuffer settings.
+@example
+ffmpeg -re -i INPUT -vcodec rawvideo -pix_fmt bgra -f fbdev /dev/fb0
+@end example
+
+See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
+
+@section opengl
+OpenGL output device.
+
+To enable this output device you need to configure FFmpeg with @code{--enable-opengl}.
+
+This output device allows one to render to OpenGL context.
+Context may be provided by application or default SDL window is created.
+
+When device renders to external context, application must implement handlers for following messages:
+@code{AV_CTL_MESSAGE_CREATE_WINDOW_BUFFER} - create OpenGL context on current thread.
+@code{AV_CTL_MESSAGE_PREPARE_WINDOW_BUFFER} - make OpenGL context current.
+@code{AV_CTL_MESSAGE_DISPLAY_WINDOW_BUFFER} - swap buffers.
+@code{AV_CTL_MESSAGE_DESTROY_WINDOW_BUFFER} - destroy OpenGL context.
+Application is also required to inform a device about current resolution by sending @code{AV_DEVICE_WINDOW_RESIZED} message.
+
+@subsection Options
+@table @option
+
+@item background
+Set background color. Black is a default.
+@item no_window
+Disables default SDL window when set to non-zero value.
+Application must provide OpenGL context and both @code{window_size_cb} and @code{window_swap_buffers_cb} callbacks when set.
+@item window_title
+Set the SDL window title, if not specified default to the filename specified for the output device.
+Ignored when @option{no_window} is set.
+
+@end table
+
+@subsection Examples
+Play a file on SDL window using OpenGL rendering:
+@example
+ffmpeg  -i INPUT -f opengl "window title"
+@end example
+
@section oss

 OSS (Open Sound System) output device.

+@section pulse
+
+PulseAudio output device.
+
+To enable this output device you need to configure FFmpeg with @code{--enable-libpulse}.
+
+More information about PulseAudio can be found on @url{http://www.pulseaudio.org}
+
+@subsection Options
+@table @option
+
+@item server
+Connect to a specific PulseAudio server, specified by an IP address.
+Default server is used when not provided.
+
+@item name
+Specify the application name PulseAudio will use when showing active clients,
+by default it is the @code{LIBAVFORMAT_IDENT} string.
+
+@item stream_name
+Specify the stream name PulseAudio will use when showing active streams,
+by default it is set to the specified output name.
+
+@item device
+Specify the device to use. Default device is used when not provided.
+List of output devices can be obtained with command @command{pactl list sinks}.
+
+@item buffer_size
+@item buffer_duration
+Control the size and duration of the PulseAudio buffer. A small buffer
+gives more control, but requires more frequent updates.
+
+@option{buffer_size} specifies size in bytes while
+@option{buffer_duration} specifies duration in milliseconds.
+
+When both options are provided then the highest value is used
+(duration is recalculated to bytes using stream parameters). If they
+are set to 0 (which is default), the device will use the default
+PulseAudio duration value. By default PulseAudio set buffer duration
+to around 2 seconds.
+@end table
+
+@subsection Examples
+Play a file on default device on default server:
+@example
+ffmpeg  -i INPUT -f pulse "stream name"
+@end example
+
@section sdl

 SDL (Simple DirectMedia Layer) output device.

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

@@ -139,6 +329,20 @@ Set the SDL window size, can be a string of the form
@var{width}x@var{height} or a video size abbreviation.
 If not specified it defaults to the size of the input video,
 downscaled according to the aspect ratio.
+
+@item window_fullscreen
+Set fullscreen mode when non-zero value is provided.
+Default value is zero.
+@end table
+
+@subsection Interactive commands
+
+The window created by the device can be controlled through the
+following interactive commands.
+
+@table @key
+@item q, ESC
+Quit the device immediately.
@end table

@subsection Examples
@@ -157,7 +361,7 @@ sndio audio output device.

 XV (XVideo) output device.

-This output device allows to show a video stream in a X Window System
+This output device allows one to show a video stream in a X Window System
 window.

@subsection Options
--- a/doc/platform.texi
+++ b/doc/platform.texi
@@ -51,8 +51,9 @@ The toolchain provided with Xcode is sufficient to build the basic
 unacelerated code.

 Mac OS X on PowerPC or ARM (iPhone) requires a preprocessor from
+@url{https://github.com/FFmpeg/gas-preprocessor} or
@url{http://github.com/yuvi/gas-preprocessor} to build the optimized
-assembler functions. Just download the Perl script and put it somewhere
+assembler functions. Put the Perl script somewhere
 in your PATH, FFmpeg's configure will pick it up automatically.

 Mac OS X on amd64 and x86 requires @command{yasm} to build most of the
@@ -108,14 +109,16 @@ libavformat) as DLLs.

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

-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.
+FFmpeg can be built with MSVC 2012 or earlier using a C99-to-C89 conversion utility
+and wrapper, or with MSVC 2013 and ICL natively.

 You will need the following prerequisites:

@itemize
-@item @uref{http://download.videolan.org/pub/contrib/c99-to-c89/, C99-to-C89 Converter & Wrapper}
+@item @uref{https://github.com/libav/c99-to-c89/, C99-to-C89 Converter & Wrapper}
+(if using MSVC 2012 or earlier)
@item @uref{http://code.google.com/p/msinttypes/, msinttypes}
+(if using MSVC 2012 or earlier)
@item @uref{http://www.mingw.org/, MSYS}
@item @uref{http://yasm.tortall.net/, YASM}
@item @uref{http://gnuwin32.sourceforge.net/packages/bc.htm, bc for Windows} if
@@ -125,14 +128,16 @@ you want to run @uref{fate.html, FATE}.
 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}.
+Place @code{yasm.exe} somewhere in your @code{PATH}. If using MSVC 2012 or
+earlier, place @code{c99wrap.exe} and @code{c99conv.exe} somewhere in your
+@code{PATH} as well.

-Next, make sure @code{inttypes.h} and any other headers and libs you want to use
-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.
+Next, make sure any other headers and libs you want to use, such as zlib, are
+located in a spot that the compiler can see. Do so by modifying the @code{LIB}
+and @code{INCLUDE} environment variables to include the @strong{Windows-style}
+paths to these directories. Alternatively, you can try and use the
+@code{--extra-cflags}/@code{--extra-ldflags} configure options. If using MSVC
+2012 or earlier, place @code{inttypes.h} somewhere the compiler can see too.

 Finally, run:

@@ -182,7 +187,9 @@ can see.
@itemize
@item Visual Studio 2010 Pro and Express
@item Visual Studio 2012 Pro and Express
+@item Visual Studio 2013 Pro and Express
@item Intel Composer XE 2013
+@item Intel Composer XE 2013 SP1
@end itemize
 Anything else is not officially supported.

--- a/doc/print_options.c
+++ b/doc/print_options.c
@@ -1,20 +1,20 @@
 /*
 * Copyright (c) 2012 Anton Khirnov
 *
- * 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
 */

@@ -27,7 +27,9 @@
 #include <float.h>

 #include "libavformat/avformat.h"
+#include "libavformat/options_table.h"
 #include "libavcodec/avcodec.h"
+#include "libavcodec/options_table.h"
 #include "libavutil/opt.h"

 static void print_usage(void)
@@ -96,18 +98,14 @@ static void show_opts(const AVOption *opts, int per_stream)

 static void show_format_opts(void)
 {
-#include "libavformat/options_table.h"
-
    printf("@section Format AVOptions\n");
-    show_opts(options, 0);
+    show_opts(avformat_options, 0);
 }

 static void show_codec_opts(void)
 {
-#include "libavcodec/options_table.h"
-
    printf("@section Codec AVOptions\n");
-    show_opts(options, 1);
+    show_opts(avcodec_options, 1);
 }

 int main(int argc, char **argv)
--- a/doc/protocols.texi
+++ b/doc/protocols.texi
@@ -1,8 +1,8 @@
@chapter Protocols
@c man begin PROTOCOLS

-Protocols are configured elements in FFmpeg which allow to access
-resources which require the use of a particular protocol.
+Protocols are configured elements in FFmpeg that enable access to
+resources that require specific protocols.

 When you configure your FFmpeg build, all the supported protocols are
 enabled by default. You can list all available ones using the
@@ -117,7 +117,19 @@ ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP/////////////

 File access protocol.

-Allow to read from or read to a file.
+Allow to read from or write to a file.
+
+A file URL can have the form:
+@example
+file:@var{filename}
+@end example
+
+where @var{filename} is the path of the file to read.
+
+An URL that does not have a protocol prefix will be assumed to be a
+file URL. Depending on the build, an URL that looks like a Windows
+path with the drive letter at the beginning will also be assumed to be
+a file URL (usually not the case in builds for unix-like systems).

 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
 use the command:
@@ -125,9 +137,19 @@ use the command:
 ffmpeg -i file:input.mpeg output.mpeg
@end example

-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".
+This protocol accepts the following options:
+
+@table @option
+@item truncate
+Truncate existing files on write, if set to 1. A value of 0 prevents
+truncating. Default value is 1.
+
+@item blocksize
+Set I/O operation maximum block size, in bytes. Default value is
+@code{INT_MAX}, which results in not limiting the requested block size.
+Setting this value reasonably low improves user termination request reaction
+time, which is valuable for files on slow medium.
+@end table

@section ftp

@@ -191,7 +213,7 @@ m3u8 files.

 HTTP (Hyper Text Transfer Protocol).

-This protocol accepts the following options.
+This protocol accepts the following options:

@table @option
@item seekable
@@ -201,32 +223,33 @@ if set to -1 it will try to autodetect if it is seekable. Default
 value is -1.

@item chunked_post
-If set to 1 use chunked transfer-encoding for posts, default is 1.
+If set to 1 use chunked Transfer-Encoding for posts, default is 1.
+
+@item content_type
+Set a specific content type for the POST messages.

@item headers
 Set custom HTTP headers, can override built in default headers. The
 value must be a string encoding the headers.

-@item content_type
-Force a content type.
-
-@item user-agent
-Override User-Agent header. If not specified the protocol will use a
-string describing the libavformat build.
-
@item multiple_requests
-Use persistent connections if set to 1. By default it is 0.
+Use persistent connections if set to 1, default is 0.

@item post_data
 Set custom HTTP post data.

+@item user-agent
+@item user_agent
+Override the User-Agent header. If not specified the protocol will use a
+string describing the libavformat build. ("Lavf/<version>")
+
@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 mime_type
-Set MIME type.
+Export the MIME type.

@item icy
 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
@@ -235,17 +258,25 @@ 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.
+If the server supports ICY metadata, this contains the ICY-specific HTTP reply
+headers, separated by 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.
+contains the last non-empty metadata packet sent by the server. It should be
+polled in regular intervals by applications interested in mid-stream metadata
+updates.

@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
 delimited by a newline character.
+
+@item offset
+Set initial byte offset.
+
+@item end_offset
+Try to limit the request to bytes preceding this offset.
@end table

@subsection HTTP Cookies
@@ -325,6 +356,16 @@ ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
@end example

+This protocol accepts the following options:
+
+@table @option
+@item blocksize
+Set I/O operation maximum block size, in bytes. Default value is
+@code{INT_MAX}, which results in not limiting the requested block size.
+Setting this value reasonably low improves user termination request reaction
+time, which is valuable if data transmission is slow.
+@end table
+
 Note that some formats (typically MOV), require the output protocol to
 be seekable, so they will fail with the pipe output protocol.

@@ -337,12 +378,18 @@ content across a TCP/IP network.

 The required syntax is:
@example
-rtmp://@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
+rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
@end example

 The accepted parameters are:
@table @option

+@item username
+An optional username (mostly for publishing).
+
+@item password
+An optional password (mostly for publishing).
+
@item server
 The address of the RTMP server.

@@ -393,7 +440,8 @@ times to construct arbitrary AMF sequences.

@item rtmp_flashver
 Version of the Flash plugin used to run the SWF player. The default
-is LNX 9,0,124,2.
+is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
+<libavformat version>).)

@item rtmp_flush_interval
 Number of packets flushed in the same request (RTMPT only). The default
@@ -443,6 +491,12 @@ For example to read with @command{ffplay} a multimedia resource named
 ffplay rtmp://myserver/vod/sample
@end example

+To publish to a password protected server, passing the playpath and
+app names separately:
+@example
+ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
+@end example
+
@section rtmpe

 Encrypted Real-Time Messaging Protocol.
@@ -483,7 +537,43 @@ The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
 for streaming multimedia content within HTTPS requests to traverse
 firewalls.

-@section rtmp, rtmpe, rtmps, rtmpt, rtmpte
+@section libssh
+
+Secure File Transfer Protocol via libssh
+
+Allow to read from or write to remote resources using SFTP protocol.
+
+Following syntax is required.
+
+@example
+sftp://[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 truncate
+Truncate existing files on write, if set to 1. A value of 0 prevents
+truncating. Default value is 1.
+
+@item private_key
+Specify the path of the file containing private key to use during authorization.
+By default libssh searches for keys in the @file{~/.ssh/} directory.
+
+@end table
+
+Example: Play a file stored on remote server.
+
+@example
+ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
+@end example
+
+@section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte

 Real-Time Messaging Protocol and its variants supported through
 librtmp.
@@ -525,10 +615,75 @@ ffplay "rtmp://myserver/live/mystream live=1"

@section rtp

-Real-Time Protocol.
+Real-time Transport Protocol.
+
+The required syntax for an RTP URL is:
+rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
+
+@var{port} specifies the RTP port to use.
+
+The following URL options are supported:
+
+@table @option
+
+@item ttl=@var{n}
+Set the TTL (Time-To-Live) value (for multicast only).
+
+@item rtcpport=@var{n}
+Set the remote RTCP port to @var{n}.
+
+@item localrtpport=@var{n}
+Set the local RTP port to @var{n}.
+
+@item localrtcpport=@var{n}'
+Set the local RTCP port to @var{n}.
+
+@item pkt_size=@var{n}
+Set max packet size (in bytes) to @var{n}.
+
+@item connect=0|1
+Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
+to 0).
+
+@item sources=@var{ip}[,@var{ip}]
+List allowed source IP addresses.
+
+@item block=@var{ip}[,@var{ip}]
+List disallowed (blocked) source IP addresses.
+
+@item write_to_source=0|1
+Send packets to the source address of the latest received packet (if
+set to 1) or to a default remote address (if set to 0).
+
+@item localport=@var{n}
+Set the local RTP port to @var{n}.
+
+This is a deprecated option. Instead, @option{localrtpport} should be
+used.
+
+@end table
+
+Important notes:
+
+@enumerate
+
+@item
+If @option{rtcpport} is not set the RTCP port will be set to the RTP
+port value plus 1.
+
+@item
+If @option{localrtpport} (the local RTP port) is not set any available
+port will be used for the local RTP and RTCP ports.
+
+@item
+If @option{localrtcpport} (the local RTCP port) is not set it will be
+set to the the local RTP port value plus 1.
+@end enumerate

@section rtsp

+Real-Time Streaming Protocol.
+
 RTSP is not technically a protocol handler in libavformat, it is a demuxer
 and muxer. The demuxer supports both normal RTSP (with data transferred
 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
@@ -543,14 +698,22 @@ The required syntax for a RTSP url is:
 rtsp://@var{hostname}[:@var{port}]/@var{path}
@end example

-The following options (set on the @command{ffmpeg}/@command{ffplay} command
-line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
-are supported:
+Options can be set on the @command{ffmpeg}/@command{ffplay} command
+line, or set in code via @code{AVOption}s or in
+@code{avformat_open_input}.

-Flags for @code{rtsp_transport}:
+The following options are supported.

@table @option
+@item initial_pause
+Do not start playing the stream immediately if set to 1. Default value
+is 0.

+@item rtsp_transport
+Set RTSP trasport protocols.
+
+It accepts the following values:
+@table @samp
@item udp
 Use UDP as lower transport protocol.

@@ -568,17 +731,56 @@ passing proxies.

 Multiple lower transport protocols may be specified, in that case they are
 tried one at a time (if the setup of one fails, the next one is tried).
-For the muxer, only the @code{tcp} and @code{udp} options are supported.
+For the muxer, only the @samp{tcp} and @samp{udp} options are supported.

-Flags for @code{rtsp_flags}:
+@item rtsp_flags
+Set RTSP flags.

-@table @option
+The following values are accepted:
+@table @samp
@item filter_src
 Accept packets only from negotiated peer address and port.
@item listen
 Act as a server, listening for an incoming connection.
@end table

+Default value is @samp{none}.
+
+@item allowed_media_types
+Set media types to accept from the server.
+
+The following flags are accepted:
+@table @samp
+@item video
+@item audio
+@item data
+@end table
+
+By default it accepts all media types.
+
+@item min_port
+Set minimum local UDP port. Default value is 5000.
+
+@item max_port
+Set maximum local UDP port. Default value is 65000.
+
+@item timeout
+Set maximum timeout (in seconds) to wait for incoming connections.
+
+A value of -1 mean infinite (default). This option implies the
+@option{rtsp_flags} set to @samp{listen}.
+
+@item reorder_queue_size
+Set number of packets to buffer for handling of reordered packets.
+
+@item stimeout
+Set socket TCP I/O timeout in micro seconds.
+
+@item user-agent
+Override User-Agent header. If not specified, it default to the
+libavformat identifier string.
+@end table
+
 When receiving data over UDP, the demuxer tries to reorder received packets
 (since they may arrive out of order, or packets may get lost totally). This
 can be disabled by setting the maximum demuxing delay to zero (via
@@ -589,36 +791,36 @@ streams to display can be chosen with @code{-vst} @var{n} and
@code{-ast} @var{n} for video and audio respectively, and can be switched
 on the fly by pressing @code{v} and @code{a}.

-Example command lines:
+@subsection Examples

-To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
+The following examples all make use of the @command{ffplay} and
+@command{ffmpeg} tools.

+@itemize
+@item
+Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
@example
 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
@end example

-To watch a stream tunneled over HTTP:
-
+@item
+Watch a stream tunneled over HTTP:
@example
 ffplay -rtsp_transport http rtsp://server/video.mp4
@end example

-To send a stream in realtime to a RTSP server, for others to watch:
-
+@item
+Send a stream in realtime to a RTSP server, for others to watch:
@example
 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
@end example

-To receive a stream in realtime:
-
+@item
+Receive a stream in realtime:
@example
 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
+@end itemize

@section sap

@@ -758,55 +960,83 @@ used as master salt.

@section tcp

-Trasmission Control Protocol.
+Transmission Control Protocol.

 The required syntax for a TCP url is:
@example
 tcp://@var{hostname}:@var{port}[?@var{options}]
@end example

-@table @option
+@var{options} contains a list of &-separated options of the form
+@var{key}=@var{val}.

-@item listen
-Listen for an incoming connection
+The list of supported options follows.
+
+@table @option
+@item listen=@var{1|0}
+Listen for an incoming connection. Default value is 0.

@item timeout=@var{microseconds}
-In read mode: if no data arrived in more than this time interval, raise error.
-In write mode: if socket cannot be written in more than this time interval, raise error.
-This also sets timeout on TCP connection establishing.
+Set raise error timeout, expressed in microseconds.

+This option is only relevant in read mode: if no data arrived in more
+than this time interval, raise error.
+
+@item listen_timeout=@var{microseconds}
+Set listen timeout, expressed in microseconds.
+@end table
+
+The following example shows how to setup a listening TCP connection
+with @command{ffmpeg}, which is then accessed with @command{ffplay}:
@example
 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
 ffplay tcp://@var{hostname}:@var{port}
@end example

-@end table
-
@section tls

-Transport Layer Security/Secure Sockets Layer
+Transport Layer Security (TLS) / Secure Sockets Layer (SSL)

 The required syntax for a TLS/SSL url is:
@example
 tls://@var{hostname}:@var{port}[?@var{options}]
@end example

+The following parameters can be set via command line options
+(or in code via @code{AVOption}s):
+
@table @option

-@item listen
-Act as a server, listening for an incoming connection.
+@item ca_file, cafile=@var{filename}
+A file containing certificate authority (CA) root certificates to treat
+as trusted. If the linked TLS library contains a default this might not
+need to be specified for verification to work, but not all libraries and
+setups have defaults built in.
+The file must be in OpenSSL PEM format.

-@item cafile=@var{filename}
-Certificate authority file. The file must be in OpenSSL PEM format.
+@item tls_verify=@var{1|0}
+If enabled, try to verify the peer that we are communicating with.
+Note, if using OpenSSL, this currently only makes sure that the
+peer certificate is signed by one of the root certificates in the CA
+database, but it does not validate that the certificate actually
+matches the host name we are trying to connect to. (With GnuTLS,
+the host name is validated as well.)

-@item cert=@var{filename}
-Certificate file. The file must be in OpenSSL PEM format.
+This is disabled by default since it requires a CA database to be
+provided by the caller in many cases.

-@item key=@var{filename}
-Private key file.
+@item cert_file, cert=@var{filename}
+A file containing a certificate to use in the handshake with the peer.
+(When operating as server, in listen mode, this is more often required
+by the peer, while client certificates only are mandated in certain
+setups.)

-@item verify=@var{0|1}
-Verify the peer's certificate.
+@item key_file, key=@var{filename}
+A file containing the private key for the certificate.
+
+@item listen=@var{1|0}
+If enabled, listen for connections on the provided port, and assume
+the server role in the handshake instead of the client role.

@end table

@@ -828,7 +1058,7 @@ ffplay tls://@var{hostname}:@var{port}

 User Datagram Protocol.

-The required syntax for a UDP url is:
+The required syntax for an UDP URL is:
@example
 udp://@var{hostname}:@var{port}[?@var{options}]
@end example
@@ -836,14 +1066,13 @@ udp://@var{hostname}:@var{port}[?@var{options}]
@var{options} contains a list of &-separated options of the form @var{key}=@var{val}.

 In case threading is enabled on the system, a circular buffer is used
-to store the incoming data, which allows to reduce loss of data due to
+to store the incoming data, which allows one to reduce loss of data due to
 UDP socket buffer overruns. The @var{fifo_size} and
@var{overrun_nonfatal} options are related to this buffer.

 The list of supported options follows.

@table @option
-
@item buffer_size=@var{size}
 Set the UDP socket buffer size in bytes. This is used both for the
 receiving and the sending buffer size.
@@ -893,24 +1122,53 @@ Survive in case of UDP receiving circular buffer overrun. Default
 value is 0.

@item timeout=@var{microseconds}
-In read mode: if no data arrived in more than this time interval, raise error.
+Set raise error timeout, expressed in microseconds.
+
+This option is only relevant in read mode: if no data arrived in more
+than this time interval, raise error.
@end table

-Some usage examples of the UDP protocol with @command{ffmpeg} follow.
+@subsection Examples

-To stream over UDP to a remote endpoint:
+@itemize
+@item
+Use @command{ffmpeg} to stream over UDP to a remote endpoint:
@example
 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
@end example

-To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
+@item
+Use @command{ffmpeg} to stream in mpegts format over UDP using 188
+sized UDP packets, using a large input buffer:
@example
 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
@end example

-To receive over UDP from a remote endpoint:
+@item
+Use @command{ffmpeg} to receive over UDP from a remote endpoint:
@example
-ffmpeg -i udp://[@var{multicast-address}]:@var{port}
+ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
+@end example
+@end itemize
+
+@section unix
+
+Unix local socket
+
+The required syntax for a Unix socket URL is:
+
+@example
+unix://@var{filepath}
@end example

+The following parameters can be set via command line options
+(or in code via @code{AVOption}s):
+
+@table @option
+@item timeout
+Timeout in ms.
+@item listen
+Create the Unix socket in listening mode.
+@end table
+
@c man end PROTOCOLS
--- a/doc/resampler.texi
+++ b/doc/resampler.texi
@@ -42,10 +42,11 @@ 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.
+Set the input/output channel layout.
+
+See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils}
+for the required syntax.

@item clev, center_mix_level
 Set the center mix level. It is a value expressed in deciBel, and must be
@@ -63,6 +64,11 @@ be in the interval [-32,32].
@item rmvol, rematrix_volume
 Set rematrix volume. Default value is 1.0.

+@item rematrix_maxval
+Set maximum output value for rematrixing.
+This can be used to prevent clipping vs. preventing volumn reduction
+A value of 1.0 prevents cliping.
+
@item flags, swr_flags
 Set flags used by the converter. Default value is 0.

--- a/doc/scaler.texi
+++ b/doc/scaler.texi
@@ -1,3 +1,4 @@
+@anchor{scaler_options}
@chapter Scaler Options
@c man begin SCALER OPTIONS

@@ -9,6 +10,7 @@ FFmpeg tools. For programmatic use, they can be set explicitly in the

@table @option

+@anchor{sws_flags}
@item sws_flags
 Set the scaler flags. This is also used to set the scaling
 algorithm. Only a single algorithm should be selected.
@@ -33,7 +35,7 @@ Select nearest neighbor rescaling algorithm.
@item area
 Select averaging area rescaling algorithm.

-@item bicubiclin
+@item bicublin
 Select bicubic scaling algorithm for the luma component, bilinear for
 chroma components.

@@ -94,6 +96,24 @@ Set scaling algorithm parameters. The specified values are specific of
 some scaling algorithms and ignored by others. The specified values
 are floating point number values.

+@item sws_dither
+Set the dithering algorithm. Accepts one of the following
+values. Default value is @samp{auto}.
+
+@table @samp
+@item auto
+automatic choice
+
+@item none
+no dithering
+
+@item bayer
+bayer dither
+
+@item ed
+error diffusion dither
+@end table
+
@end table

@c man end SCALER OPTIONS
--- a/doc/snow.txt
+++ b/doc/snow.txt
@@ -50,8 +50,10 @@ header:
        temporal_decomposition_count    u   header_state
        spatial_decomposition_count     u   header_state
        colorspace_type                 u   header_state
-        chroma_h_shift                  u   header_state
-        chroma_v_shift                  u   header_state
+        if (nb_planes > 2) {
+            chroma_h_shift              u   header_state
+            chroma_v_shift              u   header_state
+        }
        spatial_scalability             b   header_state
        max_ref_frames-1                u   header_state
        qlogs
@@ -59,7 +61,7 @@ header:
    if(!keyframe){
        update_mc                       b   header_state
        if(update_mc){
-            for(plane=0; plane<2; plane++){
+            for(plane=0; plane<nb_plane_types; plane++){
                diag_mc                 b   header_state
                htaps/2-1               u   header_state
                for(i= p->htaps/2; i; i--)
@@ -80,7 +82,7 @@ header:
    block_max_depth                     s   header_state

 qlogs:
-    for(plane=0; plane<2; plane++){
+    for(plane=0; plane<nb_plane_types; plane++){
        quant_table[plane][0][0]        s   header_state
        for(level=0; level < spatial_decomposition_count; level++){
            quant_table[plane][level][1]s   header_state
@@ -131,8 +133,10 @@ block(level):

 residual:
    residual2(luma)
-    residual2(chroma_cr)
-    residual2(chroma_cb)
+    if (nb_planes > 2) {
+        residual2(chroma_cr)
+        residual2(chroma_cb)
+    }

 residual2:
    for(level=0; level<spatial_decomposition_count; level++){
@@ -146,7 +150,7 @@ residual2:
 subband:
    FIXME

-
+nb_plane_types = gray ? 1 : 2;

 Tag description:
 ----------------
@@ -168,7 +172,11 @@ spatial_decomposition_count
    FIXME

 colorspace_type
-    0
+    0   unspecified YcbCr
+    1   Gray
+    2   Gray + Alpha
+    3   GBR
+    4   GBRA
    this MUST NOT change within a bitstream

 chroma_h_shift
--- a/doc/soc.txt
+++ b/doc/soc.txt
@@ -1,24 +0,0 @@
-Google Summer of Code and similar project guidelines
-
-Summer of Code is a project by Google in which students are paid to implement
-some nice new features for various participating open source projects ...
-
-This text is a collection of things to take care of for the next soc as
-it's a little late for this year's soc (2006).
-
-The Goal:
-Our goal in respect to soc is and must be of course exactly one thing and
-that is to improve FFmpeg, to reach this goal, code must
-* conform to the development policy and patch submission guidelines
-* must improve FFmpeg somehow (faster, smaller, "better",
-  more codecs supported, fewer bugs, cleaner, ...)
-
-for mentors and other developers to help students to reach that goal it is
-essential that changes to their codebase are publicly visible, clean and
-easy reviewable that again leads us to:
-* use of a revision control system like git
-* separation of cosmetic from non-cosmetic changes (this is almost entirely
-  ignored by mentors and students in soc 2006 which might lead to a surprise
-  when the code will be reviewed at the end before a possible inclusion in
-  FFmpeg, individual changes were generally not reviewable due to cosmetics).
-* frequent commits, so that comments can be provided early
--- a/doc/t2h.init
+++ b/doc/t2h.init
@@ -17,6 +17,7 @@ my $TEMPLATE_HEADER = $ENV{"FFMPEG_HEADER"} || <<EOT;
 </head>
 <body>
 <div id="container">
+<div id="body">
 EOT

 $PRE_BODY_CLOSE = '</div></div>';
@@ -32,7 +33,7 @@ sub FFmpeg_print_page_foot($$)
        T2H_DEFAULT_program_string() : program_string();
    print $fh '<footer class="footer pagination-right">' . "\n";
    print $fh '<span class="label label-info">' . $program_string;
-    print $fh "</span></footer></div>\n";
+    print $fh "</span></footer></div></div></body>\n";
 }

 $float = \&FFmpeg_float;
@@ -92,8 +93,6 @@ $Texi2HTML::THISDOC{program_authors}

 $description
 <meta name="keywords" content="$longtitle">
-<meta name="resource-type" content="document">
-<meta name="distribution" content="global">
 <meta name="Generator" content="$Texi2HTML::THISDOC{program}">
 $encoding
 $CSS_LINES
--- a/doc/texi2pod.pl
+++ b/doc/texi2pod.pl
@@ -1,4 +1,4 @@
-#! /usr/bin/perl
+#!/usr/bin/env perl

 #   Copyright (C) 1999, 2000, 2001 Free Software Foundation, Inc.

@@ -327,6 +327,9 @@ die "No filename or title\n" unless defined $fn && defined $tl;
 $chapters{NAME} = "$fn \- $tl\n";
 $chapters{FOOTNOTES} .= "=back\n" if exists $chapters{FOOTNOTES};

+# always use utf8
+print "=encoding utf8\n\n";
+
 unshift @chapters_sequence, "NAME";
 for $chapter (@chapters_sequence) {
    if (exists $chapters{$chapter}) {
@@ -377,8 +380,8 @@ sub postprocess
    s/\(?\@xref\{(?:[^\}]*)\}(?:[^.<]|(?:<[^<>]*>))*\.\)?//g;
    s/\s+\(\@pxref\{(?:[^\}]*)\}\)//g;
    s/;\s+\@pxref\{(?:[^\}]*)\}//g;
-    s/\@ref\{(?:[^,\}]*,)(?:[^,\}]*,)([^,\}]*).*\}/$1/g;
-    s/\@ref\{([^\}]*)\}/$1/g;
+    s/\@ref\{(?:[^,\}]*,)(?:[^,\}]*,)([^,\}]*).*\}/B<$1>/g;
+    s/\@ref\{([^\}]*)\}/B<$1>/g;
    s/\@noindent\s*//g;
    s/\@refill//g;
    s/\@gol//g;
--- a/doc/texidep.pl
+++ b/doc/texidep.pl
@@ -0,0 +1,32 @@
+#! /usr/bin/env perl
+
+# This script will print the dependency of a Texinfo file to stdout.
+# texidep.pl <src-path> <input.texi> <output.ext>
+
+use warnings;
+use strict;
+
+die unless @ARGV == 3;
+
+my ($src_path, $root, $target) = @ARGV;
+
+sub print_deps {
+    my ($file, $deps) = @_;
+    $deps->{$file} = 1;
+
+    open(my $fh, "<", "$file") or die "Cannot open file '$file': $!";
+    while (<$fh>) {
+        if (my ($i) = /^\@(?:verbatim)?include\s+(\S+)/) {
+            die "Circular dependency found in file $root\n" if exists $deps->{"doc/$1"};
+            print "$target: doc/$1\n";
+
+            # skip looking for config.texi dependencies, since it has
+            # none, and is not located in the source tree
+            if ("$1" ne "config.texi") {
+                print_deps("$src_path/doc/$1", {%$deps});
+            }
+        }
+    }
+}
+
+print_deps($root, {});
--- a/doc/utils.texi
+++ b/doc/utils.texi
@@ -96,14 +96,42 @@ year-month-day.
@anchor{time duration syntax}
@section Time duration

-The accepted syntax is:
+There are two accepted syntaxes for expressing time duration.
+
@example
-[-][HH:]MM:SS[.m...]
-[-]S+[.m...]
+[-][@var{HH}:]@var{MM}:@var{SS}[.@var{m}...]
@end example

-@var{HH} expresses the number of hours, @var{MM} the number a of minutes
-and @var{SS} the number of seconds.
+@var{HH} expresses the number of hours, @var{MM} the number of minutes
+for a maximum of 2 digits, and @var{SS} the number of seconds for a
+maximum of 2 digits. The @var{m} at the end expresses decimal value for
+@var{SS}.
+
+@emph{or}
+
+@example
+[-]@var{S}+[.@var{m}...]
+@end example
+
+@var{S} expresses the number of seconds, with the optional decimal part
+@var{m}.
+
+In both expressions, the optional @samp{-} indicates negative duration.
+
+@subsection Examples
+
+The following examples are all valid time duration:
+
+@table @samp
+@item 55
+55 seconds
+
+@item 12:03:45
+12 hours, 03 minutes and 45 seconds
+
+@item 23.189
+23.189 seconds
+@end table

@anchor{video size syntax}
@section Video size
@@ -198,6 +226,18 @@ The following abbreviations are recognized:
 3996x2160
@item 4kscope
 4096x1716
+@item nhd
+640x360
+@item hqvga
+240x160
+@item wqvga
+400x240
+@item fwqvga
+432x240
+@item hvga
+480x320
+@item qhd
+960x540
@end table

@anchor{video rate syntax}
@@ -243,18 +283,450 @@ 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
+It can be the name of a color as defined below (case insensitive match) or a
+@code{[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.
+represents the opacity value (@samp{0x00} or @samp{0.0} means completely
+transparent, @samp{0xff} or @samp{1.0} completely opaque). If the alpha
+component is not specified then @samp{0xff} is assumed.

-The string "random" will result in a random color.
+The string @samp{random} will result in a random color.

+The following names of colors are recognized:
+@table @samp
+@item AliceBlue
+0xF0F8FF
+@item AntiqueWhite
+0xFAEBD7
+@item Aqua
+0x00FFFF
+@item Aquamarine
+0x7FFFD4
+@item Azure
+0xF0FFFF
+@item Beige
+0xF5F5DC
+@item Bisque
+0xFFE4C4
+@item Black
+0x000000
+@item BlanchedAlmond
+0xFFEBCD
+@item Blue
+0x0000FF
+@item BlueViolet
+0x8A2BE2
+@item Brown
+0xA52A2A
+@item BurlyWood
+0xDEB887
+@item CadetBlue
+0x5F9EA0
+@item Chartreuse
+0x7FFF00
+@item Chocolate
+0xD2691E
+@item Coral
+0xFF7F50
+@item CornflowerBlue
+0x6495ED
+@item Cornsilk
+0xFFF8DC
+@item Crimson
+0xDC143C
+@item Cyan
+0x00FFFF
+@item DarkBlue
+0x00008B
+@item DarkCyan
+0x008B8B
+@item DarkGoldenRod
+0xB8860B
+@item DarkGray
+0xA9A9A9
+@item DarkGreen
+0x006400
+@item DarkKhaki
+0xBDB76B
+@item DarkMagenta
+0x8B008B
+@item DarkOliveGreen
+0x556B2F
+@item Darkorange
+0xFF8C00
+@item DarkOrchid
+0x9932CC
+@item DarkRed
+0x8B0000
+@item DarkSalmon
+0xE9967A
+@item DarkSeaGreen
+0x8FBC8F
+@item DarkSlateBlue
+0x483D8B
+@item DarkSlateGray
+0x2F4F4F
+@item DarkTurquoise
+0x00CED1
+@item DarkViolet
+0x9400D3
+@item DeepPink
+0xFF1493
+@item DeepSkyBlue
+0x00BFFF
+@item DimGray
+0x696969
+@item DodgerBlue
+0x1E90FF
+@item FireBrick
+0xB22222
+@item FloralWhite
+0xFFFAF0
+@item ForestGreen
+0x228B22
+@item Fuchsia
+0xFF00FF
+@item Gainsboro
+0xDCDCDC
+@item GhostWhite
+0xF8F8FF
+@item Gold
+0xFFD700
+@item GoldenRod
+0xDAA520
+@item Gray
+0x808080
+@item Green
+0x008000
+@item GreenYellow
+0xADFF2F
+@item HoneyDew
+0xF0FFF0
+@item HotPink
+0xFF69B4
+@item IndianRed
+0xCD5C5C
+@item Indigo
+0x4B0082
+@item Ivory
+0xFFFFF0
+@item Khaki
+0xF0E68C
+@item Lavender
+0xE6E6FA
+@item LavenderBlush
+0xFFF0F5
+@item LawnGreen
+0x7CFC00
+@item LemonChiffon
+0xFFFACD
+@item LightBlue
+0xADD8E6
+@item LightCoral
+0xF08080
+@item LightCyan
+0xE0FFFF
+@item LightGoldenRodYellow
+0xFAFAD2
+@item LightGreen
+0x90EE90
+@item LightGrey
+0xD3D3D3
+@item LightPink
+0xFFB6C1
+@item LightSalmon
+0xFFA07A
+@item LightSeaGreen
+0x20B2AA
+@item LightSkyBlue
+0x87CEFA
+@item LightSlateGray
+0x778899
+@item LightSteelBlue
+0xB0C4DE
+@item LightYellow
+0xFFFFE0
+@item Lime
+0x00FF00
+@item LimeGreen
+0x32CD32
+@item Linen
+0xFAF0E6
+@item Magenta
+0xFF00FF
+@item Maroon
+0x800000
+@item MediumAquaMarine
+0x66CDAA
+@item MediumBlue
+0x0000CD
+@item MediumOrchid
+0xBA55D3
+@item MediumPurple
+0x9370D8
+@item MediumSeaGreen
+0x3CB371
+@item MediumSlateBlue
+0x7B68EE
+@item MediumSpringGreen
+0x00FA9A
+@item MediumTurquoise
+0x48D1CC
+@item MediumVioletRed
+0xC71585
+@item MidnightBlue
+0x191970
+@item MintCream
+0xF5FFFA
+@item MistyRose
+0xFFE4E1
+@item Moccasin
+0xFFE4B5
+@item NavajoWhite
+0xFFDEAD
+@item Navy
+0x000080
+@item OldLace
+0xFDF5E6
+@item Olive
+0x808000
+@item OliveDrab
+0x6B8E23
+@item Orange
+0xFFA500
+@item OrangeRed
+0xFF4500
+@item Orchid
+0xDA70D6
+@item PaleGoldenRod
+0xEEE8AA
+@item PaleGreen
+0x98FB98
+@item PaleTurquoise
+0xAFEEEE
+@item PaleVioletRed
+0xD87093
+@item PapayaWhip
+0xFFEFD5
+@item PeachPuff
+0xFFDAB9
+@item Peru
+0xCD853F
+@item Pink
+0xFFC0CB
+@item Plum
+0xDDA0DD
+@item PowderBlue
+0xB0E0E6
+@item Purple
+0x800080
+@item Red
+0xFF0000
+@item RosyBrown
+0xBC8F8F
+@item RoyalBlue
+0x4169E1
+@item SaddleBrown
+0x8B4513
+@item Salmon
+0xFA8072
+@item SandyBrown
+0xF4A460
+@item SeaGreen
+0x2E8B57
+@item SeaShell
+0xFFF5EE
+@item Sienna
+0xA0522D
+@item Silver
+0xC0C0C0
+@item SkyBlue
+0x87CEEB
+@item SlateBlue
+0x6A5ACD
+@item SlateGray
+0x708090
+@item Snow
+0xFFFAFA
+@item SpringGreen
+0x00FF7F
+@item SteelBlue
+0x4682B4
+@item Tan
+0xD2B48C
+@item Teal
+0x008080
+@item Thistle
+0xD8BFD8
+@item Tomato
+0xFF6347
+@item Turquoise
+0x40E0D0
+@item Violet
+0xEE82EE
+@item Wheat
+0xF5DEB3
+@item White
+0xFFFFFF
+@item WhiteSmoke
+0xF5F5F5
+@item Yellow
+0xFFFF00
+@item YellowGreen
+0x9ACD32
+@end table
+
+@anchor{channel layout syntax}
+@section Channel Layout
+
+A channel layout specifies the spatial disposition of the channels in
+a multi-channel audio stream. To specify a channel layout, FFmpeg
+makes use of a special syntax.
+
+Individual channels are identified by an id, as given by the table
+below:
+@table @samp
+@item FL
+front left
+@item FR
+front right
+@item FC
+front center
+@item LFE
+low frequency
+@item BL
+back left
+@item BR
+back right
+@item FLC
+front left-of-center
+@item FRC
+front right-of-center
+@item BC
+back center
+@item SL
+side left
+@item SR
+side right
+@item TC
+top center
+@item TFL
+top front left
+@item TFC
+top front center
+@item TFR
+top front right
+@item TBL
+top back left
+@item TBC
+top back center
+@item TBR
+top back right
+@item DL
+downmix left
+@item DR
+downmix right
+@item WL
+wide left
+@item WR
+wide right
+@item SDL
+surround direct left
+@item SDR
+surround direct right
+@item LFE2
+low frequency 2
+@end table
+
+Standard channel layout compositions can be specified by using the
+following identifiers:
+@table @samp
+@item mono
+FC
+@item stereo
+FL+FR
+@item 2.1
+FL+FR+LFE
+@item 3.0
+FL+FR+FC
+@item 3.0(back)
+FL+FR+BC
+@item 4.0
+FL+FR+FC+BC
+@item quad
+FL+FR+BL+BR
+@item quad(side)
+FL+FR+SL+SR
+@item 3.1
+FL+FR+FC+LFE
+@item 5.0
+FL+FR+FC+BL+BR
+@item 5.0(side)
+FL+FR+FC+SL+SR
+@item 4.1
+FL+FR+FC+LFE+BC
+@item 5.1
+FL+FR+FC+LFE+BL+BR
+@item 5.1(side)
+FL+FR+FC+LFE+SL+SR
+@item 6.0
+FL+FR+FC+BC+SL+SR
+@item 6.0(front)
+FL+FR+FLC+FRC+SL+SR
+@item hexagonal
+FL+FR+FC+BL+BR+BC
+@item 6.1
+FL+FR+FC+LFE+BC+SL+SR
+@item 6.1
+FL+FR+FC+LFE+BL+BR+BC
+@item 6.1(front)
+FL+FR+LFE+FLC+FRC+SL+SR
+@item 7.0
+FL+FR+FC+BL+BR+SL+SR
+@item 7.0(front)
+FL+FR+FC+FLC+FRC+SL+SR
+@item 7.1
+FL+FR+FC+LFE+BL+BR+SL+SR
+@item 7.1(wide)
+FL+FR+FC+LFE+BL+BR+FLC+FRC
+@item 7.1(wide-side)
+FL+FR+FC+LFE+FLC+FRC+SL+SR
+@item octagonal
+FL+FR+FC+BL+BR+BC+SL+SR
+@item downmix
+DL+DR
+@end table
+
+A custom channel layout can be specified as a sequence of terms, separated by
+'+' or '|'. Each term can be:
+@itemize
+@item
+the name of a standard channel layout (e.g. @samp{mono},
+@samp{stereo}, @samp{4.0}, @samp{quad}, @samp{5.0}, etc.)
+
+@item
+the name of a single channel (e.g. @samp{FL}, @samp{FR}, @samp{FC}, @samp{LFE}, etc.)
+
+@item
+a number of channels, in decimal, optionally followed by 'c', yielding
+the default channel layout for that number of channels (see the
+function @code{av_get_default_channel_layout})
+
+@item
+a channel layout mask, in hexadecimal starting with "0x" (see the
+@code{AV_CH_*} macros in @file{libavutil/channel_layout.h}.
+@end itemize
+
+Starting from libavutil version 53 the trailing character "c" to
+specify a number of channels will be required, while a channel layout
+mask could also be specified as a decimal number (if and only if not
+followed by "c").
+
+See also the function @code{av_get_channel_layout} defined in
+@file{libavutil/channel_layout.h}.
@c man end SYNTAX

@chapter Expression Evaluation
@@ -579,13 +1051,13 @@ See reference "OpenCL Specification Version: 1.2 chapter 5.6.4".
 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()}.
+which can be obtained with @code{ffmpeg -opencl_bench} or @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()}.
+The specified index must be one of the indexes in the device list which
+can be obtained with @code{ffmpeg -opencl_bench} or @code{av_opencl_get_device_list()}.

@end table

--- a/doc/viterbi.txt
+++ b/doc/viterbi.txt
@@ -1,109 +0,0 @@
-This is a quick description of the viterbi aka dynamic programing
-algorthm.
-
-Its reason for existence is that wikipedia has become very poor on
-describing algorithms in a way that makes it useable for understanding
-them or anything else actually. It tends now to describe the very same
-algorithm under 50 different names and pages with few understandable
-by even people who fully understand the algorithm and the theory behind.
-
-Problem description: (that is what it can solve)
-assume we have a 2d table, or you could call it a graph or matrix if you
-prefer
-
-    O   O   O   O   O   O   O
-
-    O   O   O   O   O   O   O
-
-    O   O   O   O   O   O   O
-
-    O   O   O   O   O   O   O
-
-
-That table has edges connecting points from each column to the next column
-and each edge has a score like: (only some edge and scores shown to keep it
-readable)
-
-
-    O--5--O-----O-----O-----O-----O
-     2   / 7   / \   / \   / \   /
-      \ /   \ /   \ /   \ /   \ /
-    O7-/--O--/--O--/--O--/--O--/--O
-     \/ \/ 1/ \/ \/ \/ \/ \/ \/ \/
-     /\ /\ 2\ /\ /\ /\ /\ /\ /\ /\
-    O3-/--O--/--O--/--O--/--O--/--O
-      / \   / \   / \   / \   / \
-     1   \ 9   \ /   \ /   \ /   \
-    O--2--O--1--O--5--O--3--O--8--O
-
-
-
-Our goal is to find a path from left to right through it which
-minimizes the sum of the score of all edges.
-(and of course left/right is just a convention here it could be top down too)
-Similarly the minimum could be the maximum by just fliping the sign,
-Example of a path with scores:
-
-    O   O   O   O   O   O   O
-
->---O.  O   O  .O-2-O   O   O
-      5.     .7      .
-    O   O-1-O   O   O 8 O   O
-                       .
-    O   O   O   O   O   O-1-O---> (sum here is 24)
-
-
-The viterbi algorthm now solves this simply column by column
-For the previous column each point has a best path and a associated
-score:
-
-    O-----5     O
-     \
-      \
-    O  \  1     O
-        \/
-        /\
-    O  /  2     O
-      /
-     /
-    O-----2     O
-
-
-To move one column forward we just need to find the best path and associated
-scores for the next column
-here are some edges we could choose from:
-
-
-    O-----5--3--O
-     \      \8
-      \       \
-    O  \  1--9--O
-        \/  \3
-        /\     \
-    O  /  2--1--O
-      /     \2
-     /        \
-    O-----2--4--O
-
-Finding the new best paths and scores for each point of our new column is
-trivial given we know the previous column best paths and scores:
-
-    O-----0-----8
-     \
-      \
-    O  \  0----10
-        \/
-        /\
-    O  /  0-----3
-      /     \
-     /        \
-    O     0     4
-
-
-the viterbi algorthm continues exactly like this column for column until the
-end and then just picks the path with the best score (above that would be the
-one with score 3)
-
-
-Author: Michael niedermayer
-Copyright LGPL
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -30,6 +30,8 @@
 #include <stdlib.h>
 #include <errno.h>
 #include <limits.h>
+#include <stdint.h>
+
 #if HAVE_ISATTY
 #if HAVE_IO_H
 #include <io.h>
@@ -38,9 +40,9 @@
 #include <unistd.h>
 #endif
 #endif
+
 #include "libavformat/avformat.h"
 #include "libavdevice/avdevice.h"
-#include "libswscale/swscale.h"
 #include "libswresample/swresample.h"
 #include "libavutil/opt.h"
 #include "libavutil/channel_layout.h"
@@ -123,6 +125,7 @@ static int64_t getmaxrss(void);
 static int run_as_daemon  = 0;
 static int64_t video_size = 0;
 static int64_t audio_size = 0;
+static int64_t data_size = 0;
 static int64_t subtitle_size = 0;
 static int64_t extra_size = 0;
 static int nb_frames_dup = 0;
@@ -311,6 +314,7 @@ void term_exit(void)

 static volatile int received_sigterm = 0;
 static volatile int received_nb_signals = 0;
+static int main_return_code = 0;

 static void
 sigterm_handler(int sig)
@@ -319,7 +323,7 @@ sigterm_handler(int sig)
    received_nb_signals++;
    term_exit();
    if(received_nb_signals > 3)
-        exit_program(123);
+        exit(123);
 }

 void term_init(void)
@@ -466,7 +470,9 @@ static void ffmpeg_cleanup(int ret)
            bsfc = next;
        }
        output_streams[i]->bitstream_filters = NULL;
-        avcodec_free_frame(&output_streams[i]->filtered_frame);
+        av_frame_free(&output_streams[i]->filtered_frame);
+
+        av_parser_close(output_streams[i]->parser);

        av_freep(&output_streams[i]->forced_keyframes);
        av_expr_free(output_streams[i]->forced_keyframes_pexpr);
@@ -488,6 +494,7 @@ static void ffmpeg_cleanup(int ret)
        avsubtitle_free(&input_streams[i]->prev_sub.subtitle);
        av_frame_free(&input_streams[i]->sub2video.frame);
        av_freep(&input_streams[i]->filters);
+        av_freep(&input_streams[i]->hwaccel_device);
        av_freep(&input_streams[i]);
    }

@@ -507,6 +514,8 @@ static void ffmpeg_cleanup(int ret)
    if (received_sigterm) {
        av_log(NULL, AV_LOG_INFO, "Received signal %d: terminating.\n",
               (int) received_sigterm);
+    } else if (ret) {
+        av_log(NULL, AV_LOG_INFO, "Conversion failed!\n");
    }
    term_exit();
 }
@@ -542,6 +551,15 @@ static void update_benchmark(const char *fmt, ...)
    }
 }

+static void close_all_output_streams(OutputStream *ost, OSTFinished this_stream, OSTFinished others)
+{
+    int i;
+    for (i = 0; i < nb_output_streams; i++) {
+        OutputStream *ost2 = output_streams[i];
+        ost2->finished |= ost == ost2 ? this_stream : others;
+    }
+}
+
 static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
 {
    AVBitStreamFilterContext *bsfc = ost->bitstream_filters;
@@ -603,7 +621,8 @@ static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
        bsfc = bsfc->next;
    }

-    if (!(s->oformat->flags & AVFMT_NOTIMESTAMPS) &&
+    if (!(s->oformat->flags & AVFMT_NOTIMESTAMPS)) {
+     if(
        (avctx->codec_type == AVMEDIA_TYPE_AUDIO || avctx->codec_type == AVMEDIA_TYPE_VIDEO) &&
        pkt->dts != AV_NOPTS_VALUE &&
        ost->last_mux_dts != AV_NOPTS_VALUE) {
@@ -624,6 +643,16 @@ static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
            pkt->pts = FFMAX(pkt->pts, max);
        pkt->dts = max;
      }
+     }
+        if (pkt->dts != AV_NOPTS_VALUE &&
+            pkt->pts != AV_NOPTS_VALUE &&
+            pkt->dts > pkt->pts) {
+            av_log(s, AV_LOG_WARNING, "Invalid DTS: %"PRId64" PTS: %"PRId64" in output stream %d:%d\n",
+                   pkt->dts, pkt->pts,
+                   ost->file_index, ost->st->index);
+            pkt->pts = AV_NOPTS_VALUE;
+            pkt->dts = AV_NOPTS_VALUE;
+        }
    }
    ost->last_mux_dts = pkt->dts;

@@ -642,7 +671,8 @@ static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
    ret = av_interleaved_write_frame(s, pkt);
    if (ret < 0) {
        print_error("av_interleaved_write_frame()", ret);
-        exit_program(1);
+        main_return_code = 1;
+        close_all_output_streams(ost, MUXER_FINISHED | ENCODER_FINISHED, ENCODER_FINISHED);
    }
 }

@@ -650,7 +680,7 @@ static void close_output_stream(OutputStream *ost)
 {
    OutputFile *of = output_files[ost->file_index];

-    ost->finished = 1;
+    ost->finished |= ENCODER_FINISHED;
    if (of->shortest) {
        int64_t end = av_rescale_q(ost->sync_opts - ost->first_pts, ost->st->codec->time_base, AV_TIME_BASE_Q);
        of->recording_time = FFMIN(of->recording_time, end);
@@ -690,6 +720,12 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost,

    av_assert0(pkt.size || !pkt.data);
    update_benchmark(NULL);
+    if (debug_ts) {
+        av_log(NULL, AV_LOG_INFO, "encoder <- type:audio "
+               "frame_pts:%s frame_pts_time:%s time_base:%d/%d\n",
+               av_ts2str(frame->pts), av_ts2timestr(frame->pts, &enc->time_base),
+               enc->time_base.num, enc->time_base.den);
+    }
    if (avcodec_encode_audio2(enc, &pkt, frame, &got_packet) < 0) {
        av_log(NULL, AV_LOG_FATAL, "Audio encoding failed (avcodec_encode_audio2)\n");
        exit_program(1);
@@ -751,7 +787,9 @@ static void do_subtitle_out(AVFormatContext *s,
        nb = 1;

    /* shift timestamp to honor -ss and make check_recording_time() work with -t */
-    pts = sub->pts - output_files[ost->file_index]->start_time;
+    pts = sub->pts;
+    if (output_files[ost->file_index]->start_time != AV_NOPTS_VALUE)
+        pts -= output_files[ost->file_index]->start_time;
    for (i = 0; i < nb; i++) {
        ost->sync_opts = av_rescale_q(pts, AV_TIME_BASE_Q, enc->time_base);
        if (!check_recording_time(ost))
@@ -815,10 +853,29 @@ static void do_video_out(AVFormatContext *s,
    nb_frames = 1;

    format_video_sync = video_sync_method;
-    if (format_video_sync == VSYNC_AUTO)
-        format_video_sync = (s->oformat->flags & AVFMT_VARIABLE_FPS) ? ((s->oformat->flags & AVFMT_NOTIMESTAMPS) ? VSYNC_PASSTHROUGH : VSYNC_VFR) : VSYNC_CFR;
+    if (format_video_sync == VSYNC_AUTO) {
+        if(!strcmp(s->oformat->name, "avi")) {
+            format_video_sync = VSYNC_VFR;
+        } else
+            format_video_sync = (s->oformat->flags & AVFMT_VARIABLE_FPS) ? ((s->oformat->flags & AVFMT_NOTIMESTAMPS) ? VSYNC_PASSTHROUGH : VSYNC_VFR) : VSYNC_CFR;
+        if (   ist
+            && format_video_sync == VSYNC_CFR
+            && input_files[ist->file_index]->ctx->nb_streams == 1
+            && input_files[ist->file_index]->input_ts_offset == 0) {
+            format_video_sync = VSYNC_VSCFR;
+        }
+        if (format_video_sync == VSYNC_CFR && copy_ts) {
+            format_video_sync = VSYNC_VSCFR;
+        }
+    }

    switch (format_video_sync) {
+    case VSYNC_VSCFR:
+        if (ost->frame_number == 0 && delta - duration >= 0.5) {
+            av_log(NULL, AV_LOG_DEBUG, "Not duplicating %d initial frames\n", (int)lrintf(delta - duration));
+            delta = duration;
+            ost->sync_opts = lrint(sync_ipts);
+        }
    case VSYNC_CFR:
        // FIXME set to 0.5 after we fix some dts/pts bugs like in avidec.c
        if (delta < -1.1)
@@ -943,6 +1000,13 @@ static void do_video_out(AVFormatContext *s,
        }

        update_benchmark(NULL);
+        if (debug_ts) {
+            av_log(NULL, AV_LOG_INFO, "encoder <- type:video "
+                   "frame_pts:%s frame_pts_time:%s time_base:%d/%d\n",
+                   av_ts2str(in_picture->pts), av_ts2timestr(in_picture->pts, &enc->time_base),
+                   enc->time_base.num, enc->time_base.den);
+        }
+
        ret = avcodec_encode_video2(enc, &pkt, in_picture, &got_packet);
        update_benchmark("encode_video %d.%d", ost->file_index, ost->index);
        if (ret < 0) {
@@ -951,6 +1015,13 @@ static void do_video_out(AVFormatContext *s,
        }

        if (got_packet) {
+            if (debug_ts) {
+                av_log(NULL, AV_LOG_INFO, "encoder -> type:video "
+                       "pkt_pts:%s pkt_pts_time:%s pkt_dts:%s pkt_dts_time:%s\n",
+                       av_ts2str(pkt.pts), av_ts2timestr(pkt.pts, &enc->time_base),
+                       av_ts2str(pkt.dts), av_ts2timestr(pkt.dts, &enc->time_base));
+            }
+
            if (pkt.pts == AV_NOPTS_VALUE && !(enc->codec->capabilities & CODEC_CAP_DELAY))
                pkt.pts = ost->sync_opts;

@@ -1031,6 +1102,19 @@ static void do_video_stats(OutputStream *ost, int frame_size)
    }
 }

+static void finish_output_stream(OutputStream *ost)
+{
+    OutputFile *of = output_files[ost->file_index];
+    int i;
+
+    ost->finished = ENCODER_FINISHED | MUXER_FINISHED;
+
+    if (of->shortest) {
+        for (i = 0; i < of->ctx->nb_streams; i++)
+            output_streams[of->ost_index + i]->finished = ENCODER_FINISHED | MUXER_FINISHED;
+    }
+}
+
 /**
 * Get and encode new output from any of the filtergraphs, without causing
 * activity.
@@ -1047,19 +1131,21 @@ static int reap_filters(void)
    for (i = 0; i < nb_output_streams; i++) {
        OutputStream *ost = output_streams[i];
        OutputFile    *of = output_files[ost->file_index];
+        AVFilterContext *filter;
+        AVCodecContext *enc = ost->st->codec;
        int ret = 0;

        if (!ost->filter)
            continue;
+        filter = ost->filter->filter;

-        if (!ost->filtered_frame && !(ost->filtered_frame = avcodec_alloc_frame())) {
+        if (!ost->filtered_frame && !(ost->filtered_frame = av_frame_alloc())) {
            return AVERROR(ENOMEM);
-        } else
-            avcodec_get_frame_defaults(ost->filtered_frame);
+        }
        filtered_frame = ost->filtered_frame;

        while (1) {
-            ret = av_buffersink_get_frame_flags(ost->filter->filter, filtered_frame,
+            ret = av_buffersink_get_frame_flags(filter, filtered_frame,
                                               AV_BUFFERSINK_FLAG_NO_REQUEST);
            if (ret < 0) {
                if (ret != AVERROR(EAGAIN) && ret != AVERROR_EOF) {
@@ -1068,31 +1154,38 @@ static int reap_filters(void)
                }
                break;
            }
+            if (ost->finished) {
+                av_frame_unref(filtered_frame);
+                continue;
+            }
            frame_pts = AV_NOPTS_VALUE;
            if (filtered_frame->pts != AV_NOPTS_VALUE) {
-                filtered_frame->pts = frame_pts = av_rescale_q(filtered_frame->pts,
-                                                ost->filter->filter->inputs[0]->time_base,
-                                                ost->st->codec->time_base) -
-                                    av_rescale_q(of->start_time,
-                                                AV_TIME_BASE_Q,
-                                                ost->st->codec->time_base);
+                int64_t start_time = (of->start_time == AV_NOPTS_VALUE) ? 0 : of->start_time;
+                filtered_frame->pts = frame_pts =
+                    av_rescale_q(filtered_frame->pts, filter->inputs[0]->time_base, enc->time_base) -
+                    av_rescale_q(start_time, AV_TIME_BASE_Q, enc->time_base);
            }
            //if (ost->source_index >= 0)
            //    *filtered_frame= *input_streams[ost->source_index]->decoded_frame; //for me_threshold

-
-            switch (ost->filter->filter->inputs[0]->type) {
+            switch (filter->inputs[0]->type) {
            case AVMEDIA_TYPE_VIDEO:
                filtered_frame->pts = frame_pts;
                if (!ost->frame_aspect_ratio.num)
-                    ost->st->codec->sample_aspect_ratio = filtered_frame->sample_aspect_ratio;
+                    enc->sample_aspect_ratio = filtered_frame->sample_aspect_ratio;
+
+                if (debug_ts) {
+                    av_log(NULL, AV_LOG_INFO, "filter -> pts:%s pts_time:%s time_base:%d/%d\n",
+                            av_ts2str(filtered_frame->pts), av_ts2timestr(filtered_frame->pts, &enc->time_base),
+                            enc->time_base.num, enc->time_base.den);
+                }

                do_video_out(of->ctx, ost, filtered_frame);
                break;
            case AVMEDIA_TYPE_AUDIO:
                filtered_frame->pts = frame_pts;
-                if (!(ost->st->codec->codec->capabilities & CODEC_CAP_PARAM_CHANGE) &&
-                    ost->st->codec->channels != av_frame_get_channels(filtered_frame)) {
+                if (!(enc->codec->capabilities & CODEC_CAP_PARAM_CHANGE) &&
+                    enc->channels != av_frame_get_channels(filtered_frame)) {
                    av_log(NULL, AV_LOG_ERROR,
                           "Audio filter graph output is not normalized and encoder does not support parameter changes\n");
                    break;
@@ -1273,16 +1366,21 @@ static void print_report(int is_last_report, int64_t timer_start, int64_t cur_ti
    }

    if (is_last_report) {
-        int64_t raw= audio_size + video_size + subtitle_size + extra_size;
+        int64_t raw   = audio_size + video_size + data_size + subtitle_size + extra_size;
+        float percent = 0.0;
+
+        if (raw)
+            percent = 100.0 * (total_size - raw) / raw;
+
        av_log(NULL, AV_LOG_INFO, "\n");
-        av_log(NULL, AV_LOG_INFO, "video:%1.0fkB audio:%1.0fkB subtitle:%1.0f global headers:%1.0fkB muxing overhead %f%%\n",
+        av_log(NULL, AV_LOG_INFO, "video:%1.0fkB audio:%1.0fkB subtitle:%1.0f data:%1.0f global headers:%1.0fkB muxing overhead %f%%\n",
               video_size / 1024.0,
               audio_size / 1024.0,
               subtitle_size / 1024.0,
+               data_size / 1024.0,
               extra_size / 1024.0,
-               100.0 * (total_size - raw) / raw
-        );
-        if(video_size + audio_size + subtitle_size + extra_size == 0){
+               percent);
+        if(video_size + data_size + audio_size + subtitle_size + extra_size == 0){
            av_log(NULL, AV_LOG_WARNING, "Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)\n");
        }
    }
@@ -1328,6 +1426,7 @@ static void flush_encoders(void)

            if (encode) {
                AVPacket pkt;
+                int pkt_size;
                int got_packet;
                av_init_packet(&pkt);
                pkt.data = NULL;
@@ -1340,7 +1439,6 @@ static void flush_encoders(void)
                    av_log(NULL, AV_LOG_FATAL, "%s encoding failed\n", desc);
                    exit_program(1);
                }
-                *size += pkt.size;
                if (ost->logfile && enc->stats_out) {
                    fprintf(ost->logfile, "%s", enc->stats_out);
                }
@@ -1348,15 +1446,21 @@ static void flush_encoders(void)
                    stop_encoding = 1;
                    break;
                }
+                if (ost->finished & MUXER_FINISHED) {
+                    av_free_packet(&pkt);
+                    continue;
+                }
+                *size += pkt.size;
                if (pkt.pts != AV_NOPTS_VALUE)
                    pkt.pts = av_rescale_q(pkt.pts, enc->time_base, ost->st->time_base);
                if (pkt.dts != AV_NOPTS_VALUE)
                    pkt.dts = av_rescale_q(pkt.dts, enc->time_base, ost->st->time_base);
                if (pkt.duration > 0)
                    pkt.duration = av_rescale_q(pkt.duration, enc->time_base, ost->st->time_base);
+                pkt_size = pkt.size;
                write_frame(os, &pkt, ost);
                if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && vstats_filename) {
-                    do_video_stats(ost, pkt.size);
+                    do_video_stats(ost, pkt_size);
                }
            }

@@ -1377,7 +1481,10 @@ static int check_output_constraints(InputStream *ist, OutputStream *ost)
    if (ost->source_index != ist_index)
        return 0;

-    if (of->start_time && ist->pts < of->start_time)
+    if (ost->finished)
+        return 0;
+
+    if (of->start_time != AV_NOPTS_VALUE && ist->pts < of->start_time)
        return 0;

    return 1;
@@ -1386,8 +1493,10 @@ static int check_output_constraints(InputStream *ist, OutputStream *ost)
 static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *pkt)
 {
    OutputFile *of = output_files[ost->file_index];
-    int64_t ost_tb_start_time = av_rescale_q(of->start_time, AV_TIME_BASE_Q, ost->st->time_base);
-    int64_t ist_tb_start_time = av_rescale_q(of->start_time, AV_TIME_BASE_Q, ist->st->time_base);
+    InputFile   *f = input_files [ist->file_index];
+    int64_t start_time = (of->start_time == AV_NOPTS_VALUE) ? 0 : of->start_time;
+    int64_t ost_tb_start_time = av_rescale_q(start_time, AV_TIME_BASE_Q, ost->st->time_base);
+    int64_t ist_tb_start_time = av_rescale_q(start_time, AV_TIME_BASE_Q, ist->st->time_base);
    AVPicture pict;
    AVPacket opkt;

@@ -1398,7 +1507,7 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
        return;

    if (pkt->pts == AV_NOPTS_VALUE) {
-        if (!ost->frame_number && ist->pts < of->start_time &&
+        if (!ost->frame_number && ist->pts < start_time &&
            !ost->copy_prior_start)
            return;
    } else {
@@ -1408,17 +1517,29 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
    }

    if (of->recording_time != INT64_MAX &&
-        ist->pts >= of->recording_time + of->start_time) {
+        ist->pts >= of->recording_time + start_time) {
        close_output_stream(ost);
        return;
    }

+    if (f->recording_time != INT64_MAX) {
+        start_time = f->ctx->start_time;
+        if (f->start_time != AV_NOPTS_VALUE)
+            start_time += f->start_time;
+        if (ist->pts >= f->recording_time + start_time) {
+            close_output_stream(ost);
+            return;
+        }
+    }
+
    /* force the input stream PTS */
    if (ost->st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
        audio_size += pkt->size;
    else if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
        video_size += pkt->size;
        ost->sync_opts++;
+    } else if (ost->st->codec->codec_type == AVMEDIA_TYPE_DATA) {
+        data_size += pkt->size;
    } else if (ost->st->codec->codec_type == AVMEDIA_TYPE_SUBTITLE) {
        subtitle_size += pkt->size;
    }
@@ -1452,7 +1573,10 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
       && ost->st->codec->codec_id != AV_CODEC_ID_MPEG2VIDEO
       && ost->st->codec->codec_id != AV_CODEC_ID_VC1
       ) {
-        if (av_parser_change(ist->st->parser, ost->st->codec, &opkt.data, &opkt.size, pkt->data, pkt->size, pkt->flags & AV_PKT_FLAG_KEY)) {
+        if (av_parser_change(ost->parser, ost->st->codec,
+                             &opkt.data, &opkt.size,
+                             pkt->data, pkt->size,
+                             pkt->flags & AV_PKT_FLAG_KEY)) {
            opkt.buf = av_buffer_create(opkt.data, opkt.size, av_buffer_default_free, NULL, 0);
            if (!opkt.buf)
                exit_program(1);
@@ -1461,6 +1585,7 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
        opkt.data = pkt->data;
        opkt.size = pkt->size;
    }
+    av_copy_packet_side_data(&opkt, pkt);

    if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && (of->ctx->oformat->flags & AVFMT_RAWPICTURE)) {
        /* store AVPicture in AVPacket, as expected by the output format */
@@ -1501,7 +1626,7 @@ static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
    int i, ret, err = 0, resample_changed;
    AVRational decoded_frame_tb;

-    if (!ist->decoded_frame && !(ist->decoded_frame = avcodec_alloc_frame()))
+    if (!ist->decoded_frame && !(ist->decoded_frame = av_frame_alloc()))
        return AVERROR(ENOMEM);
    if (!ist->filter_frame && !(ist->filter_frame = av_frame_alloc()))
        return AVERROR(ENOMEM);
@@ -1637,7 +1762,6 @@ static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
 static int decode_video(InputStream *ist, AVPacket *pkt, int *got_output)
 {
    AVFrame *decoded_frame, *f;
-    void *buffer_to_free = NULL;
    int i, ret = 0, err = 0, resample_changed;
    int64_t best_effort_timestamp;
    AVRational *frame_sample_aspect;
@@ -1672,18 +1796,26 @@ static int decode_video(InputStream *ist, AVPacket *pkt, int *got_output)
    if(ist->top_field_first>=0)
        decoded_frame->top_field_first = ist->top_field_first;

+    if (ist->hwaccel_retrieve_data && decoded_frame->format == ist->hwaccel_pix_fmt) {
+        err = ist->hwaccel_retrieve_data(ist->st->codec, decoded_frame);
+        if (err < 0)
+            goto fail;
+    }
+    ist->hwaccel_retrieved_pix_fmt = decoded_frame->format;
+
    best_effort_timestamp= av_frame_get_best_effort_timestamp(decoded_frame);
    if(best_effort_timestamp != AV_NOPTS_VALUE)
        ist->next_pts = ist->pts = av_rescale_q(decoded_frame->pts = best_effort_timestamp, ist->st->time_base, AV_TIME_BASE_Q);

    if (debug_ts) {
        av_log(NULL, AV_LOG_INFO, "decoder -> ist_index:%d type:video "
-                "frame_pts:%s frame_pts_time:%s best_effort_ts:%"PRId64" best_effort_ts_time:%s keyframe:%d frame_type:%d \n",
-                ist->st->index, av_ts2str(decoded_frame->pts),
-                av_ts2timestr(decoded_frame->pts, &ist->st->time_base),
-                best_effort_timestamp,
-                av_ts2timestr(best_effort_timestamp, &ist->st->time_base),
-                decoded_frame->key_frame, decoded_frame->pict_type);
+               "frame_pts:%s frame_pts_time:%s best_effort_ts:%"PRId64" best_effort_ts_time:%s keyframe:%d frame_type:%d time_base:%d/%d\n",
+               ist->st->index, av_ts2str(decoded_frame->pts),
+               av_ts2timestr(decoded_frame->pts, &ist->st->time_base),
+               best_effort_timestamp,
+               av_ts2timestr(best_effort_timestamp, &ist->st->time_base),
+               decoded_frame->key_frame, decoded_frame->pict_type,
+               ist->st->time_base.num, ist->st->time_base.den);
    }

    pkt->size = 0;
@@ -1736,9 +1868,9 @@ static int decode_video(InputStream *ist, AVPacket *pkt, int *got_output)
        }
    }

+fail:
    av_frame_unref(ist->filter_frame);
    av_frame_unref(decoded_frame);
-    av_free(buffer_to_free);
    return err < 0 ? err : ret;
 }

@@ -1758,25 +1890,32 @@ static int transcode_subtitles(InputStream *ist, AVPacket *pkt, int *got_output)
    }

    if (ist->fix_sub_duration) {
+        int end = 1;
        if (ist->prev_sub.got_output) {
-            int end = av_rescale(subtitle.pts - ist->prev_sub.subtitle.pts,
-                                 1000, AV_TIME_BASE);
+            end = av_rescale(subtitle.pts - ist->prev_sub.subtitle.pts,
+                             1000, AV_TIME_BASE);
            if (end < ist->prev_sub.subtitle.end_display_time) {
                av_log(ist->st->codec, AV_LOG_DEBUG,
-                       "Subtitle duration reduced from %d to %d\n",
-                       ist->prev_sub.subtitle.end_display_time, end);
+                       "Subtitle duration reduced from %d to %d%s\n",
+                       ist->prev_sub.subtitle.end_display_time, end,
+                       end <= 0 ? ", dropping it" : "");
                ist->prev_sub.subtitle.end_display_time = end;
            }
        }
        FFSWAP(int,        *got_output, ist->prev_sub.got_output);
        FFSWAP(int,        ret,         ist->prev_sub.ret);
        FFSWAP(AVSubtitle, subtitle,    ist->prev_sub.subtitle);
+        if (end <= 0)
+            goto out;
    }

+    if (!*got_output)
+        return ret;
+
    sub2video_update(ist, &subtitle);

-    if (!*got_output || !subtitle.num_rects)
-        return ret;
+    if (!subtitle.num_rects)
+        goto out;

    for (i = 0; i < nb_output_streams; i++) {
        OutputStream *ost = output_streams[i];
@@ -1787,6 +1926,7 @@ static int transcode_subtitles(InputStream *ist, AVPacket *pkt, int *got_output)
        do_subtitle_out(output_files[ost->file_index]->ctx, ost, ist, &subtitle);
    }

+out:
    avsubtitle_free(&subtitle);
    return ret;
 }
@@ -1852,7 +1992,7 @@ static int output_packet(InputStream *ist, const AVPacket *pkt)
            if (avpkt.duration) {
                duration = av_rescale_q(avpkt.duration, ist->st->time_base, AV_TIME_BASE_Q);
            } else if(ist->st->codec->time_base.num != 0 && ist->st->codec->time_base.den != 0) {
-                int ticks= ist->st->parser ? ist->st->parser->repeat_pict+1 : ist->st->codec->ticks_per_frame;
+                int ticks= av_stream_get_parser(ist->st) ? av_stream_get_parser(ist->st)->repeat_pict+1 : ist->st->codec->ticks_per_frame;
                duration = ((int64_t)AV_TIME_BASE *
                                ist->st->codec->time_base.num * ticks) /
                                ist->st->codec->time_base.den;
@@ -1909,7 +2049,7 @@ static int output_packet(InputStream *ist, const AVPacket *pkt)
            } else if (pkt->duration) {
                ist->next_dts += av_rescale_q(pkt->duration, ist->st->time_base, AV_TIME_BASE_Q);
            } else if(ist->st->codec->time_base.num != 0) {
-                int ticks= ist->st->parser ? ist->st->parser->repeat_pict + 1 : ist->st->codec->ticks_per_frame;
+                int ticks= av_stream_get_parser(ist->st) ? av_stream_get_parser(ist->st)->repeat_pict + 1 : ist->st->codec->ticks_per_frame;
                ist->next_dts += ((int64_t)AV_TIME_BASE *
                                  ist->st->codec->time_base.num * ticks) /
                                  ist->st->codec->time_base.den;
@@ -1948,6 +2088,63 @@ static void print_sdp(void)
    av_freep(&avc);
 }

+static const HWAccel *get_hwaccel(enum AVPixelFormat pix_fmt)
+{
+    int i;
+    for (i = 0; hwaccels[i].name; i++)
+        if (hwaccels[i].pix_fmt == pix_fmt)
+            return &hwaccels[i];
+    return NULL;
+}
+
+static enum AVPixelFormat get_format(AVCodecContext *s, const enum AVPixelFormat *pix_fmts)
+{
+    InputStream *ist = s->opaque;
+    const enum AVPixelFormat *p;
+    int ret;
+
+    for (p = pix_fmts; *p != -1; p++) {
+        const AVPixFmtDescriptor *desc = av_pix_fmt_desc_get(*p);
+        const HWAccel *hwaccel;
+
+        if (!(desc->flags & AV_PIX_FMT_FLAG_HWACCEL))
+            break;
+
+        hwaccel = get_hwaccel(*p);
+        if (!hwaccel ||
+            (ist->active_hwaccel_id && ist->active_hwaccel_id != hwaccel->id) ||
+            (ist->hwaccel_id != HWACCEL_AUTO && ist->hwaccel_id != hwaccel->id))
+            continue;
+
+        ret = hwaccel->init(s);
+        if (ret < 0) {
+            if (ist->hwaccel_id == hwaccel->id) {
+                av_log(NULL, AV_LOG_FATAL,
+                       "%s hwaccel requested for input stream #%d:%d, "
+                       "but cannot be initialized.\n", hwaccel->name,
+                       ist->file_index, ist->st->index);
+                exit_program(1);
+            }
+            continue;
+        }
+        ist->active_hwaccel_id = hwaccel->id;
+        ist->hwaccel_pix_fmt   = *p;
+        break;
+    }
+
+    return *p;
+}
+
+static int get_buffer(AVCodecContext *s, AVFrame *frame, int flags)
+{
+    InputStream *ist = s->opaque;
+
+    if (ist->hwaccel_get_buffer && frame->format == ist->hwaccel_pix_fmt)
+        return ist->hwaccel_get_buffer(s, frame, flags);
+
+    return avcodec_default_get_buffer2(s, frame, flags);
+}
+
 static int init_input_stream(int ist_index, char *error, int error_len)
 {
    int ret;
@@ -1961,21 +2158,23 @@ static int init_input_stream(int ist_index, char *error, int error_len)
            return AVERROR(EINVAL);
        }

+        ist->st->codec->opaque      = ist;
+        ist->st->codec->get_format  = get_format;
+        ist->st->codec->get_buffer2 = get_buffer;
+        ist->st->codec->thread_safe_callbacks = 1;
+
        av_opt_set_int(ist->st->codec, "refcounted_frames", 1, 0);

        if (!av_dict_get(ist->opts, "threads", NULL, 0))
            av_dict_set(&ist->opts, "threads", "auto", 0);
        if ((ret = avcodec_open2(ist->st->codec, codec, &ist->opts)) < 0) {
-            char errbuf[128];
            if (ret == AVERROR_EXPERIMENTAL)
                abort_codec_experimental(codec, 0);

-            av_strerror(ret, errbuf, sizeof(errbuf));
-
            snprintf(error, error_len,
                     "Error while opening decoder for input stream "
                     "#%d:%d : %s",
-                     ist->file_index, ist->st->index, errbuf);
+                     ist->file_index, ist->st->index, av_err2str(ret));
            return ret;
        }
        assert_avoptions(ist->opts);
@@ -1983,7 +2182,6 @@ static int init_input_stream(int ist_index, char *error, int error_len)

    ist->next_pts = AV_NOPTS_VALUE;
    ist->next_dts = AV_NOPTS_VALUE;
-    ist->is_start = 1;

    return 0;
 }
@@ -2089,6 +2287,21 @@ static int transcode_init(void)
    char error[1024];
    int want_sdp = 1;

+    for (i = 0; i < nb_filtergraphs; i++) {
+        FilterGraph *fg = filtergraphs[i];
+        for (j = 0; j < fg->nb_outputs; j++) {
+            OutputFilter *ofilter = fg->outputs[j];
+            if (!ofilter->ost || ofilter->ost->source_index >= 0)
+                continue;
+            if (fg->nb_inputs != 1)
+                continue;
+            for (k = nb_input_streams-1; k >= 0 ; k--)
+                if (fg->inputs[0]->ist == input_streams[k])
+                    break;
+            ofilter->ost->source_index = k;
+        }
+    }
+
    /* init framerate emulation */
    for (i = 0; i < nb_input_files; i++) {
        InputFile *ifile = input_files[i];
@@ -2130,6 +2343,15 @@ static int transcode_init(void)
            ost->st->disposition          = ist->st->disposition;
            codec->bits_per_raw_sample    = icodec->bits_per_raw_sample;
            codec->chroma_sample_location = icodec->chroma_sample_location;
+        } else {
+            for (j=0; j<oc->nb_streams; j++) {
+                AVStream *st = oc->streams[j];
+                if (st != ost->st && st->codec->codec_type == codec->codec_type)
+                    break;
+            }
+            if (j == oc->nb_streams)
+                if (codec->codec_type == AVMEDIA_TYPE_AUDIO || codec->codec_type == AVMEDIA_TYPE_VIDEO)
+                    ost->st->disposition = AV_DISPOSITION_DEFAULT;
        }

        if (ost->stream_copy) {
@@ -2219,6 +2441,8 @@ static int transcode_init(void)
            av_reduce(&codec->time_base.num, &codec->time_base.den,
                        codec->time_base.num, codec->time_base.den, INT_MAX);

+            ost->parser = av_parser_init(codec->codec_id);
+
            switch (codec->codec_type) {
            case AVMEDIA_TYPE_AUDIO:
                if (audio_volume != 256) {
@@ -2231,6 +2455,7 @@ static int transcode_init(void)
                codec->frame_size         = icodec->frame_size;
                codec->audio_service_type = icodec->audio_service_type;
                codec->block_align        = icodec->block_align;
+                codec->delay              = icodec->delay;
                if((codec->block_align == 1 || codec->block_align == 1152 || codec->block_align == 576) && codec->codec_id == AV_CODEC_ID_MP3)
                    codec->block_align= 0;
                if(codec->codec_id == AV_CODEC_ID_AC3)
@@ -2297,12 +2522,25 @@ static int transcode_init(void)
                if (ist && !ost->frame_rate.num)
                    ost->frame_rate = ist->framerate;
                if (ist && !ost->frame_rate.num)
-                    ost->frame_rate = ist->st->r_frame_rate.num ? ist->st->r_frame_rate : (AVRational){25, 1};
+                    ost->frame_rate = ist->st->r_frame_rate;
+                if (ist && !ost->frame_rate.num) {
+                    ost->frame_rate = (AVRational){25, 1};
+                    av_log(NULL, AV_LOG_WARNING,
+                           "No information "
+                           "about the input framerate is available. Falling "
+                           "back to a default value of 25fps for output stream #%d:%d. Use the -r option "
+                           "if you want a different framerate.\n",
+                           ost->file_index, ost->index);
+                }
 //                    ost->frame_rate = ist->st->avg_frame_rate.num ? ist->st->avg_frame_rate : (AVRational){25, 1};
                if (ost->enc && ost->enc->supported_framerates && !ost->force_fps) {
                    int idx = av_find_nearest_q_idx(ost->frame_rate, ost->enc->supported_framerates);
                    ost->frame_rate = ost->enc->supported_framerates[idx];
                }
+                if (codec->codec_id == AV_CODEC_ID_MPEG4) {
+                    av_reduce(&ost->frame_rate.num, &ost->frame_rate.den,
+                              ost->frame_rate.num, ost->frame_rate.den, 65535);
+                }
            }

            switch (codec->codec_type) {
@@ -2318,7 +2556,7 @@ static int transcode_init(void)
                if (ost->filter && !(codec->time_base.num && codec->time_base.den))
                    codec->time_base = ost->filter->filter->inputs[0]->time_base;
                if (   av_q2d(codec->time_base) < 0.001 && video_sync_method != VSYNC_PASSTHROUGH
-                   && (video_sync_method == VSYNC_CFR || (video_sync_method == VSYNC_AUTO && !(oc->oformat->flags & AVFMT_VARIABLE_FPS)))){
+                   && (video_sync_method == VSYNC_CFR || video_sync_method == VSYNC_VSCFR || (video_sync_method == VSYNC_AUTO && !(oc->oformat->flags & AVFMT_VARIABLE_FPS)))){
                    av_log(oc, AV_LOG_WARNING, "Frame rate very high for a muxer not efficiently supporting it.\n"
                                               "Please consider specifying a lower framerate, a different muxer or -vsync 2\n");
                }
@@ -2336,10 +2574,17 @@ static int transcode_init(void)
                if (!strncmp(ost->enc->name, "libx264", 7) &&
                    codec->pix_fmt == AV_PIX_FMT_NONE &&
                    ost->filter->filter->inputs[0]->format != AV_PIX_FMT_YUV420P)
-                    av_log(NULL, AV_LOG_INFO,
+                    av_log(NULL, AV_LOG_WARNING,
                           "No pixel format specified, %s for H.264 encoding chosen.\n"
                           "Use -pix_fmt yuv420p for compatibility with outdated media players.\n",
                           av_get_pix_fmt_name(ost->filter->filter->inputs[0]->format));
+                if (!strncmp(ost->enc->name, "mpeg2video", 10) &&
+                    codec->pix_fmt == AV_PIX_FMT_NONE &&
+                    ost->filter->filter->inputs[0]->format != AV_PIX_FMT_YUV420P)
+                    av_log(NULL, AV_LOG_WARNING,
+                           "No pixel format specified, %s for MPEG-2 encoding chosen.\n"
+                           "Use -pix_fmt yuv420p for compatibility with outdated media players.\n",
+                           av_get_pix_fmt_name(ost->filter->filter->inputs[0]->format));
                codec->pix_fmt = ost->filter->filter->inputs[0]->format;

                if (!icodec ||
@@ -2401,7 +2646,7 @@ static int transcode_init(void)
                        codec->stats_in = logbuffer;
                    }
                    if (codec->flags & CODEC_FLAG_PASS1) {
-                        f = fopen(logfilename, "wb");
+                        f = av_fopen_utf8(logfilename, "wb");
                        if (!f) {
                            av_log(NULL, AV_LOG_FATAL, "Cannot write log file '%s' for pass-1 encoding: %s\n",
                                logfilename, strerror(errno));
@@ -2451,9 +2696,6 @@ static int transcode_init(void)
                av_log(NULL, AV_LOG_WARNING, "The bitrate parameter is set too low."
                                             " It takes bits/s as argument, not kbits/s\n");
            extra_size += ost->st->codec->extradata_size;
-
-            if (ost->st->codec->me_threshold)
-                input_streams[ost->source_index]->st->codec->debug |= FF_DEBUG_MV;
        } else {
            av_opt_set_dict(ost->st->codec, &ost->opts);
        }
@@ -2490,12 +2732,10 @@ static int transcode_init(void)
        oc = output_files[i]->ctx;
        oc->interrupt_callback = int_cb;
        if ((ret = avformat_write_header(oc, &output_files[i]->opts)) < 0) {
-            char errbuf[128];
-            av_strerror(ret, errbuf, sizeof(errbuf));
            snprintf(error, sizeof(error),
                     "Could not write header for output file #%d "
                     "(incorrect codec parameters ?): %s",
-                     i, errbuf);
+                     i, av_err2str(ret));
            ret = AVERROR(EINVAL);
            goto dump_format;
        }
@@ -2659,7 +2899,7 @@ static int check_keyboard_interaction(int64_t cur_time)
        char buf[4096], target[64], command[256], arg[256] = {0};
        double time;
        int k, n = 0;
-        fprintf(stderr, "\nEnter command: <target> <time> <command>[ <argument>]\n");
+        fprintf(stderr, "\nEnter command: <target>|all <time>|-1 <command>[ <argument>]\n");
        i = 0;
        while ((k = read_key()) != '\n' && k != '\r' && i < sizeof(buf)-1)
            if (k > 0)
@@ -2675,7 +2915,10 @@ static int check_keyboard_interaction(int64_t cur_time)
                    if (time < 0) {
                        ret = avfilter_graph_send_command(fg->graph, target, command, arg, buf, sizeof(buf),
                                                          key == 'c' ? AVFILTER_CMD_FLAG_ONE : 0);
-                        fprintf(stderr, "Command reply for stream %d: ret:%d res:%s\n", i, ret, buf);
+                        fprintf(stderr, "Command reply for stream %d: ret:%d res:\n%s", i, ret, buf);
+                    } else if (key == 'c') {
+                        fprintf(stderr, "Queing commands only on filters supporting the specific command is unsupported\n");
+                        ret = AVERROR_PATCHWELCOME;
                    } else {
                        ret = avfilter_graph_queue_command(fg->graph, target, command, arg, 0, time);
                    }
@@ -2712,7 +2955,8 @@ static int check_keyboard_interaction(int64_t cur_time)
                        "?      show this help\n"
                        "+      increase verbosity\n"
                        "-      decrease verbosity\n"
-                        "c      Send command to filtergraph\n"
+                        "c      Send command to first matching filter supporting it\n"
+                        "C      Send/Que command to all matching filters\n"
                        "D      cycle through available debug modes\n"
                        "h      dump packets/hex press to cycle through the 3 states\n"
                        "q      quit\n"
@@ -2745,11 +2989,15 @@ static void *input_thread(void *arg)

        av_dup_packet(&pkt);
        av_fifo_generic_write(f->fifo, &pkt, sizeof(pkt), NULL);
+        pthread_cond_signal(&f->fifo_cond);

        pthread_mutex_unlock(&f->fifo_lock);
    }

+    pthread_mutex_lock(&f->fifo_lock);
    f->finished = 1;
+    pthread_cond_signal(&f->fifo_cond);
+    pthread_mutex_unlock(&f->fifo_lock);
    return NULL;
 }

@@ -2801,6 +3049,10 @@ static int init_input_threads(void)
        if (!(f->fifo = av_fifo_alloc(8*sizeof(AVPacket))))
            return AVERROR(ENOMEM);

+        if (f->ctx->pb ? !f->ctx->pb->seekable :
+            strcmp(f->ctx->iformat->name, "lavfi"))
+            f->non_blocking = 1;
+
        pthread_mutex_init(&f->fifo_lock, NULL);
        pthread_cond_init (&f->fifo_cond, NULL);

@@ -2816,14 +3068,22 @@ static int get_input_packet_mt(InputFile *f, AVPacket *pkt)

    pthread_mutex_lock(&f->fifo_lock);

+    while (1) {
    if (av_fifo_size(f->fifo)) {
        av_fifo_generic_read(f->fifo, pkt, sizeof(*pkt), NULL);
        pthread_cond_signal(&f->fifo_cond);
+        break;
    } else {
-        if (f->finished)
+        if (f->finished) {
            ret = AVERROR_EOF;
-        else
+            break;
+        }
+        if (f->non_blocking) {
            ret = AVERROR(EAGAIN);
+            break;
+        }
+        pthread_cond_wait(&f->fifo_cond, &f->fifo_lock);
+    }
    }

    pthread_mutex_unlock(&f->fifo_lock);
@@ -2911,7 +3171,7 @@ static int process_input(int file_index)

                if (ost->source_index == ifile->ist_index + i &&
                    (ost->stream_copy || ost->enc->type == AVMEDIA_TYPE_SUBTITLE))
-                    close_output_stream(ost);
+                    finish_output_stream(ost);
            }
        }

@@ -3014,22 +3274,21 @@ static int process_input(int file_index)
        int64_t pkt_dts = av_rescale_q(pkt.dts, ist->st->time_base, AV_TIME_BASE_Q);
        int64_t delta   = pkt_dts - ist->next_dts;
        if (is->iformat->flags & AVFMT_TS_DISCONT) {
-        if(delta < -1LL*dts_delta_threshold*AV_TIME_BASE ||
-            (delta > 1LL*dts_delta_threshold*AV_TIME_BASE &&
-                ist->st->codec->codec_type != AVMEDIA_TYPE_SUBTITLE) ||
-            pkt_dts+1<ist->pts){
-            ifile->ts_offset -= delta;
-            av_log(NULL, AV_LOG_DEBUG,
-                   "timestamp discontinuity %"PRId64", new offset= %"PRId64"\n",
-                   delta, ifile->ts_offset);
-            pkt.dts -= av_rescale_q(delta, AV_TIME_BASE_Q, ist->st->time_base);
-            if (pkt.pts != AV_NOPTS_VALUE)
-                pkt.pts -= av_rescale_q(delta, AV_TIME_BASE_Q, ist->st->time_base);
-        }
+            if (delta < -1LL*dts_delta_threshold*AV_TIME_BASE ||
+                (delta > 1LL*dts_delta_threshold*AV_TIME_BASE &&
+                 ist->st->codec->codec_type != AVMEDIA_TYPE_SUBTITLE) ||
+                pkt_dts + AV_TIME_BASE/10 < FFMAX(ist->pts, ist->dts)) {
+                ifile->ts_offset -= delta;
+                av_log(NULL, AV_LOG_DEBUG,
+                       "timestamp discontinuity %"PRId64", new offset= %"PRId64"\n",
+                       delta, ifile->ts_offset);
+                pkt.dts -= av_rescale_q(delta, AV_TIME_BASE_Q, ist->st->time_base);
+                if (pkt.pts != AV_NOPTS_VALUE)
+                    pkt.pts -= av_rescale_q(delta, AV_TIME_BASE_Q, ist->st->time_base);
+            }
        } else {
            if ( delta < -1LL*dts_error_threshold*AV_TIME_BASE ||
-                (delta > 1LL*dts_error_threshold*AV_TIME_BASE && ist->st->codec->codec_type != AVMEDIA_TYPE_SUBTITLE)
-               ) {
+                (delta > 1LL*dts_error_threshold*AV_TIME_BASE && ist->st->codec->codec_type != AVMEDIA_TYPE_SUBTITLE)) {
                av_log(NULL, AV_LOG_WARNING, "DTS %"PRId64", next:%"PRId64" st:%d invalid dropping\n", pkt.dts, ist->next_dts, pkt.stream_index);
                pkt.dts = AV_NOPTS_VALUE;
            }
@@ -3037,8 +3296,7 @@ static int process_input(int file_index)
                int64_t pkt_pts = av_rescale_q(pkt.pts, ist->st->time_base, AV_TIME_BASE_Q);
                delta   = pkt_pts - ist->next_dts;
                if ( delta < -1LL*dts_error_threshold*AV_TIME_BASE ||
-                    (delta > 1LL*dts_error_threshold*AV_TIME_BASE && ist->st->codec->codec_type != AVMEDIA_TYPE_SUBTITLE)
-                   ) {
+                    (delta > 1LL*dts_error_threshold*AV_TIME_BASE && ist->st->codec->codec_type != AVMEDIA_TYPE_SUBTITLE)) {
                    av_log(NULL, AV_LOG_WARNING, "PTS %"PRId64", next:%"PRId64" invalid dropping st:%d\n", pkt.pts, ist->next_dts, pkt.stream_index);
                    pkt.pts = AV_NOPTS_VALUE;
                }
@@ -3062,10 +3320,8 @@ static int process_input(int file_index)

    ret = output_packet(ist, &pkt);
    if (ret < 0) {
-        char buf[128];
-        av_strerror(ret, buf, sizeof(buf));
        av_log(NULL, AV_LOG_ERROR, "Error while decoding stream #%d:%d: %s\n",
-                ist->file_index, ist->st->index, buf);
+               ist->file_index, ist->st->index, av_err2str(ret));
        if (exit_on_error)
            exit_program(1);
    }
@@ -3258,6 +3514,8 @@ static int transcode(void)
        ist = input_streams[i];
        if (ist->decoding_needed) {
            avcodec_close(ist->st->codec);
+            if (ist->hwaccel_uninit)
+                ist->hwaccel_uninit(ist->st->codec);
        }
    }

@@ -3394,9 +3652,9 @@ int main(int argc, char **argv)
    }
    av_log(NULL, AV_LOG_DEBUG, "%"PRIu64" frames successfully decoded, %"PRIu64" decoding errors\n",
           decode_error_stat[0], decode_error_stat[1]);
-    if (2*decode_error_stat[0] < decode_error_stat[1])
-        exit_program(254);
+    if ((decode_error_stat[0] + decode_error_stat[1]) * max_error_rate < decode_error_stat[1])
+        exit_program(69);

-    exit_program(received_nb_signals ? 255 : 0);
-    return 0;
+    exit_program(received_nb_signals ? 255 : main_return_code);
+    return main_return_code;
 }
--- a/ffmpeg.h
+++ b/ffmpeg.h
@@ -51,10 +51,24 @@
 #define VSYNC_PASSTHROUGH 0
 #define VSYNC_CFR         1
 #define VSYNC_VFR         2
+#define VSYNC_VSCFR       0xfe
 #define VSYNC_DROP        0xff

 #define MAX_STREAMS 1024    /* arbitrary sanity check value */

+enum HWAccelID {
+    HWACCEL_NONE = 0,
+    HWACCEL_AUTO,
+    HWACCEL_VDPAU,
+};
+
+typedef struct HWAccel {
+    const char *name;
+    int (*init)(AVCodecContext *s);
+    enum HWAccelID id;
+    enum AVPixelFormat pix_fmt;
+} HWAccel;
+
 /* select an input stream for an output stream */
 typedef struct StreamMap {
    int disabled;           /* 1 is this mapping is disabled by a negative map */
@@ -93,11 +107,16 @@ typedef struct OptionsContext {
    /* input options */
    int64_t input_ts_offset;
    int rate_emu;
+    int accurate_seek;

    SpecifierOpt *ts_scale;
    int        nb_ts_scale;
    SpecifierOpt *dump_attachment;
    int        nb_dump_attachment;
+    SpecifierOpt *hwaccels;
+    int        nb_hwaccels;
+    SpecifierOpt *hwaccel_devices;
+    int        nb_hwaccel_devices;

    /* output options */
    StreamMap *stream_maps;
@@ -152,6 +171,8 @@ typedef struct OptionsContext {
    int        nb_intra_matrices;
    SpecifierOpt *inter_matrices;
    int        nb_inter_matrices;
+    SpecifierOpt *chroma_intra_matrices;
+    int        nb_chroma_intra_matrices;
    SpecifierOpt *top_field_first;
    int        nb_top_field_first;
    SpecifierOpt *metadata_map;
@@ -234,7 +255,6 @@ typedef struct InputStream {
    int64_t filter_in_rescale_delta_last;

    double ts_scale;
-    int is_start;            /* is 1 at the start and after a discontinuity */
    int saw_first_ts;
    int showed_multi_packet_warning;
    AVDictionary *opts;
@@ -273,6 +293,19 @@ typedef struct InputStream {
    int        nb_filters;

    int reinit_filters;
+
+    /* hwaccel options */
+    enum HWAccelID hwaccel_id;
+    char  *hwaccel_device;
+
+    /* hwaccel context */
+    enum HWAccelID active_hwaccel_id;
+    void  *hwaccel_ctx;
+    void (*hwaccel_uninit)(AVCodecContext *s);
+    int  (*hwaccel_get_buffer)(AVCodecContext *s, AVFrame *frame, int flags);
+    int  (*hwaccel_retrieve_data)(AVCodecContext *s, AVFrame *frame);
+    enum AVPixelFormat hwaccel_pix_fmt;
+    enum AVPixelFormat hwaccel_retrieved_pix_fmt;
 } InputStream;

 typedef struct InputFile {
@@ -280,15 +313,20 @@ typedef struct InputFile {
    int eof_reached;      /* true if eof reached */
    int eagain;           /* true if last read attempt returned EAGAIN */
    int ist_index;        /* index of first stream in input_streams */
+    int64_t input_ts_offset;
    int64_t ts_offset;
    int64_t last_ts;
+    int64_t start_time;   /* user-specified start time in AV_TIME_BASE or AV_NOPTS_VALUE */
+    int64_t recording_time;
    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 */
    int rate_emu;
+    int accurate_seek;

 #if HAVE_PTHREADS
    pthread_t thread;           /* thread reading from this file */
+    int non_blocking;           /* reading packets from the thread should not block */
    int finished;               /* the thread has exited */
    int joined;                 /* the thread has been joined */
    pthread_mutex_t fifo_lock;  /* lock for access to fifo */
@@ -308,6 +346,11 @@ enum forced_keyframes_const {

 extern const char *const forced_keyframes_const_names[];

+typedef enum {
+    ENCODER_FINISHED = 1,
+    MUXER_FINISHED = 2,
+} OSTFinished ;
+
 typedef struct OutputStream {
    int file_index;          /* file index */
    int index;               /* stream index in the output file */
@@ -353,13 +396,15 @@ typedef struct OutputStream {

    OutputFilter *filter;
    char *avfilter;
+    char *filters;         ///< filtergraph associated to the -filter option
+    char *filters_script;  ///< filtergraph script associated to the -filter_script option

    int64_t sws_flags;
    AVDictionary *opts;
    AVDictionary *swr_opts;
    AVDictionary *resample_opts;
    char *apad;
-    int finished;        /* no more packets should be written for this stream */
+    OSTFinished finished;        /* no more packets should be written for this stream */
    int unavailable;                     /* true if the steram is unavailable (possibly temporarily) */
    int stream_copy;
    const char *attachment_filename;
@@ -367,6 +412,8 @@ typedef struct OutputStream {
    int copy_prior_start;

    int keep_pix_fmt;
+
+    AVCodecParserContext *parser;
 } OutputStream;

 typedef struct OutputFile {
@@ -416,10 +463,13 @@ extern int qp_hist;
 extern int stdin_interaction;
 extern int frame_bits_per_raw_sample;
 extern AVIOContext *progress_avio;
+extern float max_error_rate;

 extern const AVIOInterruptCB int_cb;

 extern const OptionDef options[];
+extern const HWAccel hwaccels[];
+

 void term_init(void);
 void term_exit(void);
@@ -443,4 +493,6 @@ FilterGraph *init_simple_filtergraph(InputStream *ist, OutputStream *ost);

 int ffmpeg_parse_options(int argc, char **argv);

+int vdpau_init(AVCodecContext *s);
+
 #endif /* FFMPEG_H */
--- a/ffmpeg_filter.c
+++ b/ffmpeg_filter.c
@@ -18,6 +18,8 @@
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

+#include <stdint.h>
+
 #include "ffmpeg.h"

 #include "libavfilter/avfilter.h"
@@ -42,12 +44,15 @@ enum AVPixelFormat choose_pixel_fmt(AVStream *st, AVCodec *codec, enum AVPixelFo
        const AVPixFmtDescriptor *desc = av_pix_fmt_desc_get(target);
        int has_alpha = desc ? desc->nb_components % 2 == 0 : 0;
        enum AVPixelFormat best= AV_PIX_FMT_NONE;
+        const enum AVPixelFormat mjpeg_formats[] = { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUV420P, AV_PIX_FMT_YUV422P, AV_PIX_FMT_NONE };
+        const enum AVPixelFormat ljpeg_formats[] = { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUVJ444P, AV_PIX_FMT_YUV420P,
+                                                     AV_PIX_FMT_YUV422P, AV_PIX_FMT_YUV444P, AV_PIX_FMT_BGRA, AV_PIX_FMT_NONE };
+
        if (st->codec->strict_std_compliance <= FF_COMPLIANCE_UNOFFICIAL) {
            if (st->codec->codec_id == AV_CODEC_ID_MJPEG) {
-                p = (const enum AVPixelFormat[]) { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUV420P, AV_PIX_FMT_YUV422P, AV_PIX_FMT_NONE };
+                p = mjpeg_formats;
            } else if (st->codec->codec_id == AV_CODEC_ID_LJPEG) {
-                p = (const enum AVPixelFormat[]) { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUVJ444P, AV_PIX_FMT_YUV420P,
-                                                 AV_PIX_FMT_YUV422P, AV_PIX_FMT_YUV444P, AV_PIX_FMT_BGRA, AV_PIX_FMT_NONE };
+                p =ljpeg_formats;
            }
        }
        for (; *p != AV_PIX_FMT_NONE; p++) {
@@ -92,6 +97,11 @@ 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,
@@ -276,21 +286,18 @@ 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)
+static int insert_trim(int64_t start_time, int64_t duration,
+                       AVFilterContext **last_filter, int *pad_idx,
+                       const char *filter_name)
 {
-    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];
+    enum AVMediaType type = avfilter_pad_get_type((*last_filter)->output_pads, *pad_idx);
+    const char *name = (type == AVMEDIA_TYPE_VIDEO) ? "trim" : "atrim";
    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)
+    if (duration == INT64_MAX && start_time == AV_NOPTS_VALUE)
        return 0;

    trim = avfilter_get_by_name(name);
@@ -300,18 +307,16 @@ static int insert_trim(OutputStream *ost, AVFilterContext **last_filter, int *pa
        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,
+    if (duration != INT64_MAX) {
+        ret = av_opt_set_int(ctx, "durationi", duration,
                                AV_OPT_SEARCH_CHILDREN);
    }
-    if (ret >= 0 && of->start_time) {
-        ret = av_opt_set_double(ctx, "start", (double)of->start_time / 1e6,
+    if (ret >= 0 && start_time != AV_NOPTS_VALUE) {
+        ret = av_opt_set_int(ctx, "starti", start_time,
                                AV_OPT_SEARCH_CHILDREN);
    }
    if (ret < 0) {
@@ -336,6 +341,7 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter,
 {
    char *pix_fmts;
    OutputStream *ost = ofilter->ost;
+    OutputFile    *of = output_files[ost->file_index];
    AVCodecContext *codec = ost->st->codec;
    AVFilterContext *last_filter = out->filter_ctx;
    int pad_idx = out->pad_idx;
@@ -408,7 +414,10 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter,
        pad_idx = 0;
    }

-    ret = insert_trim(ost, &last_filter, &pad_idx);
+    snprintf(name, sizeof(name), "trim for output stream %d:%d",
+             ost->file_index, ost->index);
+    ret = insert_trim(of->start_time, of->recording_time,
+                      &last_filter, &pad_idx, name);
    if (ret < 0)
        return ret;

@@ -422,9 +431,9 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter,
 static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out)
 {
    OutputStream *ost = ofilter->ost;
+    OutputFile    *of = output_files[ost->file_index];
    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];
@@ -534,7 +543,10 @@ static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter,
        }
    }

-    ret = insert_trim(ost, &last_filter, &pad_idx);
+    snprintf(name, sizeof(name), "trim for output stream %d:%d",
+             ost->file_index, ost->index);
+    ret = insert_trim(of->start_time, of->recording_time,
+                      &last_filter, &pad_idx, name);
    if (ret < 0)
        return ret;

@@ -548,7 +560,7 @@ static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter,
 {                                                                  \
    AVFilterContext *ctx = inout->filter_ctx;                      \
    AVFilterPad *pads = in ? ctx->input_pads  : ctx->output_pads;  \
-    int       nb_pads = in ? ctx->input_count : ctx->output_count; \
+    int       nb_pads = in ? ctx->nb_inputs   : ctx->nb_outputs;   \
    AVIOContext *pb;                                               \
                                                                   \
    if (avio_open_dyn_buf(&pb) < 0)                                \
@@ -612,17 +624,22 @@ static int sub2video_prepare(InputStream *ist)
 static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
                                        AVFilterInOut *in)
 {
-    AVFilterContext *first_filter = in->filter_ctx;
-    AVFilter *filter = avfilter_get_by_name("buffer");
+    AVFilterContext *last_filter;
+    const AVFilter *buffer_filt = avfilter_get_by_name("buffer");
    InputStream *ist = ifilter->ist;
+    InputFile     *f = input_files[ist->file_index];
    AVRational tb = ist->framerate.num ? av_inv_q(ist->framerate) :
                                         ist->st->time_base;
    AVRational fr = ist->framerate;
    AVRational sar;
    AVBPrint args;
    char name[255];
-    int pad_idx = in->pad_idx;
-    int ret;
+    int ret, pad_idx = 0;
+
+    if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot connect video filter to audio input\n");
+        return AVERROR(EINVAL);
+    }

    if (!fr.num)
        fr = av_guess_frame_rate(input_files[ist->file_index]->ctx, ist->st, NULL);
@@ -642,7 +659,8 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
    av_bprintf(&args,
             "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:"
             "pixel_aspect=%d/%d:sws_param=flags=%d", ist->resample_width,
-             ist->resample_height, ist->resample_pix_fmt,
+             ist->resample_height,
+             ist->hwaccel_retrieve_data ? ist->hwaccel_retrieved_pix_fmt : ist->resample_pix_fmt,
             tb.num, tb.den, sar.num, sar.den,
             SWS_BILINEAR + ((ist->st->codec->flags&CODEC_FLAG_BITEXACT) ? SWS_BITEXACT:0));
    if (fr.num && fr.den)
@@ -650,9 +668,10 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
    snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index,
             ist->file_index, ist->st->index);

-    if ((ret = avfilter_graph_create_filter(&ifilter->filter, filter, name,
+    if ((ret = avfilter_graph_create_filter(&ifilter->filter, buffer_filt, name,
                                            args.str, NULL, fg->graph)) < 0)
        return ret;
+    last_filter = ifilter->filter;

    if (ist->framerate.num) {
        AVFilterContext *setpts;
@@ -665,11 +684,10 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
                                                fg->graph)) < 0)
            return ret;

-        if ((ret = avfilter_link(setpts, 0, first_filter, pad_idx)) < 0)
+        if ((ret = avfilter_link(last_filter, 0, setpts, 0)) < 0)
            return ret;

-        first_filter = setpts;
-        pad_idx = 0;
+        last_filter = setpts;
    }

    if (do_deinterlace) {
@@ -683,14 +701,20 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
                                                fg->graph)) < 0)
            return ret;

-        if ((ret = avfilter_link(yadif, 0, first_filter, pad_idx)) < 0)
+        if ((ret = avfilter_link(last_filter, 0, yadif, 0)) < 0)
            return ret;

-        first_filter = yadif;
-        pad_idx = 0;
+        last_filter = yadif;
    }

-    if ((ret = avfilter_link(ifilter->filter, 0, first_filter, pad_idx)) < 0)
+    snprintf(name, sizeof(name), "trim for input stream %d:%d",
+             ist->file_index, ist->st->index);
+    ret = insert_trim(((f->start_time == AV_NOPTS_VALUE) || !f->accurate_seek) ?
+                      AV_NOPTS_VALUE : 0, f->recording_time, &last_filter, &pad_idx, name);
+    if (ret < 0)
+        return ret;
+
+    if ((ret = avfilter_link(last_filter, 0, in->filter_ctx, in->pad_idx)) < 0)
        return ret;
    return 0;
 }
@@ -698,13 +722,18 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
 static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
                                        AVFilterInOut *in)
 {
-    AVFilterContext *first_filter = in->filter_ctx;
-    AVFilter *filter = avfilter_get_by_name("abuffer");
+    AVFilterContext *last_filter;
+    const AVFilter *abuffer_filt = avfilter_get_by_name("abuffer");
    InputStream *ist = ifilter->ist;
-    int pad_idx = in->pad_idx;
+    InputFile     *f = input_files[ist->file_index];
    AVBPrint args;
    char name[255];
-    int ret;
+    int ret, pad_idx = 0;
+
+    if (ist->st->codec->codec_type != AVMEDIA_TYPE_AUDIO) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot connect audio filter to non audio input\n");
+        return AVERROR(EINVAL);
+    }

    av_bprint_init(&args, 0, AV_BPRINT_SIZE_AUTOMATIC);
    av_bprintf(&args, "time_base=%d/%d:sample_rate=%d:sample_fmt=%s",
@@ -719,10 +748,11 @@ static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
    snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index,
             ist->file_index, ist->st->index);

-    if ((ret = avfilter_graph_create_filter(&ifilter->filter, filter,
+    if ((ret = avfilter_graph_create_filter(&ifilter->filter, abuffer_filt,
                                            name, args.str, NULL,
                                            fg->graph)) < 0)
        return ret;
+    last_filter = ifilter->filter;

 #define AUTO_INSERT_FILTER_INPUT(opt_name, filter_name, arg) do {                 \
    AVFilterContext *filt_ctx;                                              \
@@ -738,11 +768,11 @@ static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
    if (ret < 0)                                                            \
        return ret;                                                         \
                                                                            \
-    ret = avfilter_link(filt_ctx, 0, first_filter, pad_idx);                \
+    ret = avfilter_link(last_filter, 0, filt_ctx, 0);                       \
    if (ret < 0)                                                            \
        return ret;                                                         \
                                                                            \
-    first_filter = filt_ctx;                                                  \
+    last_filter = filt_ctx;                                                 \
 } while (0)

    if (audio_sync_method > 0) {
@@ -778,7 +808,15 @@ static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
        snprintf(args, sizeof(args), "%f", audio_volume / 256.);
        AUTO_INSERT_FILTER_INPUT("-vol", "volume", args);
    }
-    if ((ret = avfilter_link(ifilter->filter, 0, first_filter, pad_idx)) < 0)
+
+    snprintf(name, sizeof(name), "trim for input stream %d:%d",
+             ist->file_index, ist->st->index);
+    ret = insert_trim(((f->start_time == AV_NOPTS_VALUE) || !f->accurate_seek) ?
+                      AV_NOPTS_VALUE : 0, f->recording_time, &last_filter, &pad_idx, name);
+    if (ret < 0)
+        return ret;
+
+    if ((ret = avfilter_link(last_filter, 0, in->filter_ctx, in->pad_idx)) < 0)
        return ret;

    return 0;
@@ -790,6 +828,12 @@ static int configure_input_filter(FilterGraph *fg, InputFilter *ifilter,
    av_freep(&ifilter->name);
    DESCRIBE_FILTER_LINK(ifilter, in, 1);

+    if (!ifilter->ist->dec) {
+        av_log(NULL, AV_LOG_ERROR,
+               "No decoder for stream #%d:%d, filtering impossible\n",
+               ifilter->ist->file_index, ifilter->ist->st->index);
+        return AVERROR_DECODER_NOT_FOUND;
+    }
    switch (avfilter_pad_get_type(in->filter_ctx->input_pads, in->pad_idx)) {
    case AVMEDIA_TYPE_VIDEO: return configure_input_video_filter(fg, ifilter, in);
    case AVMEDIA_TYPE_AUDIO: return configure_input_audio_filter(fg, ifilter, in);
@@ -833,6 +877,10 @@ int configure_filtergraph(FilterGraph *fg)
        if (strlen(args))
            args[strlen(args) - 1] = '\0';
        fg->graph->resample_lavr_opts = av_strdup(args);
+
+        e = av_dict_get(ost->opts, "threads", NULL, 0);
+        if (e)
+            av_opt_set(fg->graph, "threads", e->value, 0);
    }

    if ((ret = avfilter_graph_parse2(fg->graph, graph_desc, &inputs, &outputs)) < 0)
--- a/ffmpeg_opt.c
+++ b/ffmpeg_opt.c
@@ -62,6 +62,14 @@
            outvar = o->name[i].u.type;\
    }\
 }
+
+const HWAccel hwaccels[] = {
+#if HAVE_VDPAU_X11
+    { "vdpau", vdpau_init, HWACCEL_VDPAU, AV_PIX_FMT_VDPAU },
+#endif
+    { 0 },
+};
+
 char *vstats_filename;

 float audio_drift_threshold = 0.1;
@@ -84,6 +92,7 @@ int print_stats       = -1;
 int qp_hist           = 0;
 int stdin_interaction = 1;
 int frame_bits_per_raw_sample = 0;
+float max_error_rate  = 2.0/3;


 static int intra_only         = 0;
@@ -95,9 +104,7 @@ static int do_psnr            = 0;
 static int input_sync;
 static int override_ffserver  = 0;

-static int64_t recording_time = INT64_MAX;
-
-static void uninit_options(OptionsContext *o, int is_input)
+static void uninit_options(OptionsContext *o)
 {
    const OptionDef *po = options;
    int i;
@@ -127,28 +134,19 @@ static void uninit_options(OptionsContext *o, int is_input)
    av_freep(&o->audio_channel_maps);
    av_freep(&o->streamid_map);
    av_freep(&o->attachments);
-
-    if (is_input)
-        recording_time = o->recording_time;
-    else
-        recording_time = INT64_MAX;
 }

-static void init_options(OptionsContext *o, int is_input)
+static void init_options(OptionsContext *o)
 {
    memset(o, 0, sizeof(*o));

-    if (!is_input && recording_time != INT64_MAX) {
-        o->recording_time = recording_time;
-        av_log(NULL, AV_LOG_WARNING,
-                "-t is not an input option, keeping it for the next output;"
-                " consider fixing your command line.\n");
-    } else
-        o->recording_time = INT64_MAX;
    o->stop_time = INT64_MAX;
    o->mux_max_delay  = 0.7;
+    o->start_time     = AV_NOPTS_VALUE;
+    o->recording_time = INT64_MAX;
    o->limit_filesize = UINT64_MAX;
    o->chapters_input_file = INT_MAX;
+    o->accurate_seek  = 1;
 }

 /* return a copy of the input with the stream specifiers removed from the keys */
@@ -561,13 +559,14 @@ static AVCodec *choose_decoder(OptionsContext *o, AVFormatContext *s, AVStream *
 static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
 {
    int i;
-    char *next, *codec_tag = NULL;

    for (i = 0; i < ic->nb_streams; i++) {
        AVStream *st = ic->streams[i];
        AVCodecContext *dec = st->codec;
        InputStream *ist = av_mallocz(sizeof(*ist));
-        char *framerate = NULL;
+        char *framerate = NULL, *hwaccel = NULL, *hwaccel_device = NULL;
+        char *codec_tag = NULL;
+        char *next;

        if (!ist)
            exit_program(1);
@@ -603,7 +602,7 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
        case AVMEDIA_TYPE_VIDEO:
            if(!ist->dec)
                ist->dec = avcodec_find_decoder(dec->codec_id);
-            if (dec->lowres) {
+            if (av_codec_get_lowres(dec)) {
                dec->flags |= CODEC_FLAG_EMU_EDGE;
            }

@@ -622,6 +621,41 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
            ist->top_field_first = -1;
            MATCH_PER_STREAM_OPT(top_field_first, i, ist->top_field_first, ic, st);

+            MATCH_PER_STREAM_OPT(hwaccels, str, hwaccel, ic, st);
+            if (hwaccel) {
+                if (!strcmp(hwaccel, "none"))
+                    ist->hwaccel_id = HWACCEL_NONE;
+                else if (!strcmp(hwaccel, "auto"))
+                    ist->hwaccel_id = HWACCEL_AUTO;
+                else {
+                    int i;
+                    for (i = 0; hwaccels[i].name; i++) {
+                        if (!strcmp(hwaccels[i].name, hwaccel)) {
+                            ist->hwaccel_id = hwaccels[i].id;
+                            break;
+                        }
+                    }
+
+                    if (!ist->hwaccel_id) {
+                        av_log(NULL, AV_LOG_FATAL, "Unrecognized hwaccel: %s.\n",
+                               hwaccel);
+                        av_log(NULL, AV_LOG_FATAL, "Supported hwaccels: ");
+                        for (i = 0; hwaccels[i].name; i++)
+                            av_log(NULL, AV_LOG_FATAL, "%s ", hwaccels[i].name);
+                        av_log(NULL, AV_LOG_FATAL, "\n");
+                        exit_program(1);
+                    }
+                }
+            }
+
+            MATCH_PER_STREAM_OPT(hwaccel_devices, str, hwaccel_device, ic, st);
+            if (hwaccel_device) {
+                ist->hwaccel_device = av_strdup(hwaccel_device);
+                if (!ist->hwaccel_device)
+                    exit_program(1);
+            }
+            ist->hwaccel_pix_fmt = AV_PIX_FMT_NONE;
+
            break;
        case AVMEDIA_TYPE_AUDIO:
            ist->guess_layout_max = INT_MAX;
@@ -659,11 +693,15 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)

 static void assert_file_overwrite(const char *filename)
 {
-    if ((!file_overwrite || no_file_overwrite) &&
-        (strchr(filename, ':') == NULL || filename[1] == ':' ||
-         av_strstart(filename, "file:", NULL))) {
-        if (avio_check(filename, 0) == 0) {
-            if (stdin_interaction && (!no_file_overwrite || file_overwrite)) {
+    if (file_overwrite && no_file_overwrite) {
+        fprintf(stderr, "Error, both -y and -n supplied. Exiting.\n");
+        exit_program(1);
+    }
+
+    if (!file_overwrite) {
+        const char *proto_name = avio_find_protocol_name(filename);
+        if (proto_name && !strcmp(proto_name, "file") && avio_check(filename, 0) == 0) {
+            if (stdin_interaction && !no_file_overwrite) {
                fprintf(stderr,"File '%s' already exists. Overwrite ? [y/N] ", filename);
                fflush(stderr);
                term_exit();
@@ -791,6 +829,14 @@ static int open_input_file(OptionsContext *o, const char *filename)
        find_codec_or_die(audio_codec_name   , AVMEDIA_TYPE_AUDIO   , 0)->id : AV_CODEC_ID_NONE;
    ic->subtitle_codec_id= subtitle_codec_name ?
        find_codec_or_die(subtitle_codec_name, AVMEDIA_TYPE_SUBTITLE, 0)->id : AV_CODEC_ID_NONE;
+
+    if (video_codec_name)
+        av_format_set_video_codec   (ic, find_codec_or_die(video_codec_name   , AVMEDIA_TYPE_VIDEO   , 0));
+    if (audio_codec_name)
+        av_format_set_audio_codec   (ic, find_codec_or_die(audio_codec_name   , AVMEDIA_TYPE_AUDIO   , 0));
+    if (subtitle_codec_name)
+        av_format_set_subtitle_codec(ic, find_codec_or_die(subtitle_codec_name, AVMEDIA_TYPE_SUBTITLE, 0));
+
    ic->flags |= AVFMT_FLAG_NONBLOCK;
    ic->interrupt_callback = int_cb;

@@ -819,13 +865,13 @@ static int open_input_file(OptionsContext *o, const char *filename)
        exit_program(1);
    }

-    timestamp = o->start_time;
+    timestamp = (o->start_time == AV_NOPTS_VALUE) ? 0 : o->start_time;
    /* add the stream start time */
    if (ic->start_time != AV_NOPTS_VALUE)
        timestamp += ic->start_time;

    /* if seeking requested, we execute it */
-    if (o->start_time != 0) {
+    if (o->start_time != AV_NOPTS_VALUE) {
        ret = avformat_seek_file(ic, -1, INT64_MIN, timestamp, timestamp, 0);
        if (ret < 0) {
            av_log(NULL, AV_LOG_WARNING, "%s: could not seek to position %0.3f\n",
@@ -847,9 +893,13 @@ static int open_input_file(OptionsContext *o, const char *filename)

    f->ctx        = ic;
    f->ist_index  = nb_input_streams - ic->nb_streams;
+    f->start_time = o->start_time;
+    f->recording_time = o->recording_time;
+    f->input_ts_offset = o->input_ts_offset;
    f->ts_offset  = o->input_ts_offset - (copy_ts ? 0 : timestamp);
    f->nb_streams = ic->nb_streams;
    f->rate_emu   = o->rate_emu;
+    f->accurate_seek = o->accurate_seek;

    /* check if all codec options have been used */
    unused_opts = strip_specifiers(o->g->codec_opts);
@@ -865,8 +915,13 @@ static int open_input_file(OptionsContext *o, const char *filename)
        const AVClass *class = avcodec_get_class();
        const AVOption *option = av_opt_find(&class, e->key, NULL, 0,
                                             AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ);
-        if (!option)
+        const AVClass *fclass = avformat_get_class();
+        const AVOption *foption = av_opt_find(&fclass, e->key, NULL, 0,
+                                             AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ);
+        if (!option || foption)
            continue;
+
+
        if (!(option->flags & AV_OPT_FLAG_DECODING_PARAM)) {
            av_log(NULL, AV_LOG_ERROR, "Codec AVOption %s (%s) specified for "
                   "input file #%d (%s) is not a decoding option.\n", e->key,
@@ -923,14 +978,14 @@ static uint8_t *get_line(AVIOContext *s)

 static int get_preset_file_2(const char *preset_name, const char *codec_name, AVIOContext **s)
 {
-    int i, ret = 1;
+    int i, ret = -1;
    char filename[1000];
    const char *base[3] = { getenv("AVCONV_DATADIR"),
                            getenv("HOME"),
                            AVCONV_DATADIR,
                            };

-    for (i = 0; i < FF_ARRAY_ELEMS(base) && ret; i++) {
+    for (i = 0; i < FF_ARRAY_ELEMS(base) && ret < 0; i++) {
        if (!base[i])
            continue;
        if (codec_name) {
@@ -938,7 +993,7 @@ static int get_preset_file_2(const char *preset_name, const char *codec_name, AV
                     i != 1 ? "" : "/.avconv", codec_name, preset_name);
            ret = avio_open2(s, filename, AVIO_FLAG_READ, &int_cb, NULL);
        }
-        if (ret) {
+        if (ret < 0) {
            snprintf(filename, sizeof(filename), "%s%s/%s.avpreset", base[i],
                     i != 1 ? "" : "/.avconv", preset_name);
            ret = avio_open2(s, filename, AVIO_FLAG_READ, &int_cb, NULL);
@@ -1145,26 +1200,36 @@ static char *get_ost_filters(OptionsContext *o, AVFormatContext *oc,
                             OutputStream *ost)
 {
    AVStream *st = ost->st;
-    char *filter = NULL, *filter_script = NULL;

-    MATCH_PER_STREAM_OPT(filter_scripts, str, filter_script, oc, st);
-    MATCH_PER_STREAM_OPT(filters, str, filter, oc, st);
-
-    if (filter_script && filter) {
+    if (ost->filters_script && ost->filters) {
        av_log(NULL, AV_LOG_ERROR, "Both -filter and -filter_script set for "
               "output stream #%d:%d.\n", nb_output_files, st->index);
        exit_program(1);
    }

-    if (filter_script)
-        return read_file(filter_script);
-    else if (filter)
-        return av_strdup(filter);
+    if (ost->filters_script)
+        return read_file(ost->filters_script);
+    else if (ost->filters)
+        return av_strdup(ost->filters);

    return av_strdup(st->codec->codec_type == AVMEDIA_TYPE_VIDEO ?
                     "null" : "anull");
 }

+static void check_streamcopy_filters(OptionsContext *o, AVFormatContext *oc,
+                                     const OutputStream *ost, enum AVMediaType type)
+{
+    if (ost->filters_script || ost->filters) {
+        av_log(NULL, AV_LOG_ERROR,
+               "%s '%s' was defined for %s output stream %d:%d but codec copy was selected.\n"
+               "Filtering and streamcopy cannot be used together.\n",
+               ost->filters ? "Filtergraph" : "Filtergraph script",
+               ost->filters ? ost->filters : ost->filters_script,
+               av_get_media_type_string(type), ost->file_index, ost->index);
+        exit_program(1);
+    }
+}
+
 static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, int source_index)
 {
    AVStream *st;
@@ -1193,11 +1258,15 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
        ost->frame_aspect_ratio = q;
    }

+    MATCH_PER_STREAM_OPT(filter_scripts, str, ost->filters_script, oc, st);
+    MATCH_PER_STREAM_OPT(filters,        str, ost->filters,        oc, st);
+
    if (!ost->stream_copy) {
        const char *p = NULL;
        char *frame_size = NULL;
        char *frame_pix_fmt = NULL;
        char *intra_matrix = NULL, *inter_matrix = NULL;
+        char *chroma_intra_matrix = NULL;
        int do_pass = 0;
        int i;

@@ -1230,6 +1299,16 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
            }
            parse_matrix_coeffs(video_enc->intra_matrix, intra_matrix);
        }
+        MATCH_PER_STREAM_OPT(chroma_intra_matrices, str, chroma_intra_matrix, oc, st);
+        if (chroma_intra_matrix) {
+            uint16_t *p = av_mallocz(sizeof(*video_enc->chroma_intra_matrix) * 64);
+            if (!p) {
+                av_log(NULL, AV_LOG_FATAL, "Could not allocate memory for intra matrix.\n");
+                exit_program(1);
+            }
+            av_codec_set_chroma_intra_matrix(video_enc, p);
+            parse_matrix_coeffs(p, chroma_intra_matrix);
+        }
        MATCH_PER_STREAM_OPT(inter_matrices, str, inter_matrix, oc, st);
        if (inter_matrix) {
            if (!(video_enc->inter_matrix = av_mallocz(sizeof(*video_enc->inter_matrix) * 64))) {
@@ -1305,6 +1384,9 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
        MATCH_PER_STREAM_OPT(copy_initial_nonkeyframes, i, ost->copy_initial_nonkeyframes, oc ,st);
    }

+    if (ost->stream_copy)
+        check_streamcopy_filters(o, oc, ost, AVMEDIA_TYPE_VIDEO);
+
    return ost;
 }

@@ -1321,6 +1403,9 @@ static OutputStream *new_audio_stream(OptionsContext *o, AVFormatContext *oc, in
    audio_enc = st->codec;
    audio_enc->codec_type = AVMEDIA_TYPE_AUDIO;

+    MATCH_PER_STREAM_OPT(filter_scripts, str, ost->filters_script, oc, st);
+    MATCH_PER_STREAM_OPT(filters,        str, ost->filters,        oc, st);
+
    if (!ost->stream_copy) {
        char *sample_fmt = NULL;

@@ -1358,6 +1443,9 @@ static OutputStream *new_audio_stream(OptionsContext *o, AVFormatContext *oc, in
        }
    }

+    if (ost->stream_copy)
+        check_streamcopy_filters(o, oc, ost, AVMEDIA_TYPE_AUDIO);
+
    return ost;
 }

@@ -1446,7 +1534,8 @@ static int copy_chapters(InputFile *ifile, OutputFile *ofile, int copy_metadata)

    for (i = 0; i < is->nb_chapters; i++) {
        AVChapter *in_ch = is->chapters[i], *out_ch;
-        int64_t ts_off   = av_rescale_q(ofile->start_time - ifile->ts_offset,
+        int64_t start_time = (ofile->start_time == AV_NOPTS_VALUE) ? 0 : ofile->start_time;
+        int64_t ts_off   = av_rescale_q(start_time - ifile->ts_offset,
                                       AV_TIME_BASE_Q, in_ch->time_base);
        int64_t rt       = (ofile->recording_time == INT64_MAX) ? INT64_MAX :
                           av_rescale_q(ofile->recording_time, AV_TIME_BASE_Q, in_ch->time_base);
@@ -1541,6 +1630,18 @@ static void init_output_filter(OutputFilter *ofilter, OptionsContext *o,
        exit_program(1);
    }

+    if (ost->avfilter && (ost->filters || ost->filters_script)) {
+        const char *opt = ost->filters ? "-vf/-af/-filter" : "-filter_script";
+        av_log(NULL, AV_LOG_ERROR,
+               "%s '%s' was specified through the %s option "
+               "for output stream %d:%d, which is fed from a complex filtergraph.\n"
+               "%s and -filter_complex cannot be used together for the same stream.\n",
+               ost->filters ? "Filtergraph" : "Filtergraph script",
+               ost->filters ? ost->filters : ost->filters_script,
+               opt, ost->file_index, ost->index, opt);
+        exit_program(1);
+    }
+
    if (configure_output_filter(ofilter->graph, ofilter, ofilter->out_tmp) < 0) {
        av_log(NULL, AV_LOG_FATAL, "Error configuring filter.\n");
        exit_program(1);
@@ -1581,11 +1682,12 @@ static int open_output_file(OptionsContext *o, const char *filename)
    }

    if (o->stop_time != INT64_MAX && o->recording_time == INT64_MAX) {
-        if (o->stop_time <= o->start_time) {
+        int64_t start_time = o->start_time == AV_NOPTS_VALUE ? 0 : o->start_time;
+        if (o->stop_time <= start_time) {
            av_log(NULL, AV_LOG_WARNING, "-to value smaller than -ss; ignoring -to.\n");
            o->stop_time = INT64_MAX;
        } else {
-            o->recording_time = o->stop_time - o->start_time;
+            o->recording_time = o->stop_time - start_time;
        }
    }

@@ -1681,7 +1783,7 @@ static int open_output_file(OptionsContext *o, const char *filename)
        /* pick the "best" stream of each type */

        /* video: highest resolution */
-        if (!o->video_disable && oc->oformat->video_codec != AV_CODEC_ID_NONE) {
+        if (!o->video_disable && av_guess_codec(oc->oformat, NULL, filename, NULL, AVMEDIA_TYPE_VIDEO) != AV_CODEC_ID_NONE) {
            int area = 0, idx = -1;
            int qcr = avformat_query_codec(oc->oformat, oc->oformat->video_codec, 0);
            for (i = 0; i < nb_input_streams; i++) {
@@ -1703,7 +1805,7 @@ static int open_output_file(OptionsContext *o, const char *filename)
        }

        /* audio: most channels */
-        if (!o->audio_disable && oc->oformat->audio_codec != AV_CODEC_ID_NONE) {
+        if (!o->audio_disable && av_guess_codec(oc->oformat, NULL, filename, NULL, AVMEDIA_TYPE_AUDIO) != AV_CODEC_ID_NONE) {
            int channels = 0, idx = -1;
            for (i = 0; i < nb_input_streams; i++) {
                ist = input_streams[i];
@@ -1719,7 +1821,7 @@ static int open_output_file(OptionsContext *o, const char *filename)

        /* subtitles: pick first */
        MATCH_PER_TYPE_OPT(codec_names, str, subtitle_codec_name, oc, "s");
-        if (!o->subtitle_disable && (oc->oformat->subtitle_codec != AV_CODEC_ID_NONE || subtitle_codec_name)) {
+        if (!o->subtitle_disable && (avcodec_find_encoder(oc->oformat->subtitle_codec) || subtitle_codec_name)) {
            for (i = 0; i < nb_input_streams; i++)
                if (input_streams[i]->st->codec->codec_type == AVMEDIA_TYPE_SUBTITLE) {
                    new_subtitle_stream(o, oc, i);
@@ -2017,7 +2119,8 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
            for (j = 0; j < nb_input_files; j++) {
                for (i = 0; i < input_files[j]->nb_streams; i++) {
                    AVCodecContext *c = input_files[j]->ctx->streams[i]->codec;
-                    if (c->codec_type != AVMEDIA_TYPE_VIDEO)
+                    if (c->codec_type != AVMEDIA_TYPE_VIDEO ||
+                        !c->time_base.num)
                        continue;
                    fr = c->time_base.den * 1000 / c->time_base.num;
                    if (fr == 25000) {
@@ -2047,23 +2150,22 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
        opt_video_codec(o, "c:v", "mpeg1video");
        opt_audio_codec(o, "c:a", "mp2");
        parse_option(o, "f", "vcd", options);
-        av_dict_set(&o->g->codec_opts, "b:v", arg, 0);

        parse_option(o, "s", norm == PAL ? "352x288" : "352x240", options);
        parse_option(o, "r", frame_rates[norm], options);
-        av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", 0);
+        opt_default(NULL, "g", norm == PAL ? "15" : "18");

-        av_dict_set(&o->g->codec_opts, "b:v", "1150000", 0);
-        av_dict_set(&o->g->codec_opts, "maxrate", "1150000", 0);
-        av_dict_set(&o->g->codec_opts, "minrate", "1150000", 0);
-        av_dict_set(&o->g->codec_opts, "bufsize", "327680", 0); // 40*1024*8;
+        opt_default(NULL, "b:v", "1150000");
+        opt_default(NULL, "maxrate", "1150000");
+        opt_default(NULL, "minrate", "1150000");
+        opt_default(NULL, "bufsize", "327680"); // 40*1024*8;

-        av_dict_set(&o->g->codec_opts, "b:a", "224000", 0);
+        opt_default(NULL, "b:a", "224000");
        parse_option(o, "ar", "44100", options);
        parse_option(o, "ac", "2", options);

-        av_dict_set(&o->g->format_opts, "packetsize", "2324", 0);
-        av_dict_set(&o->g->format_opts, "muxrate", "1411200", 0); // 2352 * 75 * 8;
+        opt_default(NULL, "packetsize", "2324");
+        opt_default(NULL, "muxrate", "1411200"); // 2352 * 75 * 8;

        /* We have to offset the PTS, so that it is consistent with the SCR.
           SCR starts at 36000, but the first two packs contain only padding
@@ -2080,18 +2182,18 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
        parse_option(o, "s", norm == PAL ? "480x576" : "480x480", options);
        parse_option(o, "r", frame_rates[norm], options);
        parse_option(o, "pix_fmt", "yuv420p", options);
-        av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", 0);
+        opt_default(NULL, "g", norm == PAL ? "15" : "18");

-        av_dict_set(&o->g->codec_opts, "b:v", "2040000", 0);
-        av_dict_set(&o->g->codec_opts, "maxrate", "2516000", 0);
-        av_dict_set(&o->g->codec_opts, "minrate", "0", 0); // 1145000;
-        av_dict_set(&o->g->codec_opts, "bufsize", "1835008", 0); // 224*1024*8;
-        av_dict_set(&o->g->codec_opts, "scan_offset", "1", 0);
+        opt_default(NULL, "b:v", "2040000");
+        opt_default(NULL, "maxrate", "2516000");
+        opt_default(NULL, "minrate", "0"); // 1145000;
+        opt_default(NULL, "bufsize", "1835008"); // 224*1024*8;
+        opt_default(NULL, "scan_offset", "1");

-        av_dict_set(&o->g->codec_opts, "b:a", "224000", 0);
+        opt_default(NULL, "b:a", "224000");
        parse_option(o, "ar", "44100", options);

-        av_dict_set(&o->g->format_opts, "packetsize", "2324", 0);
+        opt_default(NULL, "packetsize", "2324");

    } else if (!strcmp(arg, "dvd")) {

@@ -2102,17 +2204,17 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
        parse_option(o, "s", norm == PAL ? "720x576" : "720x480", options);
        parse_option(o, "r", frame_rates[norm], options);
        parse_option(o, "pix_fmt", "yuv420p", options);
-        av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", 0);
+        opt_default(NULL, "g", norm == PAL ? "15" : "18");

-        av_dict_set(&o->g->codec_opts, "b:v", "6000000", 0);
-        av_dict_set(&o->g->codec_opts, "maxrate", "9000000", 0);
-        av_dict_set(&o->g->codec_opts, "minrate", "0", 0); // 1500000;
-        av_dict_set(&o->g->codec_opts, "bufsize", "1835008", 0); // 224*1024*8;
+        opt_default(NULL, "b:v", "6000000");
+        opt_default(NULL, "maxrate", "9000000");
+        opt_default(NULL, "minrate", "0"); // 1500000;
+        opt_default(NULL, "bufsize", "1835008"); // 224*1024*8;

-        av_dict_set(&o->g->format_opts, "packetsize", "2048", 0);  // from www.mpucoder.com: DVD sectors contain 2048 bytes of data, this is also the size of one pack.
-        av_dict_set(&o->g->format_opts, "muxrate", "10080000", 0); // from mplex project: data_rate = 1260000. mux_rate = data_rate * 8
+        opt_default(NULL, "packetsize", "2048");  // from www.mpucoder.com: DVD sectors contain 2048 bytes of data, this is also the size of one pack.
+        opt_default(NULL, "muxrate", "10080000"); // from mplex project: data_rate = 1260000. mux_rate = data_rate * 8

-        av_dict_set(&o->g->codec_opts, "b:a", "448000", 0);
+        opt_default(NULL, "b:a", "448000");
        parse_option(o, "ar", "48000", options);

    } else if (!strncmp(arg, "dv", 2)) {
@@ -2131,6 +2233,10 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
        av_log(NULL, AV_LOG_ERROR, "Unknown target: %s\n", arg);
        return AVERROR(EINVAL);
    }
+
+    av_dict_copy(&o->g->codec_opts,  codec_opts, AV_DICT_DONT_OVERWRITE);
+    av_dict_copy(&o->g->format_opts, format_opts, AV_DICT_DONT_OVERWRITE);
+
    return 0;
 }

@@ -2449,7 +2555,9 @@ void show_help_default(const char *opt, const char *arg)
        int flags = AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_ENCODING_PARAM;
        show_help_children(avcodec_get_class(), flags);
        show_help_children(avformat_get_class(), flags);
+#if CONFIG_SWSCALE
        show_help_children(sws_get_class(), flags);
+#endif
        show_help_children(swr_get_class(), AV_OPT_FLAG_AUDIO_PARAM);
        show_help_children(avfilter_get_class(), AV_OPT_FLAG_VIDEO_PARAM | AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM);
    }
@@ -2481,7 +2589,7 @@ static int open_files(OptionGroupList *l, const char *inout,
        OptionGroup *g = &l->groups[i];
        OptionsContext o;

-        init_options(&o, !strcmp(inout, "input"));
+        init_options(&o);
        o.g = g;

        ret = parse_optgroup(&o, g);
@@ -2493,7 +2601,7 @@ static int open_files(OptionGroupList *l, const char *inout,

        av_log(NULL, AV_LOG_DEBUG, "Opening an %s file: %s.\n", inout, g->arg);
        ret = open_file(&o, g->arg);
-        uninit_options(&o, !strcmp(inout, "input"));
+        uninit_options(&o);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Error opening %s file %s.\n",
                   inout, g->arg);
@@ -2578,7 +2686,7 @@ const OptionDef options[] = {
    { "y",              OPT_BOOL,                                    {              &file_overwrite },
        "overwrite output files" },
    { "n",              OPT_BOOL,                                    {              &no_file_overwrite },
-        "do not overwrite output files" },
+        "never overwrite output files" },
    { "c",              HAS_ARG | OPT_STRING | OPT_SPEC |
                        OPT_INPUT | OPT_OUTPUT,                      { .off       = OFFSET(codec_names) },
        "codec name", "codec" },
@@ -2601,7 +2709,8 @@ const OptionDef options[] = {
    { "map_chapters",   HAS_ARG | OPT_INT | OPT_EXPERT | OPT_OFFSET |
                        OPT_OUTPUT,                                  { .off = OFFSET(chapters_input_file) },
        "set chapters mapping", "input_file_index" },
-    { "t",              HAS_ARG | OPT_TIME | OPT_OFFSET | OPT_INPUT | OPT_OUTPUT, { .off = OFFSET(recording_time) },
+    { "t",              HAS_ARG | OPT_TIME | OPT_OFFSET |
+                        OPT_INPUT | OPT_OUTPUT,                      { .off = OFFSET(recording_time) },
        "record or transcode \"duration\" seconds of audio/video",
        "duration" },
    { "to",             HAS_ARG | OPT_TIME | OPT_OFFSET | OPT_OUTPUT,  { .off = OFFSET(stop_time) },
@@ -2611,6 +2720,9 @@ const OptionDef options[] = {
    { "ss",             HAS_ARG | OPT_TIME | OPT_OFFSET |
                        OPT_INPUT | OPT_OUTPUT,                      { .off = OFFSET(start_time) },
        "set the start time offset", "time_off" },
+    { "accurate_seek",  OPT_BOOL | OPT_OFFSET | OPT_EXPERT |
+                        OPT_INPUT,                                   { .off = OFFSET(accurate_seek) },
+        "enable/disable accurate seeking with -ss" },
    { "itsoffset",      HAS_ARG | OPT_TIME | OPT_OFFSET |
                        OPT_EXPERT | OPT_INPUT,                      { .off = OFFSET(input_ts_offset) },
        "set the input ts offset", "time_off" },
@@ -2674,7 +2786,7 @@ const OptionDef options[] = {
    { "frames",         OPT_INT64 | HAS_ARG | OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(max_frames) },
        "set the number of frames to record", "number" },
    { "tag",            OPT_STRING | HAS_ARG | OPT_SPEC |
-                        OPT_EXPERT | OPT_OUTPUT,                     { .off = OFFSET(codec_tags) },
+                        OPT_EXPERT | OPT_OUTPUT | OPT_INPUT,         { .off = OFFSET(codec_tags) },
        "force codec tag/fourcc", "fourcc/tag" },
    { "q",              HAS_ARG | OPT_EXPERT | OPT_DOUBLE |
                        OPT_SPEC | OPT_OUTPUT,                       { .off = OFFSET(qscale) },
@@ -2706,6 +2818,8 @@ const OptionDef options[] = {
        "extract an attachment into a file", "filename" },
    { "debug_ts",       OPT_BOOL | OPT_EXPERT,                       { &debug_ts },
        "print timestamp debugging info" },
+    { "max_error_rate",  HAS_ARG | OPT_FLOAT,                        { &max_error_rate },
+        "maximum error rate", "ratio of errors (0.0: no errors, 1.0: 100% errors) above which ffmpeg returns an error instead of success." },

    /* video options */
    { "vframes",      OPT_VIDEO | HAS_ARG  | OPT_PERFILE | OPT_OUTPUT,           { .func_arg = opt_video_frames },
@@ -2763,6 +2877,9 @@ const OptionDef options[] = {
    { "inter_matrix", OPT_VIDEO | HAS_ARG | OPT_EXPERT  | OPT_STRING | OPT_SPEC |
                      OPT_OUTPUT,                                                { .off = OFFSET(inter_matrices) },
        "specify inter matrix coeffs", "matrix" },
+    { "chroma_intra_matrix", OPT_VIDEO | HAS_ARG | OPT_EXPERT  | OPT_STRING | OPT_SPEC |
+                      OPT_OUTPUT,                                                { .off = OFFSET(chroma_intra_matrices) },
+        "specify intra matrix coeffs", "matrix" },
    { "top",          OPT_VIDEO | HAS_ARG | OPT_EXPERT  | OPT_INT| OPT_SPEC |
                      OPT_INPUT | OPT_OUTPUT,                                    { .off = OFFSET(top_field_first) },
        "top=1/bottom=0/auto=-1 field first", "" },
@@ -2784,6 +2901,12 @@ const OptionDef options[] = {
        "force key frames at specified timestamps", "timestamps" },
    { "b",            OPT_VIDEO | HAS_ARG | OPT_PERFILE | OPT_OUTPUT,            { .func_arg = opt_bitrate },
        "video bitrate (please use -b:v)", "bitrate" },
+    { "hwaccel",          OPT_VIDEO | OPT_STRING | HAS_ARG | OPT_EXPERT |
+                          OPT_SPEC | OPT_INPUT,                                  { .off = OFFSET(hwaccels) },
+        "use HW accelerated decoding", "hwaccel name" },
+    { "hwaccel_device",   OPT_VIDEO | OPT_STRING | HAS_ARG | OPT_EXPERT |
+                          OPT_SPEC | OPT_INPUT,                                  { .off = OFFSET(hwaccel_devices) },
+        "select a device for HW acceleration" "devicename" },

    /* audio options */
    { "aframes",        OPT_AUDIO | HAS_ARG  | OPT_PERFILE | OPT_OUTPUT,           { .func_arg = opt_audio_frames },
--- a/ffmpeg_vdpau.c
+++ b/ffmpeg_vdpau.c
@@ -0,0 +1,335 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <stdint.h>
+
+#include <vdpau/vdpau.h>
+#include <vdpau/vdpau_x11.h>
+
+#include <X11/Xlib.h>
+
+#include "ffmpeg.h"
+
+#include "libavcodec/vdpau.h"
+
+#include "libavutil/avassert.h"
+#include "libavutil/buffer.h"
+#include "libavutil/frame.h"
+#include "libavutil/pixfmt.h"
+
+typedef struct VDPAUContext {
+    Display *dpy;
+
+    VdpDevice  device;
+    VdpDecoder decoder;
+    VdpGetProcAddress *get_proc_address;
+
+    VdpGetErrorString                               *get_error_string;
+    VdpGetInformationString                         *get_information_string;
+    VdpDeviceDestroy                                *device_destroy;
+    VdpDecoderCreate                                *decoder_create;
+    VdpDecoderDestroy                               *decoder_destroy;
+    VdpDecoderRender                                *decoder_render;
+    VdpVideoSurfaceCreate                           *video_surface_create;
+    VdpVideoSurfaceDestroy                          *video_surface_destroy;
+    VdpVideoSurfaceGetBitsYCbCr                     *video_surface_get_bits;
+    VdpVideoSurfaceGetParameters                    *video_surface_get_parameters;
+    VdpVideoSurfaceQueryGetPutBitsYCbCrCapabilities *video_surface_query;
+
+    AVFrame *tmp_frame;
+
+    enum AVPixelFormat pix_fmt;
+    VdpYCbCrFormat vdpau_format;
+} VDPAUContext;
+
+static void vdpau_uninit(AVCodecContext *s)
+{
+    InputStream  *ist = s->opaque;
+    VDPAUContext *ctx = ist->hwaccel_ctx;
+
+    ist->hwaccel_uninit        = NULL;
+    ist->hwaccel_get_buffer    = NULL;
+    ist->hwaccel_retrieve_data = NULL;
+
+    if (ctx->decoder_destroy)
+        ctx->decoder_destroy(ctx->decoder);
+
+    if (ctx->device_destroy)
+        ctx->device_destroy(ctx->device);
+
+    if (ctx->dpy)
+        XCloseDisplay(ctx->dpy);
+
+    av_frame_free(&ctx->tmp_frame);
+
+    av_freep(&ist->hwaccel_ctx);
+    av_freep(&s->hwaccel_context);
+}
+
+static void vdpau_release_buffer(void *opaque, uint8_t *data)
+{
+    VdpVideoSurface surface = *(VdpVideoSurface*)data;
+    VDPAUContext *ctx = opaque;
+
+    ctx->video_surface_destroy(surface);
+    av_freep(&data);
+}
+
+static int vdpau_get_buffer(AVCodecContext *s, AVFrame *frame, int flags)
+{
+    InputStream         *ist = s->opaque;
+    VDPAUContext        *ctx = ist->hwaccel_ctx;
+    VdpVideoSurface *surface;
+    VdpStatus err;
+
+    av_assert0(frame->format == AV_PIX_FMT_VDPAU);
+
+    surface = av_malloc(sizeof(*surface));
+    if (!surface)
+        return AVERROR(ENOMEM);
+
+    frame->buf[0] = av_buffer_create((uint8_t*)surface, sizeof(*surface),
+                                     vdpau_release_buffer, ctx,
+                                     AV_BUFFER_FLAG_READONLY);
+    if (!frame->buf[0]) {
+        av_freep(&surface);
+        return AVERROR(ENOMEM);
+    }
+
+    // properly we should keep a pool of surfaces instead of creating
+    // them anew for each frame, but since we don't care about speed
+    // much in this code, we don't bother
+    err = ctx->video_surface_create(ctx->device, VDP_CHROMA_TYPE_420,
+                                    frame->width, frame->height, surface);
+    if (err != VDP_STATUS_OK) {
+        av_log(NULL, AV_LOG_ERROR, "Error allocating a VDPAU video surface: %s\n",
+               ctx->get_error_string(err));
+        av_buffer_unref(&frame->buf[0]);
+        return AVERROR_UNKNOWN;
+    }
+
+    frame->data[3] = (uint8_t*)(uintptr_t)*surface;
+
+    return 0;
+}
+
+static int vdpau_retrieve_data(AVCodecContext *s, AVFrame *frame)
+{
+    VdpVideoSurface surface = (VdpVideoSurface)(uintptr_t)frame->data[3];
+    InputStream        *ist = s->opaque;
+    VDPAUContext       *ctx = ist->hwaccel_ctx;
+    VdpStatus err;
+    int ret, chroma_type;
+
+    err = ctx->video_surface_get_parameters(surface, &chroma_type,
+                                            &ctx->tmp_frame->width,
+                                            &ctx->tmp_frame->height);
+    if (err != VDP_STATUS_OK) {
+        av_log(NULL, AV_LOG_ERROR, "Error getting surface parameters: %s\n",
+               ctx->get_error_string(err));
+        return AVERROR_UNKNOWN;
+    }
+    ctx->tmp_frame->format = ctx->pix_fmt;
+
+    ret = av_frame_get_buffer(ctx->tmp_frame, 32);
+    if (ret < 0)
+        return ret;
+
+    ctx->tmp_frame->width  = frame->width;
+    ctx->tmp_frame->height = frame->height;
+
+    err = ctx->video_surface_get_bits(surface, ctx->vdpau_format,
+                                      (void * const *)ctx->tmp_frame->data,
+                                      ctx->tmp_frame->linesize);
+    if (err != VDP_STATUS_OK) {
+        av_log(NULL, AV_LOG_ERROR, "Error retrieving frame data from VDPAU: %s\n",
+               ctx->get_error_string(err));
+        ret = AVERROR_UNKNOWN;
+        goto fail;
+    }
+
+    if (ctx->vdpau_format == VDP_YCBCR_FORMAT_YV12)
+        FFSWAP(uint8_t*, ctx->tmp_frame->data[1], ctx->tmp_frame->data[2]);
+
+    ret = av_frame_copy_props(ctx->tmp_frame, frame);
+    if (ret < 0)
+        goto fail;
+
+    av_frame_unref(frame);
+    av_frame_move_ref(frame, ctx->tmp_frame);
+    return 0;
+
+fail:
+    av_frame_unref(ctx->tmp_frame);
+    return ret;
+}
+
+static const int vdpau_formats[][2] = {
+    { VDP_YCBCR_FORMAT_YV12, AV_PIX_FMT_YUV420P },
+    { VDP_YCBCR_FORMAT_NV12, AV_PIX_FMT_NV12 },
+    { VDP_YCBCR_FORMAT_YUYV, AV_PIX_FMT_YUYV422 },
+    { VDP_YCBCR_FORMAT_UYVY, AV_PIX_FMT_UYVY422 },
+};
+
+static int vdpau_alloc(AVCodecContext *s)
+{
+    InputStream  *ist = s->opaque;
+    int loglevel = (ist->hwaccel_id == HWACCEL_AUTO) ? AV_LOG_VERBOSE : AV_LOG_ERROR;
+    AVVDPAUContext *vdpau_ctx;
+    VDPAUContext *ctx;
+    const char *display, *vendor;
+    VdpStatus err;
+    int i;
+
+    ctx = av_mallocz(sizeof(*ctx));
+    if (!ctx)
+        return AVERROR(ENOMEM);
+
+    ist->hwaccel_ctx           = ctx;
+    ist->hwaccel_uninit        = vdpau_uninit;
+    ist->hwaccel_get_buffer    = vdpau_get_buffer;
+    ist->hwaccel_retrieve_data = vdpau_retrieve_data;
+
+    ctx->tmp_frame = av_frame_alloc();
+    if (!ctx->tmp_frame)
+        goto fail;
+
+    ctx->dpy = XOpenDisplay(ist->hwaccel_device);
+    if (!ctx->dpy) {
+        av_log(NULL, loglevel, "Cannot open the X11 display %s.\n",
+               XDisplayName(ist->hwaccel_device));
+        goto fail;
+    }
+    display = XDisplayString(ctx->dpy);
+
+    err = vdp_device_create_x11(ctx->dpy, XDefaultScreen(ctx->dpy), &ctx->device,
+                                &ctx->get_proc_address);
+    if (err != VDP_STATUS_OK) {
+        av_log(NULL, loglevel, "VDPAU device creation on X11 display %s failed.\n",
+               display);
+        goto fail;
+    }
+
+#define GET_CALLBACK(id, result)                                                \
+do {                                                                            \
+    void *tmp;                                                                  \
+    err = ctx->get_proc_address(ctx->device, id, &tmp);                         \
+    if (err != VDP_STATUS_OK) {                                                 \
+        av_log(NULL, loglevel, "Error getting the " #id " callback.\n");        \
+        goto fail;                                                              \
+    }                                                                           \
+    ctx->result = tmp;                                                          \
+} while (0)
+
+    GET_CALLBACK(VDP_FUNC_ID_GET_ERROR_STRING,               get_error_string);
+    GET_CALLBACK(VDP_FUNC_ID_GET_INFORMATION_STRING,         get_information_string);
+    GET_CALLBACK(VDP_FUNC_ID_DEVICE_DESTROY,                 device_destroy);
+    GET_CALLBACK(VDP_FUNC_ID_DECODER_CREATE,                 decoder_create);
+    GET_CALLBACK(VDP_FUNC_ID_DECODER_DESTROY,                decoder_destroy);
+    GET_CALLBACK(VDP_FUNC_ID_DECODER_RENDER,                 decoder_render);
+    GET_CALLBACK(VDP_FUNC_ID_VIDEO_SURFACE_CREATE,           video_surface_create);
+    GET_CALLBACK(VDP_FUNC_ID_VIDEO_SURFACE_DESTROY,          video_surface_destroy);
+    GET_CALLBACK(VDP_FUNC_ID_VIDEO_SURFACE_GET_BITS_Y_CB_CR, video_surface_get_bits);
+    GET_CALLBACK(VDP_FUNC_ID_VIDEO_SURFACE_GET_PARAMETERS,   video_surface_get_parameters);
+    GET_CALLBACK(VDP_FUNC_ID_VIDEO_SURFACE_QUERY_GET_PUT_BITS_Y_CB_CR_CAPABILITIES,
+                 video_surface_query);
+
+    for (i = 0; i < FF_ARRAY_ELEMS(vdpau_formats); i++) {
+        VdpBool supported;
+        err = ctx->video_surface_query(ctx->device, VDP_CHROMA_TYPE_420,
+                                       vdpau_formats[i][0], &supported);
+        if (err != VDP_STATUS_OK) {
+            av_log(NULL, loglevel,
+                   "Error querying VDPAU surface capabilities: %s\n",
+                   ctx->get_error_string(err));
+            goto fail;
+        }
+        if (supported)
+            break;
+    }
+    if (i == FF_ARRAY_ELEMS(vdpau_formats)) {
+        av_log(NULL, loglevel,
+               "No supported VDPAU format for retrieving the data.\n");
+        return AVERROR(EINVAL);
+    }
+    ctx->vdpau_format = vdpau_formats[i][0];
+    ctx->pix_fmt      = vdpau_formats[i][1];
+
+    vdpau_ctx = av_vdpau_alloc_context();
+    if (!vdpau_ctx)
+        goto fail;
+    vdpau_ctx->render = ctx->decoder_render;
+
+    s->hwaccel_context = vdpau_ctx;
+
+    ctx->get_information_string(&vendor);
+    av_log(NULL, AV_LOG_VERBOSE, "Using VDPAU -- %s -- on X11 display %s, "
+           "to decode input stream #%d:%d.\n", vendor,
+           display, ist->file_index, ist->st->index);
+
+    return 0;
+
+fail:
+    av_log(NULL, loglevel, "VDPAU init failed for stream #%d:%d.\n",
+           ist->file_index, ist->st->index);
+    vdpau_uninit(s);
+    return AVERROR(EINVAL);
+}
+
+int vdpau_init(AVCodecContext *s)
+{
+    InputStream *ist = s->opaque;
+    int loglevel = (ist->hwaccel_id == HWACCEL_AUTO) ? AV_LOG_VERBOSE : AV_LOG_ERROR;
+    AVVDPAUContext *vdpau_ctx;
+    VDPAUContext *ctx;
+    VdpStatus err;
+    int profile, ret;
+
+    if (!ist->hwaccel_ctx) {
+        ret = vdpau_alloc(s);
+        if (ret < 0)
+            return ret;
+    }
+    ctx       = ist->hwaccel_ctx;
+    vdpau_ctx = s->hwaccel_context;
+
+    ret = av_vdpau_get_profile(s, &profile);
+    if (ret < 0) {
+        av_log(NULL, loglevel, "No known VDPAU decoder profile for this stream.\n");
+        return AVERROR(EINVAL);
+    }
+
+    if (ctx->decoder)
+        ctx->decoder_destroy(ctx->decoder);
+
+    err = ctx->decoder_create(ctx->device, profile,
+                              s->coded_width, s->coded_height,
+                              16, &ctx->decoder);
+    if (err != VDP_STATUS_OK) {
+        av_log(NULL, loglevel, "Error creating the VDPAU decoder: %s\n",
+               ctx->get_error_string(err));
+        return AVERROR_UNKNOWN;
+    }
+
+    vdpau_ctx->decoder = ctx->decoder;
+
+    ist->hwaccel_get_buffer    = vdpau_get_buffer;
+    ist->hwaccel_retrieve_data = vdpau_retrieve_data;
+
+    return 0;
+}
--- a/ffplay.c
+++ b/ffplay.c
@@ -28,6 +28,8 @@
 #include <math.h>
 #include <limits.h>
 #include <signal.h>
+#include <stdint.h>
+
 #include "libavutil/avstring.h"
 #include "libavutil/colorspace.h"
 #include "libavutil/mathematics.h"
@@ -121,6 +123,7 @@ typedef struct PacketQueue {

 typedef struct VideoPicture {
    double pts;             // presentation timestamp for this picture
+    double duration;        // estimated duration based on frame rate
    int64_t pos;            // byte position in file
    SDL_Overlay *bmp;
    int width, height; /* source height & width */
@@ -142,6 +145,8 @@ typedef struct AudioParams {
    int channels;
    int64_t channel_layout;
    enum AVSampleFormat fmt;
+    int frame_size;
+    int bytes_per_sec;
 } AudioParams;

 typedef struct Clock {
@@ -177,6 +182,8 @@ typedef struct VideoState {
    int read_pause_return;
    AVFormatContext *ic;
    int realtime;
+    int audio_finished;
+    int video_finished;

    Clock audclk;
    Clock vidclk;
@@ -216,6 +223,7 @@ typedef struct VideoState {
    int frame_drops_early;
    int frame_drops_late;
    AVFrame *frame;
+    int64_t audio_frame_next_pts;

    enum ShowMode {
        SHOW_MODE_NONE = -1, SHOW_MODE_VIDEO = 0, SHOW_MODE_WAVES, SHOW_MODE_RDFT, SHOW_MODE_NB
@@ -239,13 +247,8 @@ typedef struct VideoState {
    SDL_cond *subpq_cond;

    double frame_timer;
-    double frame_last_pts;
-    double frame_last_duration;
-    double frame_last_dropped_pts;
    double frame_last_returned_time;
    double frame_last_filter_delay;
-    int64_t frame_last_dropped_pos;
-    int frame_last_dropped_serial;
    int video_stream;
    AVStream *video_st;
    PacketQueue videoq;
@@ -305,7 +308,6 @@ static int workaround_bugs = 1;
 static int fast = 0;
 static int genpts = 0;
 static int lowres = 0;
-static int idct = FF_IDCT_AUTO;
 static int error_concealment = 3;
 static int decoder_reorder_pts = -1;
 static int autoexit;
@@ -357,8 +359,6 @@ int64_t get_valid_channel_layout(int64_t channel_layout, int channels)
        return 0;
 }

-static int packet_queue_put(PacketQueue *q, AVPacket *pkt);
-
 static int packet_queue_put_private(PacketQueue *q, AVPacket *pkt)
 {
    MyAVPacketList *pkt1;
@@ -405,6 +405,16 @@ static int packet_queue_put(PacketQueue *q, AVPacket *pkt)
    return ret;
 }

+static int packet_queue_put_nullpacket(PacketQueue *q, int stream_index)
+{
+    AVPacket pkt1, *pkt = &pkt1;
+    av_init_packet(pkt);
+    pkt->data = NULL;
+    pkt->size = 0;
+    pkt->stream_index = stream_index;
+    return packet_queue_put(q, pkt);
+}
+
 /* packet queue handling */
 static void packet_queue_init(PacketQueue *q)
 {
@@ -773,6 +783,14 @@ static void blend_subrect(AVPicture *dst, const AVSubtitleRect *rect, int imgw,
    }
 }

+static void free_picture(VideoPicture *vp)
+{
+     if (vp->bmp) {
+         SDL_FreeYUVOverlay(vp->bmp);
+         vp->bmp = NULL;
+     }
+}
+
 static void free_subpicture(SubPicture *sp)
 {
    avsubtitle_free(&sp->sub);
@@ -975,7 +993,8 @@ static void video_audio_display(VideoState *s)
                }
                av_rdft_calc(s->rdft, data[ch]);
            }
-            // least efficient way to do this, we should of course directly access it but its more than fast enough
+            /* Least efficient way to do this, we should of course
+             * directly access it but it is more than fast enough. */
            for (y = 0; y < s->height; y++) {
                double w = 1 / sqrt(nb_freq);
                int a = sqrt(w * sqrt(data[0][2 * y + 0] * data[0][2 * y + 0] + data[0][2 * y + 1] * data[0][2 * y + 1]));
@@ -1000,7 +1019,6 @@ static void video_audio_display(VideoState *s)

 static void stream_close(VideoState *is)
 {
-    VideoPicture *vp;
    int i;
    /* XXX: use a special url_shutdown call to abort parse cleanly */
    is->abort_request = 1;
@@ -1010,13 +1028,10 @@ static void stream_close(VideoState *is)
    packet_queue_destroy(&is->subtitleq);

    /* free all pictures */
-    for (i = 0; i < VIDEO_PICTURE_QUEUE_SIZE; i++) {
-        vp = &is->pictq[i];
-        if (vp->bmp) {
-            SDL_FreeYUVOverlay(vp->bmp);
-            vp->bmp = NULL;
-        }
-    }
+    for (i = 0; i < VIDEO_PICTURE_QUEUE_SIZE; i++)
+        free_picture(&is->pictq[i]);
+    for (i = 0; i < SUBPICTURE_QUEUE_SIZE; i++)
+        free_subpicture(&is->subpq[i]);
    SDL_DestroyMutex(is->pictq_mutex);
    SDL_DestroyCond(is->pictq_cond);
    SDL_DestroyMutex(is->subpq_mutex);
@@ -1051,20 +1066,24 @@ static void sigterm_handler(int sig)
    exit(123);
 }

+static void set_default_window_size(VideoPicture *vp)
+{
+    SDL_Rect rect;
+    calculate_display_rect(&rect, 0, 0, INT_MAX, vp->height, vp);
+    default_width  = rect.w;
+    default_height = rect.h;
+}
+
 static int video_open(VideoState *is, int force_set_video_mode, VideoPicture *vp)
 {
    int flags = SDL_HWSURFACE | SDL_ASYNCBLIT | SDL_HWACCEL;
    int w,h;
-    SDL_Rect rect;

    if (is_full_screen) flags |= SDL_FULLSCREEN;
    else                flags |= SDL_RESIZABLE;

-    if (vp && vp->width) {
-        calculate_display_rect(&rect, 0, 0, INT_MAX, vp->height, vp);
-        default_width  = rect.w;
-        default_height = rect.h;
-    }
+    if (vp && vp->width)
+        set_default_window_size(vp);

    if (is_full_screen && fs_screen_width) {
        w = fs_screen_width;
@@ -1275,6 +1294,18 @@ static double compute_target_delay(double delay, VideoState *is)
    return delay;
 }

+static double vp_duration(VideoState *is, VideoPicture *vp, VideoPicture *nextvp) {
+    if (vp->serial == nextvp->serial) {
+        double duration = nextvp->pts - vp->pts;
+        if (isnan(duration) || duration <= 0 || duration > is->max_frame_duration)
+            return vp->duration;
+        else
+            return duration;
+    } else {
+        return 0.0;
+    }
+}
+
 static void pictq_next_picture(VideoState *is) {
    /* update queue size and signal for next picture */
    if (++is->pictq_rindex == VIDEO_PICTURE_QUEUE_SIZE)
@@ -1310,14 +1341,12 @@ static void update_video_pts(VideoState *is, double pts, int64_t pos, int serial
    set_clock(&is->vidclk, pts, serial);
    sync_clock_to_slave(&is->extclk, &is->vidclk);
    is->video_current_pos = pos;
-    is->frame_last_pts = pts;
 }

 /* called to display each frame */
 static void video_refresh(void *opaque, double *remaining_time)
 {
    VideoState *is = opaque;
-    VideoPicture *vp;
    double time;

    SubPicture *sp, *sp2;
@@ -1340,37 +1369,34 @@ static void video_refresh(void *opaque, double *remaining_time)
            redisplay = pictq_prev_picture(is);
 retry:
        if (is->pictq_size == 0) {
-            SDL_LockMutex(is->pictq_mutex);
-            if (is->frame_last_dropped_pts != AV_NOPTS_VALUE && is->frame_last_dropped_pts > is->frame_last_pts) {
-                update_video_pts(is, is->frame_last_dropped_pts, is->frame_last_dropped_pos, is->frame_last_dropped_serial);
-                is->frame_last_dropped_pts = AV_NOPTS_VALUE;
-            }
-            SDL_UnlockMutex(is->pictq_mutex);
            // nothing to do, no picture to display in the queue
        } else {
            double last_duration, duration, delay;
+            VideoPicture *vp, *lastvp;
+
            /* dequeue the picture */
            vp = &is->pictq[is->pictq_rindex];
+            lastvp = &is->pictq[(is->pictq_rindex + VIDEO_PICTURE_QUEUE_SIZE - 1) % VIDEO_PICTURE_QUEUE_SIZE];

            if (vp->serial != is->videoq.serial) {
                pictq_next_picture(is);
+                is->video_current_pos = -1;
                redisplay = 0;
                goto retry;
            }

+            if (lastvp->serial != vp->serial && !redisplay)
+                is->frame_timer = av_gettime() / 1000000.0;
+
            if (is->paused)
                goto display;

            /* compute nominal last_duration */
-            last_duration = vp->pts - is->frame_last_pts;
-            if (!isnan(last_duration) && last_duration > 0 && last_duration < is->max_frame_duration) {
-                /* if duration of the last frame was sane, update last_duration in video state */
-                is->frame_last_duration = last_duration;
-            }
+            last_duration = vp_duration(is, lastvp, vp);
            if (redisplay)
                delay = 0.0;
            else
-                delay = compute_target_delay(is->frame_last_duration, is);
+                delay = compute_target_delay(last_duration, is);

            time= av_gettime()/1000000.0;
            if (time < is->frame_timer + delay && !redisplay) {
@@ -1389,7 +1415,7 @@ retry:

            if (is->pictq_size > 1) {
                VideoPicture *nextvp = &is->pictq[(is->pictq_rindex + 1) % VIDEO_PICTURE_QUEUE_SIZE];
-                duration = nextvp->pts - vp->pts;
+                duration = vp_duration(is, vp, nextvp);
                if(!is->step && (redisplay || framedrop>0 || (framedrop && get_master_sync_type(is) != AV_SYNC_VIDEO_MASTER)) && time > is->frame_timer + duration){
                    if (!redisplay)
                        is->frame_drops_late++;
@@ -1490,8 +1516,7 @@ static void alloc_picture(VideoState *is)

    vp = &is->pictq[is->pictq_windex];

-    if (vp->bmp)
-        SDL_FreeYUVOverlay(vp->bmp);
+    free_picture(vp);

    video_open(is, 0, vp);

@@ -1533,7 +1558,7 @@ static void duplicate_right_border_pixels(SDL_Overlay *bmp) {
    }
 }

-static int queue_picture(VideoState *is, AVFrame *src_frame, double pts, int64_t pos, int serial)
+static int queue_picture(VideoState *is, AVFrame *src_frame, double pts, double duration, int64_t pos, int serial)
 {
    VideoPicture *vp;

@@ -1583,7 +1608,7 @@ static int queue_picture(VideoState *is, AVFrame *src_frame, double pts, int64_t
        }
        /* if the queue is aborted, we have to pop the pending ALLOC event or wait for the allocation to complete */
        if (is->videoq.abort_request && SDL_PeepEvents(&event, 1, SDL_GETEVENT, SDL_EVENTMASK(FF_ALLOC_EVENT)) != 1) {
-            while (!vp->allocated) {
+            while (!vp->allocated && !is->abort_request) {
                SDL_CondWait(is->pictq_cond, is->pictq_mutex);
            }
        }
@@ -1630,6 +1655,7 @@ static int queue_picture(VideoState *is, AVFrame *src_frame, double pts, int64_t
        SDL_UnlockYUVOverlay(vp->bmp);

        vp->pts = pts;
+        vp->duration = duration;
        vp->pos = pos;
        vp->serial = serial;

@@ -1652,24 +1678,15 @@ static int get_video_frame(VideoState *is, AVFrame *frame, AVPacket *pkt, int *s

    if (pkt->data == flush_pkt.data) {
        avcodec_flush_buffers(is->video_st->codec);
-
-        SDL_LockMutex(is->pictq_mutex);
-        // Make sure there are no long delay timers (ideally we should just flush the queue but that's harder)
-        while (is->pictq_size && !is->videoq.abort_request) {
-            SDL_CondWait(is->pictq_cond, is->pictq_mutex);
-        }
-        is->video_current_pos = -1;
-        is->frame_last_pts = AV_NOPTS_VALUE;
-        is->frame_last_duration = 0;
-        is->frame_timer = (double)av_gettime() / 1000000.0;
-        is->frame_last_dropped_pts = AV_NOPTS_VALUE;
-        SDL_UnlockMutex(is->pictq_mutex);
        return 0;
    }

    if(avcodec_decode_video2(is->video_st->codec, frame, &got_picture, pkt) < 0)
        return 0;

+    if (!got_picture && !pkt->data)
+        is->video_finished = *serial;
+
    if (got_picture) {
        int ret = 1;
        double dpts = NAN;
@@ -1688,23 +1705,17 @@ static int get_video_frame(VideoState *is, AVFrame *frame, AVPacket *pkt, int *s
        frame->sample_aspect_ratio = av_guess_sample_aspect_ratio(is->ic, is->video_st, frame);

        if (framedrop>0 || (framedrop && get_master_sync_type(is) != AV_SYNC_VIDEO_MASTER)) {
-            SDL_LockMutex(is->pictq_mutex);
-            if (is->frame_last_pts != AV_NOPTS_VALUE && frame->pts != AV_NOPTS_VALUE) {
-                double clockdiff = get_clock(&is->vidclk) - get_master_clock(is);
-                double ptsdiff = dpts - is->frame_last_pts;
-                if (!isnan(clockdiff) && fabs(clockdiff) < AV_NOSYNC_THRESHOLD &&
-                    !isnan(ptsdiff) && ptsdiff > 0 && ptsdiff < AV_NOSYNC_THRESHOLD &&
-                    clockdiff + ptsdiff - is->frame_last_filter_delay < 0 &&
+            if (frame->pts != AV_NOPTS_VALUE) {
+                double diff = dpts - get_master_clock(is);
+                if (!isnan(diff) && fabs(diff) < AV_NOSYNC_THRESHOLD &&
+                    diff - is->frame_last_filter_delay < 0 &&
+                    *serial == is->vidclk.serial &&
                    is->videoq.nb_packets) {
-                    is->frame_last_dropped_pos = pkt->pos;
-                    is->frame_last_dropped_pts = dpts;
-                    is->frame_last_dropped_serial = *serial;
                    is->frame_drops_early++;
                    av_frame_unref(frame);
                    ret = 0;
                }
            }
-            SDL_UnlockMutex(is->pictq_mutex);
        }

        return ret;
@@ -1716,7 +1727,8 @@ static int get_video_frame(VideoState *is, AVFrame *frame, AVPacket *pkt, int *s
 static int configure_filtergraph(AVFilterGraph *graph, const char *filtergraph,
                                 AVFilterContext *source_ctx, AVFilterContext *sink_ctx)
 {
-    int ret;
+    int ret, i;
+    int nb_filters = graph->nb_filters;
    AVFilterInOut *outputs = NULL, *inputs = NULL;

    if (filtergraph) {
@@ -1744,6 +1756,10 @@ static int configure_filtergraph(AVFilterGraph *graph, const char *filtergraph,
            goto fail;
    }

+    /* Reorder the filters to ensure that inputs of the custom filters are merged first */
+    for (i = 0; i < graph->nb_filters - nb_filters; i++)
+        FFSWAP(AVFilterContext*, graph->filters[i], graph->filters[i + nb_filters]);
+
    ret = avfilter_graph_config(graph, NULL);
 fail:
    avfilter_inout_free(&outputs);
@@ -1814,6 +1830,8 @@ static int configure_audio_filters(VideoState *is, const char *afilters, int for
    int64_t channel_layouts[2] = { 0, -1 };
    int channels[2] = { 0, -1 };
    AVFilterContext *filt_asrc = NULL, *filt_asink = NULL;
+    char aresample_swr_opts[512] = "";
+    AVDictionaryEntry *e = NULL;
    char asrc_args[256];
    int ret;

@@ -1821,6 +1839,12 @@ static int configure_audio_filters(VideoState *is, const char *afilters, int for
    if (!(is->agraph = avfilter_graph_alloc()))
        return AVERROR(ENOMEM);

+    while ((e = av_dict_get(swr_opts, "", e, AV_DICT_IGNORE_SUFFIX)))
+        av_strlcatf(aresample_swr_opts, sizeof(aresample_swr_opts), "%s=%s:", e->key, e->value);
+    if (strlen(aresample_swr_opts))
+        aresample_swr_opts[strlen(aresample_swr_opts)-1] = '\0';
+    av_opt_set(is->agraph, "aresample_swr_opts", aresample_swr_opts, 0);
+
    ret = snprintf(asrc_args, sizeof(asrc_args),
                   "sample_rate=%d:sample_fmt=%s:channels=%d:time_base=%d/%d",
                   is->audio_filter_src.freq, av_get_sample_fmt_name(is->audio_filter_src.fmt),
@@ -1882,8 +1906,11 @@ static int video_thread(void *arg)
    VideoState *is = arg;
    AVFrame *frame = av_frame_alloc();
    double pts;
+    double duration;
    int ret;
    int serial = 0;
+    AVRational tb = is->video_st->time_base;
+    AVRational frame_rate = av_guess_frame_rate(is->ic, is->video_st, NULL);

 #if CONFIG_AVFILTER
    AVFilterGraph *graph = avfilter_graph_alloc();
@@ -1898,7 +1925,6 @@ static int video_thread(void *arg)
        while (is->paused && !is->videoq.abort_request)
            SDL_Delay(10);

-        avcodec_get_frame_defaults(frame);
        av_free_packet(&pkt);

        ret = get_video_frame(is, frame, &pkt, &serial);
@@ -1925,7 +1951,6 @@ static int video_thread(void *arg)
                event.type = FF_QUIT_EVENT;
                event.user.data1 = is;
                SDL_PushEvent(&event);
-                av_free_packet(&pkt);
                goto the_end;
            }
            filt_in  = is->in_video_filter;
@@ -1934,20 +1959,20 @@ static int video_thread(void *arg)
            last_h = frame->height;
            last_format = frame->format;
            last_serial = serial;
+            frame_rate = filt_out->inputs[0]->frame_rate;
        }

        ret = av_buffersrc_add_frame(filt_in, frame);
        if (ret < 0)
            goto the_end;
-        av_frame_unref(frame);
-        avcodec_get_frame_defaults(frame);
-        av_free_packet(&pkt);

        while (ret >= 0) {
            is->frame_last_returned_time = av_gettime() / 1000000.0;

            ret = av_buffersink_get_frame_flags(filt_out, frame, 0);
            if (ret < 0) {
+                if (ret == AVERROR_EOF)
+                    is->video_finished = serial;
                ret = 0;
                break;
            }
@@ -1955,22 +1980,20 @@ static int video_thread(void *arg)
            is->frame_last_filter_delay = av_gettime() / 1000000.0 - is->frame_last_returned_time;
            if (fabs(is->frame_last_filter_delay) > AV_NOSYNC_THRESHOLD / 10.0)
                is->frame_last_filter_delay = 0;
-
-            pts = (frame->pts == AV_NOPTS_VALUE) ? NAN : frame->pts * av_q2d(filt_out->inputs[0]->time_base);
-            ret = queue_picture(is, frame, pts, av_frame_get_pkt_pos(frame), serial);
+            tb = filt_out->inputs[0]->time_base;
+#endif
+            duration = (frame_rate.num && frame_rate.den ? av_q2d((AVRational){frame_rate.den, frame_rate.num}) : 0);
+            pts = (frame->pts == AV_NOPTS_VALUE) ? NAN : frame->pts * av_q2d(tb);
+            ret = queue_picture(is, frame, pts, duration, av_frame_get_pkt_pos(frame), serial);
            av_frame_unref(frame);
+#if CONFIG_AVFILTER
        }
-#else
-        pts = (frame->pts == AV_NOPTS_VALUE) ? NAN : frame->pts * av_q2d(is->video_st->time_base);
-        ret = queue_picture(is, frame, pts, pkt.pos, serial);
-        av_frame_unref(frame);
 #endif

        if (ret < 0)
            goto the_end;
    }
 the_end:
-    avcodec_flush_buffers(is->video_st->codec);
 #if CONFIG_AVFILTER
    avfilter_graph_free(&graph);
 #endif
@@ -2045,6 +2068,8 @@ static int subtitle_thread(void *arg)
            SDL_LockMutex(is->subpq_mutex);
            is->subpq_size++;
            SDL_UnlockMutex(is->subpq_mutex);
+        } else if (got_subtitle) {
+            avsubtitle_free(&sp->sub);
        }
        av_free_packet(pkt);
    }
@@ -2129,8 +2154,6 @@ static int audio_decode_frame(VideoState *is)
    int64_t dec_channel_layout;
    int got_frame;
    av_unused double audio_clock0;
-    int new_packet = 0;
-    int flush_complete = 0;
    int wanted_nb_samples;
    AVRational tb;
    int ret;
@@ -2138,13 +2161,12 @@ static int audio_decode_frame(VideoState *is)

    for (;;) {
        /* NOTE: the audio packet can contain several frames */
-        while (pkt_temp->size > 0 || (!pkt_temp->data && new_packet) || is->audio_buf_frames_pending) {
+        while (pkt_temp->stream_index != -1 || is->audio_buf_frames_pending) {
            if (!is->frame) {
-                if (!(is->frame = avcodec_alloc_frame()))
+                if (!(is->frame = av_frame_alloc()))
                    return AVERROR(ENOMEM);
            } else {
                av_frame_unref(is->frame);
-                avcodec_get_frame_defaults(is->frame);
            }

            if (is->audioq.serial != is->audio_pkt_temp_serial)
@@ -2154,9 +2176,6 @@ static int audio_decode_frame(VideoState *is)
                return -1;

            if (!is->audio_buf_frames_pending) {
-                if (flush_complete)
-                    break;
-                new_packet = 0;
                len1 = avcodec_decode_audio4(dec, is->frame, &got_frame, pkt_temp);
                if (len1 < 0) {
                    /* if error, we skip the frame */
@@ -2164,23 +2183,32 @@ static int audio_decode_frame(VideoState *is)
                    break;
                }

+                pkt_temp->dts =
+                pkt_temp->pts = AV_NOPTS_VALUE;
                pkt_temp->data += len1;
                pkt_temp->size -= len1;
+                if (pkt_temp->data && pkt_temp->size <= 0 || !pkt_temp->data && !got_frame)
+                    pkt_temp->stream_index = -1;
+                if (!pkt_temp->data && !got_frame)
+                    is->audio_finished = is->audio_pkt_temp_serial;

-                if (!got_frame) {
-                    /* stop sending empty packets if the decoder is finished */
-                    if (!pkt_temp->data && dec->codec->capabilities & CODEC_CAP_DELAY)
-                        flush_complete = 1;
+                if (!got_frame)
                    continue;
-                }

                tb = (AVRational){1, is->frame->sample_rate};
                if (is->frame->pts != AV_NOPTS_VALUE)
                    is->frame->pts = av_rescale_q(is->frame->pts, dec->time_base, tb);
                else if (is->frame->pkt_pts != AV_NOPTS_VALUE)
                    is->frame->pts = av_rescale_q(is->frame->pkt_pts, is->audio_st->time_base, tb);
-                if (pkt_temp->pts != AV_NOPTS_VALUE)
-                    pkt_temp->pts += (double) is->frame->nb_samples / is->frame->sample_rate / av_q2d(is->audio_st->time_base);
+                else if (is->audio_frame_next_pts != AV_NOPTS_VALUE)
+#if CONFIG_AVFILTER
+                    is->frame->pts = av_rescale_q(is->audio_frame_next_pts, (AVRational){1, is->audio_filter_src.freq}, tb);
+#else
+                    is->frame->pts = av_rescale_q(is->audio_frame_next_pts, (AVRational){1, is->audio_src.freq}, tb);
+#endif
+
+                if (is->frame->pts != AV_NOPTS_VALUE)
+                    is->audio_frame_next_pts = is->frame->pts + is->frame->nb_samples;

 #if CONFIG_AVFILTER
                dec_channel_layout = get_valid_channel_layout(is->frame->channel_layout, av_frame_get_channels(is->frame));
@@ -2213,7 +2241,6 @@ static int audio_decode_frame(VideoState *is)

                if ((ret = av_buffersrc_add_frame(is->in_audio_filter, is->frame)) < 0)
                    return ret;
-                av_frame_unref(is->frame);
 #endif
            }
 #if CONFIG_AVFILTER
@@ -2222,6 +2249,8 @@ static int audio_decode_frame(VideoState *is)
                    is->audio_buf_frames_pending = 0;
                    continue;
                }
+                if (ret == AVERROR_EOF)
+                    is->audio_finished = is->audio_pkt_temp_serial;
                return ret;
            }
            is->audio_buf_frames_pending = 1;
@@ -2318,6 +2347,7 @@ static int audio_decode_frame(VideoState *is)
        if (pkt->data)
            av_free_packet(pkt);
        memset(pkt_temp, 0, sizeof(*pkt_temp));
+        pkt_temp->stream_index = -1;

        if (is->audioq.abort_request) {
            return -1;
@@ -2327,13 +2357,15 @@ static int audio_decode_frame(VideoState *is)
            SDL_CondSignal(is->continue_read_thread);

        /* read next packet */
-        if ((new_packet = packet_queue_get(&is->audioq, pkt, 1, &is->audio_pkt_temp_serial)) < 0)
+        if ((packet_queue_get(&is->audioq, pkt, 1, &is->audio_pkt_temp_serial)) < 0)
            return -1;

        if (pkt->data == flush_pkt.data) {
            avcodec_flush_buffers(dec);
-            flush_complete = 0;
            is->audio_buf_frames_pending = 0;
+            is->audio_frame_next_pts = AV_NOPTS_VALUE;
+            if ((is->ic->iformat->flags & (AVFMT_NOBINSEARCH | AVFMT_NOGENSEARCH | AVFMT_NO_BYTE_SEEK)) && !is->ic->iformat->read_seek)
+                is->audio_frame_next_pts = is->audio_st->start_time;
        }

        *pkt_temp = *pkt;
@@ -2345,8 +2377,6 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
 {
    VideoState *is = opaque;
    int audio_size, len1;
-    int bytes_per_sec;
-    int frame_size = av_samples_get_buffer_size(NULL, is->audio_tgt.channels, 1, is->audio_tgt.fmt, 1);

    audio_callback_time = av_gettime();

@@ -2356,7 +2386,7 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
           if (audio_size < 0) {
                /* if error, just output silence */
               is->audio_buf      = is->silence_buf;
-               is->audio_buf_size = sizeof(is->silence_buf) / frame_size * frame_size;
+               is->audio_buf_size = sizeof(is->silence_buf) / is->audio_tgt.frame_size * is->audio_tgt.frame_size;
           } else {
               if (is->show_mode != SHOW_MODE_VIDEO)
                   update_sample_display(is, (int16_t *)is->audio_buf, audio_size);
@@ -2372,11 +2402,10 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
        stream += len1;
        is->audio_buf_index += len1;
    }
-    bytes_per_sec = is->audio_tgt.freq * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt);
    is->audio_write_buf_size = is->audio_buf_size - is->audio_buf_index;
    /* Let's assume the audio driver that is used by SDL has two periods. */
    if (!isnan(is->audio_clock)) {
-        set_clock_at(&is->audclk, is->audio_clock - (double)(2 * is->audio_hw_buf_size + is->audio_write_buf_size) / bytes_per_sec, is->audio_clock_serial, audio_callback_time / 1000000.0);
+        set_clock_at(&is->audclk, is->audio_clock - (double)(2 * is->audio_hw_buf_size + is->audio_write_buf_size) / is->audio_tgt.bytes_per_sec, is->audio_clock_serial, audio_callback_time / 1000000.0);
        sync_clock_to_slave(&is->extclk, &is->audclk);
    }
 }
@@ -2385,7 +2414,7 @@ static int audio_open(void *opaque, int64_t wanted_channel_layout, int wanted_nb
 {
    SDL_AudioSpec wanted_spec, spec;
    const char *env;
-    const int next_nb_channels[] = {0, 0, 1, 6, 2, 6, 4, 6};
+    static const int next_nb_channels[] = {0, 0, 1, 6, 2, 6, 4, 6};

    env = SDL_getenv("SDL_AUDIO_CHANNELS");
    if (env) {
@@ -2435,6 +2464,12 @@ static int audio_open(void *opaque, int64_t wanted_channel_layout, int wanted_nb
    audio_hw_params->freq = spec.freq;
    audio_hw_params->channel_layout = wanted_channel_layout;
    audio_hw_params->channels =  spec.channels;
+    audio_hw_params->frame_size = av_samples_get_buffer_size(NULL, audio_hw_params->channels, 1, audio_hw_params->fmt, 1);
+    audio_hw_params->bytes_per_sec = av_samples_get_buffer_size(NULL, audio_hw_params->channels, audio_hw_params->freq, audio_hw_params->fmt, 1);
+    if (audio_hw_params->bytes_per_sec <= 0 || audio_hw_params->frame_size <= 0) {
+        av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size failed\n");
+        return -1;
+    }
    return spec.size;
 }

@@ -2450,6 +2485,7 @@ static int stream_component_open(VideoState *is, int stream_index)
    int sample_rate, nb_channels;
    int64_t channel_layout;
    int ret;
+    int stream_lowres = lowres;

    if (stream_index < 0 || stream_index >= ic->nb_streams)
        return -1;
@@ -2474,16 +2510,15 @@ static int stream_component_open(VideoState *is, int stream_index)

    avctx->codec_id = codec->id;
    avctx->workaround_bugs   = workaround_bugs;
-    avctx->lowres            = lowres;
-    if(avctx->lowres > codec->max_lowres){
+    if(stream_lowres > av_codec_get_max_lowres(codec)){
        av_log(avctx, AV_LOG_WARNING, "The maximum value for lowres supported by the decoder is %d\n",
-                codec->max_lowres);
-        avctx->lowres= codec->max_lowres;
+                av_codec_get_max_lowres(codec));
+        stream_lowres = av_codec_get_max_lowres(codec);
    }
-    avctx->idct_algo         = idct;
+    av_codec_set_lowres(avctx, stream_lowres);
    avctx->error_concealment = error_concealment;

-    if(avctx->lowres) avctx->flags |= CODEC_FLAG_EMU_EDGE;
+    if(stream_lowres) avctx->flags |= CODEC_FLAG_EMU_EDGE;
    if (fast)   avctx->flags2 |= CODEC_FLAG2_FAST;
    if(codec->capabilities & CODEC_CAP_DR1)
        avctx->flags |= CODEC_FLAG_EMU_EDGE;
@@ -2491,8 +2526,8 @@ static int stream_component_open(VideoState *is, int stream_index)
    opts = filter_codec_opts(codec_opts, avctx->codec_id, ic, ic->streams[stream_index], codec);
    if (!av_dict_get(opts, "threads", NULL, 0))
        av_dict_set(&opts, "threads", "auto", 0);
-    if (avctx->lowres)
-        av_dict_set(&opts, "lowres", av_asprintf("%d", avctx->lowres), AV_DICT_DONT_STRDUP_VAL);
+    if (stream_lowres)
+        av_dict_set(&opts, "lowres", av_asprintf("%d", stream_lowres), AV_DICT_DONT_STRDUP_VAL);
    if (avctx->codec_type == AVMEDIA_TYPE_VIDEO || avctx->codec_type == AVMEDIA_TYPE_AUDIO)
        av_dict_set(&opts, "refcounted_frames", "1", 0);
    if (avcodec_open2(avctx, codec, &opts) < 0)
@@ -2539,10 +2574,11 @@ static int stream_component_open(VideoState *is, int stream_index)
        is->audio_diff_avg_count = 0;
        /* since we do not have a precise anough audio fifo fullness,
           we correct audio sync only if larger than this threshold */
-        is->audio_diff_threshold = 2.0 * is->audio_hw_buf_size / av_samples_get_buffer_size(NULL, is->audio_tgt.channels, is->audio_tgt.freq, is->audio_tgt.fmt, 1);
+        is->audio_diff_threshold = 2.0 * is->audio_hw_buf_size / is->audio_tgt.bytes_per_sec;

        memset(&is->audio_pkt, 0, sizeof(is->audio_pkt));
        memset(&is->audio_pkt_temp, 0, sizeof(is->audio_pkt_temp));
+        is->audio_pkt_temp.stream_index = -1;

        is->audio_stream = stream_index;
        is->audio_st = ic->streams[stream_index];
@@ -2783,6 +2819,13 @@ static int read_thread(void *arg)
    }

    is->show_mode = show_mode;
+    if (st_index[AVMEDIA_TYPE_VIDEO] >= 0) {
+        AVStream *st = ic->streams[st_index[AVMEDIA_TYPE_VIDEO]];
+        AVCodecContext *avctx = st->codec;
+        VideoPicture vp = {.width = avctx->width, .height = avctx->height, .sar = av_guess_sample_aspect_ratio(ic, st, NULL)};
+        if (vp.width)
+            set_default_window_size(&vp);
+    }

    /* open the streams */
    if (st_index[AVMEDIA_TYPE_AUDIO] >= 0) {
@@ -2801,7 +2844,8 @@ static int read_thread(void *arg)
    }

    if (is->video_stream < 0 && is->audio_stream < 0) {
-        av_log(NULL, AV_LOG_FATAL, "%s: could not open codecs\n", is->filename);
+        av_log(NULL, AV_LOG_FATAL, "Failed to open file '%s' or configure filtergraph\n",
+               is->filename);
        ret = -1;
        goto fail;
    }
@@ -2871,6 +2915,7 @@ static int read_thread(void *arg)
                if ((ret = av_copy_packet(&copy, &is->video_st->attached_pic)) < 0)
                    goto fail;
                packet_queue_put(&is->videoq, &copy);
+                packet_queue_put_nullpacket(&is->videoq, is->video_stream);
            }
            is->queue_attachments_req = 0;
        }
@@ -2888,30 +2933,24 @@ static int read_thread(void *arg)
            SDL_UnlockMutex(wait_mutex);
            continue;
        }
+        if (!is->paused &&
+            (!is->audio_st || is->audio_finished == is->audioq.serial) &&
+            (!is->video_st || (is->video_finished == is->videoq.serial && is->pictq_size == 0))) {
+            if (loop != 1 && (!loop || --loop)) {
+                stream_seek(is, start_time != AV_NOPTS_VALUE ? start_time : 0, 0, 0);
+            } else if (autoexit) {
+                ret = AVERROR_EOF;
+                goto fail;
+            }
+        }
        if (eof) {
-            if (is->video_stream >= 0) {
-                av_init_packet(pkt);
-                pkt->data = NULL;
-                pkt->size = 0;
-                pkt->stream_index = is->video_stream;
-                packet_queue_put(&is->videoq, pkt);
-            }
-            if (is->audio_stream >= 0) {
-                av_init_packet(pkt);
-                pkt->data = NULL;
-                pkt->size = 0;
-                pkt->stream_index = is->audio_stream;
-                packet_queue_put(&is->audioq, pkt);
-            }
+            if (is->video_stream >= 0)
+                packet_queue_put_nullpacket(&is->videoq, is->video_stream);
+            if (is->audio_stream >= 0)
+                packet_queue_put_nullpacket(&is->audioq, is->audio_stream);
+            if (is->subtitle_stream >= 0)
+                packet_queue_put_nullpacket(&is->subtitleq, is->subtitle_stream);
            SDL_Delay(10);
-            if (is->audioq.size + is->videoq.size + is->subtitleq.size == 0) {
-                if (loop != 1 && (!loop || --loop)) {
-                    stream_seek(is, start_time != AV_NOPTS_VALUE ? start_time : 0, 0, 0);
-                } else if (autoexit) {
-                    ret = AVERROR_EOF;
-                    goto fail;
-                }
-            }
            eof=0;
            continue;
        }
@@ -3018,6 +3057,8 @@ static void stream_cycle_channel(VideoState *is, int codec_type)
    int start_index, stream_index;
    int old_index;
    AVStream *st;
+    AVProgram *p = NULL;
+    int nb_streams = is->ic->nb_streams;

    if (codec_type == AVMEDIA_TYPE_VIDEO) {
        start_index = is->last_video_stream;
@@ -3030,8 +3071,22 @@ static void stream_cycle_channel(VideoState *is, int codec_type)
        old_index = is->subtitle_stream;
    }
    stream_index = start_index;
+
+    if (codec_type != AVMEDIA_TYPE_VIDEO && is->video_stream != -1) {
+        p = av_find_program_from_stream(ic, NULL, is->video_stream);
+        if (p) {
+            nb_streams = p->nb_stream_indexes;
+            for (start_index = 0; start_index < nb_streams; start_index++)
+                if (p->stream_index[start_index] == stream_index)
+                    break;
+            if (start_index == nb_streams)
+                start_index = -1;
+            stream_index = start_index;
+        }
+    }
+
    for (;;) {
-        if (++stream_index >= is->ic->nb_streams)
+        if (++stream_index >= nb_streams)
        {
            if (codec_type == AVMEDIA_TYPE_SUBTITLE)
            {
@@ -3045,7 +3100,7 @@ static void stream_cycle_channel(VideoState *is, int codec_type)
        }
        if (stream_index == start_index)
            return;
-        st = ic->streams[stream_index];
+        st = is->ic->streams[p ? p->stream_index[stream_index] : stream_index];
        if (st->codec->codec_type == codec_type) {
            /* check that parameters are OK */
            switch (codec_type) {
@@ -3063,6 +3118,8 @@ static void stream_cycle_channel(VideoState *is, int codec_type)
        }
    }
 the_end:
+    if (p && stream_index != -1)
+        stream_index = p->stream_index[stream_index];
    stream_component_close(is, old_index);
    stream_component_open(is, stream_index);
 }
@@ -3113,6 +3170,33 @@ static void refresh_loop_wait_event(VideoState *is, SDL_Event *event) {
    }
 }

+static void seek_chapter(VideoState *is, int incr)
+{
+    int64_t pos = get_master_clock(is) * AV_TIME_BASE;
+    int i;
+
+    if (!is->ic->nb_chapters)
+        return;
+
+    /* find the current chapter */
+    for (i = 0; i < is->ic->nb_chapters; i++) {
+        AVChapter *ch = is->ic->chapters[i];
+        if (av_compare_ts(pos, AV_TIME_BASE_Q, ch->start, ch->time_base) < 0) {
+            i--;
+            break;
+        }
+    }
+
+    i += incr;
+    i = FFMAX(i, 0);
+    if (i >= is->ic->nb_chapters)
+        return;
+
+    av_log(NULL, AV_LOG_VERBOSE, "Seeking to chapter %d.\n", i);
+    stream_seek(is, av_rescale_q(is->ic->chapters[i]->start, is->ic->chapters[i]->time_base,
+                                 AV_TIME_BASE_Q), 0, 0);
+}
+
 /* handle an event sent by the GUI */
 static void event_loop(VideoState *cur_stream)
 {
@@ -3150,6 +3234,11 @@ static void event_loop(VideoState *cur_stream)
            case SDLK_v:
                stream_cycle_channel(cur_stream, AVMEDIA_TYPE_VIDEO);
                break;
+            case SDLK_c:
+                stream_cycle_channel(cur_stream, AVMEDIA_TYPE_VIDEO);
+                stream_cycle_channel(cur_stream, AVMEDIA_TYPE_AUDIO);
+                stream_cycle_channel(cur_stream, AVMEDIA_TYPE_SUBTITLE);
+                break;
            case SDLK_t:
                stream_cycle_channel(cur_stream, AVMEDIA_TYPE_SUBTITLE);
                break;
@@ -3157,11 +3246,19 @@ static void event_loop(VideoState *cur_stream)
                toggle_audio_display(cur_stream);
                break;
            case SDLK_PAGEUP:
-                incr = 600.0;
-                goto do_seek;
+                if (cur_stream->ic->nb_chapters <= 1) {
+                    incr = 600.0;
+                    goto do_seek;
+                }
+                seek_chapter(cur_stream, 1);
+                break;
            case SDLK_PAGEDOWN:
-                incr = -600.0;
-                goto do_seek;
+                if (cur_stream->ic->nb_chapters <= 1) {
+                    incr = -600.0;
+                    goto do_seek;
+                }
+                seek_chapter(cur_stream, -1);
+                break;
            case SDLK_LEFT:
                incr = -10.0;
                goto do_seek;
@@ -3402,7 +3499,6 @@ static const OptionDef options[] = {
    { "genpts", OPT_BOOL | OPT_EXPERT, { &genpts }, "generate pts", "" },
    { "drp", OPT_INT | HAS_ARG | OPT_EXPERT, { &decoder_reorder_pts }, "let decoder reorder pts 0=off 1=on -1=auto", ""},
    { "lowres", OPT_INT | HAS_ARG | OPT_EXPERT, { &lowres }, "", "" },
-    { "idct", OPT_INT | HAS_ARG | OPT_EXPERT, { &idct }, "set idct algo",  "algo" },
    { "ec", OPT_INT | HAS_ARG | OPT_EXPERT, { &error_concealment }, "set error concealment options",  "bit_mask" },
    { "sync", HAS_ARG | OPT_EXPERT, { .func_arg = opt_sync }, "set audio-video sync. type (type=audio/video/ext)", "type" },
    { "autoexit", OPT_BOOL | OPT_EXPERT, { &autoexit }, "exit at the end", "" },
@@ -3452,9 +3548,10 @@ void show_help_default(const char *opt, const char *arg)
           "q, ESC              quit\n"
           "f                   toggle full screen\n"
           "p, SPC              pause\n"
-           "a                   cycle audio channel\n"
+           "a                   cycle audio channel in the current program\n"
           "v                   cycle video channel\n"
-           "t                   cycle subtitle channel\n"
+           "t                   cycle subtitle channel in the current program\n"
+           "c                   cycle program\n"
           "w                   show audio waves\n"
           "s                   activate frame-step mode\n"
           "left/right          seek backward/forward 10 seconds\n"
@@ -3494,7 +3591,6 @@ int main(int argc, char **argv)
    parse_loglevel(argc, argv, options);

    /* register all codecs, demux and protocols */
-    avcodec_register_all();
 #if CONFIG_AVDEVICE
    avdevice_register_all();
 #endif
--- a/ffprobe.c
+++ b/ffprobe.c
--- a/ffserver.c
+++ b/ffserver.c
@@ -36,6 +36,7 @@
 #include "libavformat/network.h"
 #include "libavformat/os_support.h"
 #include "libavformat/rtpdec.h"
+#include "libavformat/rtpproto.h"
 #include "libavformat/rtsp.h"
 #include "libavformat/avio_internal.h"
 #include "libavformat/internal.h"
@@ -47,6 +48,7 @@
 #include "libavutil/dict.h"
 #include "libavutil/intreadwrite.h"
 #include "libavutil/mathematics.h"
+#include "libavutil/pixdesc.h"
 #include "libavutil/random_seed.h"
 #include "libavutil/parseutils.h"
 #include "libavutil/opt.h"
@@ -63,9 +65,6 @@
 #include <time.h>
 #include <sys/wait.h>
 #include <signal.h>
-#if HAVE_DLFCN_H
-#include <dlfcn.h>
-#endif

 #include "cmdutils.h"

@@ -217,6 +216,7 @@ typedef struct FFStream {
    struct FFStream *feed;   /* feed we are using (can be null if
                                coming from file) */
    AVDictionary *in_opts;   /* input parameters */
+    AVDictionary *metadata;  /* metadata to set on the stream */
    AVInputFormat *ifmt;       /* if non NULL, force input format */
    AVOutputFormat *fmt;
    IPAddressACL *acl;
@@ -229,10 +229,6 @@ typedef struct FFStream {
    int feed_streams[MAX_STREAMS]; /* index of streams in the feed */
    char feed_filename[1024]; /* file name of the feed storage, or
                                 input file name for a stream */
-    char author[512];
-    char title[512];
-    char copyright[512];
-    char comment[512];
    pid_t pid;  /* Of ffmpeg process */
    time_t pid_start;  /* Of ffmpeg process */
    char **child_argv;
@@ -291,8 +287,7 @@ static void rtsp_cmd_describe(HTTPContext *c, const char *url);
 static void rtsp_cmd_options(HTTPContext *c, const char *url);
 static void rtsp_cmd_setup(HTTPContext *c, const char *url, RTSPMessageHeader *h);
 static void rtsp_cmd_play(HTTPContext *c, const char *url, RTSPMessageHeader *h);
-static void rtsp_cmd_pause(HTTPContext *c, const char *url, RTSPMessageHeader *h);
-static void rtsp_cmd_teardown(HTTPContext *c, const char *url, RTSPMessageHeader *h);
+static void rtsp_cmd_interrupt(HTTPContext *c, const char *url, RTSPMessageHeader *h, int pause_only);

 /* SDP handling */
 static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer,
@@ -328,6 +323,14 @@ 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];
@@ -502,8 +505,9 @@ static void start_children(FFStream *feed)
                char *slash;
                int i;

+                /* replace "ffserver" with "ffmpeg" in the path of current
+                 * program. Ignore user provided path */
                av_strlcpy(pathname, my_program_name, sizeof(pathname));
-
                slash = strrchr(pathname, '/');
                if (!slash)
                    slash = pathname;
@@ -623,17 +627,16 @@ static void start_multicast(void)
                }
            }

-            /* change state to send data */
            rtp_c->state = HTTPSTATE_SEND_DATA;
        }
    }
 }

-/* main loop of the http server */
+/* main loop of the HTTP server */
 static int http_server(void)
 {
    int server_fd = 0, rtsp_server_fd = 0;
-    int ret, delay, delay1;
+    int ret, delay;
    struct pollfd *poll_table, *poll_entry;
    HTTPContext *c, *c_next;

@@ -697,18 +700,19 @@ static int http_server(void)
            case HTTPSTATE_SEND_DATA:
            case HTTPSTATE_SEND_DATA_TRAILER:
                if (!c->is_packetized) {
-                    /* for TCP, we output as much as we can (may need to put a limit) */
+                    /* for TCP, we output as much as we can
+                     * (may need to put a limit) */
                    c->poll_entry = poll_entry;
                    poll_entry->fd = fd;
                    poll_entry->events = POLLOUT;
                    poll_entry++;
                } else {
                    /* when ffserver is doing the timing, we work by
-                       looking at which packet need to be sent every
+                       looking at which packet needs to be sent every
                       10 ms */
-                    delay1 = 10; /* one tick wait XXX: 10 ms assumed */
-                    if (delay1 < delay)
-                        delay = delay1;
+                    /* one tick wait XXX: 10 ms assumed */
+                    if (delay > 10)
+                        delay = 10;
                }
                break;
            case HTTPSTATE_WAIT_REQUEST:
@@ -748,8 +752,8 @@ static int http_server(void)
        for(c = first_http_ctx; c != NULL; c = c_next) {
            c_next = c->next;
            if (handle_connection(c) < 0) {
-                /* close and free the connection */
                log_connection(c);
+                /* close and free the connection */
                close_connection(c);
            }
        }
@@ -998,9 +1002,7 @@ static int handle_connection(HTTPContext *c)
        if (len < 0) {
            if (ff_neterrno() != AVERROR(EAGAIN) &&
                ff_neterrno() != AVERROR(EINTR)) {
-                /* error : close connection */
-                av_freep(&c->pb_buffer);
-                return -1;
+                goto close_connection;
            }
        } else {
            c->buffer_ptr += len;
@@ -1012,7 +1014,8 @@ static int handle_connection(HTTPContext *c)
                /* if error, exit */
                if (c->http_error)
                    return -1;
-                /* all the buffer was sent : synchronize to the incoming stream */
+                /* all the buffer was sent : synchronize to the incoming
+                 * stream */
                c->state = HTTPSTATE_SEND_DATA_HEADER;
                c->buffer_ptr = c->buffer_end = c->buffer;
            }
@@ -1023,7 +1026,7 @@ static int handle_connection(HTTPContext *c)
    case HTTPSTATE_SEND_DATA_HEADER:
    case HTTPSTATE_SEND_DATA_TRAILER:
        /* for packetized output, we consider we can always write (the
-           input streams sets the speed). It may be better to verify
+           input streams set the speed). It may be better to verify
           that we do not rely too much on the kernel queues */
        if (!c->is_packetized) {
            if (c->poll_entry->revents & (POLLERR | POLLHUP))
@@ -1057,10 +1060,8 @@ static int handle_connection(HTTPContext *c)
        break;

    case RTSPSTATE_SEND_REPLY:
-        if (c->poll_entry->revents & (POLLERR | POLLHUP)) {
-            av_freep(&c->pb_buffer);
-            return -1;
-        }
+        if (c->poll_entry->revents & (POLLERR | POLLHUP))
+            goto close_connection;
        /* no need to write if no events */
        if (!(c->poll_entry->revents & POLLOUT))
            return 0;
@@ -1068,9 +1069,7 @@ static int handle_connection(HTTPContext *c)
        if (len < 0) {
            if (ff_neterrno() != AVERROR(EAGAIN) &&
                ff_neterrno() != AVERROR(EINTR)) {
-                /* error : close connection */
-                av_freep(&c->pb_buffer);
-                return -1;
+                goto close_connection;
            }
        } else {
            c->buffer_ptr += len;
@@ -1115,6 +1114,10 @@ static int handle_connection(HTTPContext *c)
        return -1;
    }
    return 0;
+
+close_connection:
+    av_freep(&c->pb_buffer);
+    return -1;
 }

 static int extract_rates(char *rates, int ratelen, const char *request)
@@ -1453,7 +1456,7 @@ static int validate_acl(FFStream *stream, HTTPContext *c)
 }

 /* compute the real filename of a file by matching it without its
-   extensions to all the stream filenames */
+   extensions to all the stream's filenames */
 static void compute_real_filename(char *filename, int max_size)
 {
    char file1[1024];
@@ -1487,7 +1490,7 @@ enum RedirType {
    REDIR_SDP,
 };

-/* parse http request and prepare header */
+/* parse HTTP request and prepare header */
 static int http_parse_request(HTTPContext *c)
 {
    const char *p;
@@ -1858,7 +1861,7 @@ static int http_parse_request(HTTPContext *c)
        goto send_error;
    }

-    /* prepare http header */
+    /* prepare HTTP header */
    c->buffer[0] = 0;
    av_strlcatf(c->buffer, c->buffer_size, "HTTP/1.0 200 OK\r\n");
    mime_type = c->stream->fmt->mime_type;
@@ -1887,6 +1890,7 @@ 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"
@@ -1936,7 +1940,7 @@ static void compute_status(HTTPContext *c)
    }

    avio_printf(pb, "HTTP/1.0 200 OK\r\n");
-    avio_printf(pb, "Content-type: %s\r\n", "text/html");
+    avio_printf(pb, "Content-type: text/html\r\n");
    avio_printf(pb, "Pragma: no-cache\r\n");
    avio_printf(pb, "\r\n");

@@ -2175,8 +2179,10 @@ static int open_input_stream(HTTPContext *c, const char *info)
        buf_size = FFM_PACKET_SIZE;
        /* compute position (absolute time) */
        if (av_find_info_tag(buf, sizeof(buf), "date", info)) {
-            if ((ret = av_parse_time(&stream_pos, buf, 0)) < 0)
+            if ((ret = av_parse_time(&stream_pos, buf, 0)) < 0) {
+                http_log("Invalid date specification '%s' for stream\n", buf);
                return ret;
+            }
        } else if (av_find_info_tag(buf, sizeof(buf), "buffer", info)) {
            int prebuffer = strtol(buf, 0, 10);
            stream_pos = av_gettime() - prebuffer * (int64_t)1000000;
@@ -2187,18 +2193,22 @@ static int open_input_stream(HTTPContext *c, const char *info)
        buf_size = 0;
        /* compute position (relative time) */
        if (av_find_info_tag(buf, sizeof(buf), "date", info)) {
-            if ((ret = av_parse_time(&stream_pos, buf, 1)) < 0)
+            if ((ret = av_parse_time(&stream_pos, buf, 1)) < 0) {
+                http_log("Invalid date specification '%s' for stream\n", buf);
                return ret;
+            }
        } else
            stream_pos = 0;
    }
-    if (input_filename[0] == '\0')
-        return -1;
+    if (!input_filename[0]) {
+        http_log("No filename was specified for stream\n");
+        return AVERROR(EINVAL);
+    }

    /* open stream */
    if ((ret = avformat_open_input(&s, input_filename, c->stream->ifmt, &c->stream->in_opts)) < 0) {
-        http_log("could not open %s: %d\n", input_filename, ret);
-        return -1;
+        http_log("Could not open input '%s': %s\n", input_filename, av_err2str(ret));
+        return ret;
    }

    /* set buffer size */
@@ -2206,14 +2216,15 @@ static int open_input_stream(HTTPContext *c, const char *info)

    s->flags |= AVFMT_FLAG_GENPTS;
    c->fmt_in = s;
-    if (strcmp(s->iformat->name, "ffm") && avformat_find_stream_info(c->fmt_in, NULL) < 0) {
-        http_log("Could not find stream info '%s'\n", input_filename);
+    if (strcmp(s->iformat->name, "ffm") &&
+        (ret = avformat_find_stream_info(c->fmt_in, NULL)) < 0) {
+        http_log("Could not find stream info for input '%s'\n", input_filename);
        avformat_close_input(&s);
-        return -1;
+        return ret;
    }

-    /* choose stream as clock source (we favorize video stream if
-       present) for packet sending */
+    /* choose stream as clock source (we favor the video stream if
+     * present) for packet sending */
    c->pts_stream_index = 0;
    for(i=0;i<c->stream->nb_streams;i++) {
        if (c->pts_stream_index == 0 &&
@@ -2262,12 +2273,10 @@ static int http_prepare_data(HTTPContext *c)
    av_freep(&c->pb_buffer);
    switch(c->state) {
    case HTTPSTATE_SEND_DATA_HEADER:
-        memset(&c->fmt_ctx, 0, sizeof(c->fmt_ctx));
-        av_dict_set(&c->fmt_ctx.metadata, "author"   , c->stream->author   , 0);
-        av_dict_set(&c->fmt_ctx.metadata, "comment"  , c->stream->comment  , 0);
-        av_dict_set(&c->fmt_ctx.metadata, "copyright", c->stream->copyright, 0);
-        av_dict_set(&c->fmt_ctx.metadata, "title"    , c->stream->title    , 0);
-
+        ctx = avformat_alloc_context();
+        c->fmt_ctx = *ctx;
+        av_freep(&ctx);
+        av_dict_copy(&(c->fmt_ctx.metadata), c->stream->metadata, 0);
        c->fmt_ctx.streams = av_mallocz(sizeof(AVStream *) * c->stream->nb_streams);

        for(i=0;i<c->stream->nb_streams;i++) {
@@ -2282,8 +2291,8 @@ static int http_prepare_data(HTTPContext *c)

            *(c->fmt_ctx.streams[i]) = *src;
            c->fmt_ctx.streams[i]->priv_data = 0;
-            c->fmt_ctx.streams[i]->codec->frame_number = 0; /* XXX: should be done in
-                                           AVStream, not in codec */
+            /* XXX: should be done in AVStream, not in codec */
+            c->fmt_ctx.streams[i]->codec->frame_number = 0;
        }
        /* set output format parameters */
        c->fmt_ctx.oformat = c->stream->fmt;
@@ -2301,13 +2310,14 @@ static int http_prepare_data(HTTPContext *c)
        /*
         * HACK to avoid mpeg ps muxer to spit many underflow errors
         * Default value from FFmpeg
-         * Try to set it use configuration option
+         * Try to set it using configuration option
         */
        c->fmt_ctx.max_delay = (int)(0.7*AV_TIME_BASE);

-        if (avformat_write_header(&c->fmt_ctx, NULL) < 0) {
-            http_log("Error writing output header\n");
-            return -1;
+        if ((ret = avformat_write_header(&c->fmt_ctx, NULL)) < 0) {
+            http_log("Error writing output header for stream '%s': %s\n",
+                     c->stream->filename, av_err2str(ret));
+            return ret;
        }
        av_dict_free(&c->fmt_ctx.metadata);

@@ -2351,7 +2361,7 @@ static int http_prepare_data(HTTPContext *c)
                        goto redo;
                    } else {
                    no_loop:
-                        /* must send trailer now because eof or error */
+                        /* must send trailer now because EOF or error */
                        c->state = HTTPSTATE_SEND_DATA_TRAILER;
                    }
                }
@@ -2393,8 +2403,8 @@ static int http_prepare_data(HTTPContext *c)
                send_it:
                    ist = c->fmt_in->streams[source_index];
                    /* specific handling for RTP: we use several
-                       output stream (one for each RTP
-                       connection). XXX: need more abstract handling */
+                     * output streams (one for each RTP connection).
+                     * XXX: need more abstract handling */
                    if (c->is_packetized) {
                        /* compute send time and duration */
                        c->cur_pts = av_rescale_q(pkt.dts, ist->time_base, AV_TIME_BASE_Q);
@@ -2438,8 +2448,9 @@ static int http_prepare_data(HTTPContext *c)
                    if (pkt.pts != AV_NOPTS_VALUE)
                        pkt.pts = av_rescale_q(pkt.pts, ist->time_base, ost->time_base);
                    pkt.duration = av_rescale_q(pkt.duration, ist->time_base, ost->time_base);
-                    if (av_write_frame(ctx, &pkt) < 0) {
-                        http_log("Error writing frame to output\n");
+                    if ((ret = av_write_frame(ctx, &pkt)) < 0) {
+                        http_log("Error writing frame to output for stream '%s': %s\n",
+                                 c->stream->filename, av_err2str(ret));
                        c->state = HTTPSTATE_SEND_DATA_TRAILER;
                    }

@@ -2483,7 +2494,7 @@ static int http_prepare_data(HTTPContext *c)

 /* should convert the format at the same time */
 /* send data starting at c->buffer_ptr to the output connection
-   (either UDP or TCP connection) */
+ * (either UDP or TCP) */
 static int http_send_data(HTTPContext *c)
 {
    int len, ret;
@@ -2606,19 +2617,26 @@ static int http_send_data(HTTPContext *c)
 static int http_start_receive_data(HTTPContext *c)
 {
    int fd;
+    int ret;

-    if (c->stream->feed_opened)
-        return -1;
+    if (c->stream->feed_opened) {
+        http_log("Stream feed '%s' was not opened\n", c->stream->feed_filename);
+        return AVERROR(EINVAL);
+    }

    /* Don't permit writing to this one */
-    if (c->stream->readonly)
-        return -1;
+    if (c->stream->readonly) {
+        http_log("Cannot write to read-only file '%s'\n", c->stream->feed_filename);
+        return AVERROR(EINVAL);
+    }

    /* open feed */
    fd = open(c->stream->feed_filename, O_RDWR);
    if (fd < 0) {
-        http_log("Error opening feeder file: %s\n", strerror(errno));
-        return -1;
+        ret = AVERROR(errno);
+        http_log("Could not open feed file '%s': %s\n",
+                 c->stream->feed_filename, strerror(errno));
+        return ret;
    }
    c->feed_fd = fd;

@@ -2627,13 +2645,19 @@ static int http_start_receive_data(HTTPContext *c)
        ffm_write_write_index(c->feed_fd, FFM_PACKET_SIZE);
        http_log("Truncating feed file '%s'\n", c->stream->feed_filename);
        if (ftruncate(c->feed_fd, FFM_PACKET_SIZE) < 0) {
-            http_log("Error truncating feed file: %s\n", strerror(errno));
-            return -1;
+            ret = AVERROR(errno);
+            http_log("Error truncating feed file '%s': %s\n",
+                     c->stream->feed_filename, strerror(errno));
+            return ret;
        }
    } else {
-        if ((c->stream->feed_write_index = ffm_read_write_index(fd)) < 0) {
-            http_log("Error reading write index from feed file: %s\n", strerror(errno));
-            return -1;
+        ret = ffm_read_write_index(fd);
+        if (ret < 0) {
+            http_log("Error reading write index from feed file '%s': %s\n",
+                     c->stream->feed_filename, strerror(errno));
+            return ret;
+        } else {
+            c->stream->feed_write_index = ret;
        }
    }

@@ -2937,9 +2961,9 @@ static int rtsp_parse_request(HTTPContext *c)
    else if (!strcmp(cmd, "PLAY"))
        rtsp_cmd_play(c, url, header);
    else if (!strcmp(cmd, "PAUSE"))
-        rtsp_cmd_pause(c, url, header);
+        rtsp_cmd_interrupt(c, url, header, 1);
    else if (!strcmp(cmd, "TEARDOWN"))
-        rtsp_cmd_teardown(c, url, header);
+        rtsp_cmd_interrupt(c, url, header, 0);
    else
        rtsp_reply_error(c, RTSP_STATUS_METHOD);

@@ -2962,15 +2986,18 @@ static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer,
    AVFormatContext *avc;
    AVStream *avs = NULL;
    AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);
+    AVDictionaryEntry *entry = av_dict_get(stream->metadata, "title", NULL, 0);
    int i;

+    *pbuffer = NULL;
+
    avc =  avformat_alloc_context();
    if (avc == NULL || !rtp_format) {
        return -1;
    }
    avc->oformat = rtp_format;
    av_dict_set(&avc->metadata, "title",
-               stream->title[0] ? stream->title : "No Title", 0);
+                entry ? entry->value : "No Title", 0);
    avc->nb_streams = stream->nb_streams;
    if (stream->is_multicast) {
        snprintf(avc->filename, 1024, "rtp://%s:%d?multicast=1?ttl=%d",
@@ -3000,7 +3027,7 @@ static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer,
    av_free(avc);
    av_free(avs);

-    return strlen(*pbuffer);
+    return *pbuffer ? strlen(*pbuffer) : AVERROR(ENOMEM);
 }

 static void rtsp_cmd_options(HTTPContext *c, const char *url)
@@ -3292,7 +3319,7 @@ static void rtsp_cmd_play(HTTPContext *c, const char *url, RTSPMessageHeader *h)
    avio_printf(c->pb, "\r\n");
 }

-static void rtsp_cmd_pause(HTTPContext *c, const char *url, RTSPMessageHeader *h)
+static void rtsp_cmd_interrupt(HTTPContext *c, const char *url, RTSPMessageHeader *h, int pause_only)
 {
    HTTPContext *rtp_c;

@@ -3302,29 +3329,14 @@ static void rtsp_cmd_pause(HTTPContext *c, const char *url, RTSPMessageHeader *h
        return;
    }

-    if (rtp_c->state != HTTPSTATE_SEND_DATA &&
-        rtp_c->state != HTTPSTATE_WAIT_FEED) {
-        rtsp_reply_error(c, RTSP_STATUS_STATE);
-        return;
-    }
-
-    rtp_c->state = HTTPSTATE_READY;
-    rtp_c->first_pts = AV_NOPTS_VALUE;
-    /* now everything is OK, so we can send the connection parameters */
-    rtsp_reply_header(c, RTSP_STATUS_OK);
-    /* session ID */
-    avio_printf(c->pb, "Session: %s\r\n", rtp_c->session_id);
-    avio_printf(c->pb, "\r\n");
-}
-
-static void rtsp_cmd_teardown(HTTPContext *c, const char *url, RTSPMessageHeader *h)
-{
-    HTTPContext *rtp_c;
-
-    rtp_c = find_rtp_session_with_url(url, h->session_id);
-    if (!rtp_c) {
-        rtsp_reply_error(c, RTSP_STATUS_SESSION);
-        return;
+    if (pause_only) {
+        if (rtp_c->state != HTTPSTATE_SEND_DATA &&
+            rtp_c->state != HTTPSTATE_WAIT_FEED) {
+            rtsp_reply_error(c, RTSP_STATUS_STATE);
+            return;
+        }
+        rtp_c->state = HTTPSTATE_READY;
+        rtp_c->first_pts = AV_NOPTS_VALUE;
    }

    /* now everything is OK, so we can send the connection parameters */
@@ -3333,11 +3345,10 @@ static void rtsp_cmd_teardown(HTTPContext *c, const char *url, RTSPMessageHeader
    avio_printf(c->pb, "Session: %s\r\n", rtp_c->session_id);
    avio_printf(c->pb, "\r\n");

-    /* abort the session */
-    close_connection(rtp_c);
+    if (!pause_only)
+        close_connection(rtp_c);
 }

-
 /********************************************************************/
 /* RTP handling */

@@ -3482,7 +3493,8 @@ static int rtp_new_av_stream(HTTPContext *c,
             ipaddr, ntohs(dest_addr->sin_port),
             c->stream->filename, stream_index, c->protocol);

-    /* normally, no packets should be output here, but the packet size may be checked */
+    /* normally, no packets should be output here, but the packet size may
+     * be checked */
    if (ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size) < 0) {
        /* XXX: close stream */
        goto fail;
@@ -3524,7 +3536,7 @@ static AVStream *add_av_stream1(FFStream *stream, AVCodecContext *codec, int cop
        }
    } else {
        /* live streams must use the actual feed's codec since it may be
-         * updated later to carry extradata needed by the streams.
+         * updated later to carry extradata needed by them.
         */
        fst->codec = codec;
    }
@@ -3661,9 +3673,14 @@ static void build_file_streams(void)
                av_dict_set(&stream->in_opts, "mpeg2ts_compute_pcr", "1", 0);
            }

-            http_log("Opening file '%s'\n", stream->feed_filename);
+            if (!stream->feed_filename[0]) {
+                http_log("Unspecified feed file for stream '%s'\n", stream->filename);
+                goto fail;
+            }
+
+            http_log("Opening feed file '%s' for stream '%s'\n", stream->feed_filename, stream->filename);
            if ((ret = avformat_open_input(&infile, stream->feed_filename, stream->ifmt, &stream->in_opts)) < 0) {
-                http_log("Could not open '%s': %d\n", stream->feed_filename, ret);
+                http_log("Could not open '%s': %s\n", stream->feed_filename, av_err2str(ret));
                /* remove stream (no need to spend more time on it) */
            fail:
                remove_stream(stream);
@@ -3788,7 +3805,7 @@ static void build_feed_streams(void)
            }
        }
        if (avio_check(feed->feed_filename, AVIO_FLAG_WRITE) <= 0) {
-            AVFormatContext s1 = {0}, *s = &s1;
+            AVFormatContext *s = avformat_alloc_context();

            if (feed->readonly) {
                http_log("Unable to create feed file '%s' as it is marked readonly\n",
@@ -3812,6 +3829,9 @@ static void build_feed_streams(void)
            /* XXX: need better api */
            av_freep(&s->priv_data);
            avio_close(s->pb);
+            s->streams = NULL;
+            s->nb_streams = 0;
+            avformat_free_context(s);
        }
        /* get feed size and write index */
        fd = open(feed->feed_filename, O_RDONLY);
@@ -3906,7 +3926,7 @@ static void add_codec(FFStream *stream, AVCodecContext *av)
        av->rc_buffer_aggressivity = 1.0;

        if (!av->rc_eq)
-            av->rc_eq = "tex^qComp";
+            av->rc_eq = av_strdup("tex^qComp");
        if (!av->i_quant_factor)
            av->i_quant_factor = -0.8;
        if (!av->b_quant_factor)
@@ -3934,53 +3954,15 @@ static void add_codec(FFStream *stream, AVCodecContext *av)
    memcpy(st->codec, av, sizeof(AVCodecContext));
 }

-static enum AVCodecID opt_audio_codec(const char *arg)
+static enum AVCodecID opt_codec(const char *name, enum AVMediaType type)
 {
-    AVCodec *p= avcodec_find_encoder_by_name(arg);
+    AVCodec *codec = avcodec_find_encoder_by_name(name);

-    if (p == NULL || p->type != AVMEDIA_TYPE_AUDIO)
+    if (!codec || codec->type != type)
        return AV_CODEC_ID_NONE;
-
-    return p->id;
+    return codec->id;
 }

-static enum AVCodecID opt_video_codec(const char *arg)
-{
-    AVCodec *p= avcodec_find_encoder_by_name(arg);
-
-    if (p == NULL || p->type != AVMEDIA_TYPE_VIDEO)
-        return AV_CODEC_ID_NONE;
-
-    return p->id;
-}
-
-/* simplistic plugin support */
-
-#if HAVE_DLOPEN
-static void load_module(const char *filename)
-{
-    void *dll;
-    void (*init_func)(void);
-    dll = dlopen(filename, RTLD_NOW);
-    if (!dll) {
-        fprintf(stderr, "Could not load module '%s' - %s\n",
-                filename, dlerror());
-        return;
-    }
-
-    init_func = dlsym(dll, "ffserver_module_init");
-    if (!init_func) {
-        fprintf(stderr,
-                "%s: init function 'ffserver_module_init()' not found\n",
-                filename);
-        dlclose(dll);
-        return;
-    }
-
-    init_func();
-}
-#endif
-
 static int ffserver_opt_default(const char *opt, const char *arg,
                       AVCodecContext *avctx, int type)
 {
@@ -4017,9 +3999,9 @@ static int ffserver_opt_preset(const char *arg,
            break;
        }
        if(!strcmp(tmp, "acodec")){
-            *audio_id = opt_audio_codec(tmp2);
+            *audio_id = opt_codec(tmp2, AVMEDIA_TYPE_AUDIO);
        }else if(!strcmp(tmp, "vcodec")){
-            *video_id = opt_video_codec(tmp2);
+            *video_id = opt_codec(tmp2, AVMEDIA_TYPE_VIDEO);
        }else if(!strcmp(tmp, "scodec")){
            /* opt_subtitle_codec(tmp2); */
        }else if(ffserver_opt_default(tmp, tmp2, avctx, type) < 0){
@@ -4034,8 +4016,7 @@ static int ffserver_opt_preset(const char *arg,
    return ret;
 }

-static AVOutputFormat *ffserver_guess_format(const char *short_name, const char *filename,
-                                             const char *mime_type)
+static AVOutputFormat *ffserver_guess_format(const char *short_name, const char *filename, const char *mime_type)
 {
    AVOutputFormat *fmt = av_guess_format(short_name, filename, mime_type);

@@ -4053,12 +4034,12 @@ static AVOutputFormat *ffserver_guess_format(const char *short_name, const char
    return fmt;
 }

-static void report_config_error(const char *filename, int line_num, int *errors, const char *fmt, ...)
+static void report_config_error(const char *filename, int line_num, int log_level, int *errors, const char *fmt, ...)
 {
    va_list vl;
    va_start(vl, fmt);
-    fprintf(stderr, "%s:%d: ", filename, line_num);
-    vfprintf(stderr, fmt, vl);
+    av_log(NULL, log_level, "%s:%d: ", filename, line_num);
+    av_vlog(NULL, log_level, fmt, vl);
    va_end(vl);

    (*errors)++;
@@ -4069,21 +4050,23 @@ static int parse_ffconfig(const char *filename)
    FILE *f;
    char line[1024];
    char cmd[64];
-    char arg[1024];
+    char arg[1024], arg2[1024];
    const char *p;
-    int val, errors, line_num;
+    int val, errors, warnings, line_num;
    FFStream **last_stream, *stream, *redirect;
    FFStream **last_feed, *feed, *s;
    AVCodecContext audio_enc, video_enc;
    enum AVCodecID audio_id, video_id;
+    int ret = 0;

    f = fopen(filename, "r");
    if (!f) {
-        perror(filename);
-        return -1;
+        ret = AVERROR(errno);
+        av_log(NULL, AV_LOG_ERROR, "Could not open the configuration file '%s'\n", filename);
+        return ret;
    }

-    errors = 0;
+    errors = warnings = 0;
    line_num = 0;
    first_stream = NULL;
    last_stream = &first_stream;
@@ -4094,8 +4077,9 @@ static int parse_ffconfig(const char *filename)
    redirect = NULL;
    audio_id = AV_CODEC_ID_NONE;
    video_id = AV_CODEC_ID_NONE;
+#define ERROR(...)   report_config_error(filename, line_num, AV_LOG_ERROR,   &errors,   __VA_ARGS__)
+#define WARNING(...) report_config_error(filename, line_num, AV_LOG_WARNING, &warnings, __VA_ARGS__)

-#define ERROR(...) report_config_error(filename, line_num, &errors, __VA_ARGS__)
    for(;;) {
        if (fgets(line, sizeof(line), f) == NULL)
            break;
@@ -4121,7 +4105,7 @@ static int parse_ffconfig(const char *filename)
                ERROR("%s:%d: Invalid host/IP address: %s\n", arg);
            }
        } else if (!av_strcasecmp(cmd, "NoDaemon")) {
-            // do nothing here, its the default now
+            WARNING("NoDaemon option has no effect, you should remove it\n");
        } else if (!av_strcasecmp(cmd, "RTSPPort")) {
            get_arg(arg, sizeof(arg), &p);
            val = atoi(arg);
@@ -4168,6 +4152,10 @@ static int parse_ffconfig(const char *filename)
                ERROR("Already in a tag\n");
            } else {
                feed = av_mallocz(sizeof(FFStream));
+                if (!feed) {
+                    ret = AVERROR(ENOMEM);
+                    goto end;
+                }
                get_arg(feed->filename, sizeof(feed->filename), &p);
                q = strrchr(feed->filename, '>');
                if (*q)
@@ -4180,7 +4168,7 @@ static int parse_ffconfig(const char *filename)
                }

                feed->fmt = av_guess_format("ffm", NULL, NULL);
-                /* defaut feed file */
+                /* default feed file */
                snprintf(feed->feed_filename, sizeof(feed->feed_filename),
                         "/tmp/%s.ffm", feed->filename);
                feed->feed_max_size = 5 * 1024 * 1024;
@@ -4199,36 +4187,50 @@ static int parse_ffconfig(const char *filename)
                int i;

                feed->child_argv = av_mallocz(64 * sizeof(char *));
-
+                if (!feed->child_argv) {
+                    ret = AVERROR(ENOMEM);
+                    goto end;
+                }
                for (i = 0; i < 62; i++) {
                    get_arg(arg, sizeof(arg), &p);
                    if (!arg[0])
                        break;

                    feed->child_argv[i] = av_strdup(arg);
+                    if (!feed->child_argv[i]) {
+                        ret = AVERROR(ENOMEM);
+                        goto end;
+                    }
                }

-                feed->child_argv[i] = av_asprintf("http://%s:%d/%s",
-                        (my_http_addr.sin_addr.s_addr == INADDR_ANY) ? "127.0.0.1" :
-                    inet_ntoa(my_http_addr.sin_addr),
-                    ntohs(my_http_addr.sin_port), feed->filename);
+                feed->child_argv[i] =
+                    av_asprintf("http://%s:%d/%s",
+                                (my_http_addr.sin_addr.s_addr == INADDR_ANY) ? "127.0.0.1" :
+                                inet_ntoa(my_http_addr.sin_addr), ntohs(my_http_addr.sin_port),
+                                feed->filename);
+                if (!feed->child_argv[i]) {
+                    ret = AVERROR(ENOMEM);
+                    goto end;
+                }
            }
-        } else if (!av_strcasecmp(cmd, "ReadOnlyFile")) {
+        } else if (!av_strcasecmp(cmd, "File") || !av_strcasecmp(cmd, "ReadOnlyFile")) {
            if (feed) {
                get_arg(feed->feed_filename, sizeof(feed->feed_filename), &p);
-                feed->readonly = 1;
+                feed->readonly = !av_strcasecmp(cmd, "ReadOnlyFile");
            } else if (stream) {
                get_arg(stream->feed_filename, sizeof(stream->feed_filename), &p);
            }
-        } else if (!av_strcasecmp(cmd, "File")) {
-            if (feed) {
-                get_arg(feed->feed_filename, sizeof(feed->feed_filename), &p);
-            } else if (stream)
-                get_arg(stream->feed_filename, sizeof(stream->feed_filename), &p);
        } else if (!av_strcasecmp(cmd, "Truncate")) {
            if (feed) {
                get_arg(arg, sizeof(arg), &p);
-                feed->truncate = strtod(arg, NULL);
+                /* assume Truncate is true in case no argument is specified */
+                if (!arg[0]) {
+                    feed->truncate = 1;
+                } else {
+                    WARNING("Truncate N syntax in configuration file is deprecated, "
+                            "use Truncate alone with no arguments\n");
+                    feed->truncate = strtod(arg, NULL);
+                }
            }
        } else if (!av_strcasecmp(cmd, "FileMaxSize")) {
            if (feed) {
@@ -4268,6 +4270,10 @@ static int parse_ffconfig(const char *filename)
            } else {
                FFStream *s;
                stream = av_mallocz(sizeof(FFStream));
+                if (!stream) {
+                    ret = AVERROR(ENOMEM);
+                    goto end;
+                }
                get_arg(stream->filename, sizeof(stream->filename), &p);
                q = strrchr(stream->filename, '>');
                if (q)
@@ -4305,7 +4311,7 @@ static int parse_ffconfig(const char *filename)
                    sfeed = sfeed->next_feed;
                }
                if (!sfeed)
-                    ERROR("feed '%s' not defined\n", arg);
+                    ERROR("Feed with name '%s' for stream '%s' is not defined\n", arg, stream->filename);
                else
                    stream->feed = sfeed;
            }
@@ -4344,18 +4350,36 @@ static int parse_ffconfig(const char *filename)
            } else {
                ERROR("FaviconURL only permitted for status streams\n");
            }
-        } else if (!av_strcasecmp(cmd, "Author")) {
-            if (stream)
-                get_arg(stream->author, sizeof(stream->author), &p);
-        } else if (!av_strcasecmp(cmd, "Comment")) {
-            if (stream)
-                get_arg(stream->comment, sizeof(stream->comment), &p);
-        } else if (!av_strcasecmp(cmd, "Copyright")) {
-            if (stream)
-                get_arg(stream->copyright, sizeof(stream->copyright), &p);
-        } else if (!av_strcasecmp(cmd, "Title")) {
-            if (stream)
-                get_arg(stream->title, sizeof(stream->title), &p);
+        } else if (!av_strcasecmp(cmd, "Author")    ||
+                   !av_strcasecmp(cmd, "Comment")   ||
+                   !av_strcasecmp(cmd, "Copyright") ||
+                   !av_strcasecmp(cmd, "Title")) {
+            get_arg(arg, sizeof(arg), &p);
+
+            if (stream) {
+                char key[32];
+                int i, ret;
+
+                for (i = 0; i < strlen(cmd); i++)
+                    key[i] = av_tolower(cmd[i]);
+                key[i] = 0;
+                WARNING("'%s' option in configuration file is deprecated, "
+                        "use 'Metadata %s VALUE' instead\n", cmd, key);
+                if ((ret = av_dict_set(&stream->metadata, key, arg, 0)) < 0) {
+                    ERROR("Could not set metadata '%s' to value '%s': %s\n",
+                          key, arg, av_err2str(ret));
+                }
+            }
+        } else if (!av_strcasecmp(cmd, "Metadata")) {
+            get_arg(arg, sizeof(arg), &p);
+            get_arg(arg2, sizeof(arg2), &p);
+            if (stream) {
+                int ret;
+                if ((ret = av_dict_set(&stream->metadata, arg, arg2, 0)) < 0) {
+                    ERROR("Could not set metadata '%s' to value '%s': %s\n",
+                          arg, arg2, av_err2str(ret));
+                }
+            }
        } else if (!av_strcasecmp(cmd, "Preroll")) {
            get_arg(arg, sizeof(arg), &p);
            if (stream)
@@ -4365,13 +4389,13 @@ static int parse_ffconfig(const char *filename)
                stream->send_on_key = 1;
        } else if (!av_strcasecmp(cmd, "AudioCodec")) {
            get_arg(arg, sizeof(arg), &p);
-            audio_id = opt_audio_codec(arg);
+            audio_id = opt_codec(arg, AVMEDIA_TYPE_AUDIO);
            if (audio_id == AV_CODEC_ID_NONE) {
                ERROR("Unknown AudioCodec: %s\n", arg);
            }
        } else if (!av_strcasecmp(cmd, "VideoCodec")) {
            get_arg(arg, sizeof(arg), &p);
-            video_id = opt_video_codec(arg);
+            video_id = opt_codec(arg, AVMEDIA_TYPE_VIDEO);
            if (video_id == AV_CODEC_ID_NONE) {
                ERROR("Unknown VideoCodec: %s\n", arg);
            }
@@ -4391,11 +4415,6 @@ static int parse_ffconfig(const char *filename)
            get_arg(arg, sizeof(arg), &p);
            if (stream)
                audio_enc.sample_rate = atoi(arg);
-        } else if (!av_strcasecmp(cmd, "AudioQuality")) {
-            get_arg(arg, sizeof(arg), &p);
-            if (stream) {
-//                audio_enc.quality = atof(arg) * 1000;
-            }
        } else if (!av_strcasecmp(cmd, "VideoBitRateRange")) {
            if (stream) {
                int minrate, maxrate;
@@ -4437,10 +4456,14 @@ static int parse_ffconfig(const char *filename)
        } else if (!av_strcasecmp(cmd, "VideoSize")) {
            get_arg(arg, sizeof(arg), &p);
            if (stream) {
-                av_parse_video_size(&video_enc.width, &video_enc.height, arg);
-                if ((video_enc.width % 16) != 0 ||
-                    (video_enc.height % 16) != 0) {
-                    ERROR("Image size must be a multiple of 16\n");
+                ret = av_parse_video_size(&video_enc.width, &video_enc.height, arg);
+                if (ret < 0) {
+                    ERROR("Invalid video size '%s'\n", arg);
+                } else {
+                    if ((video_enc.width % 16) != 0 ||
+                        (video_enc.height % 16) != 0) {
+                        ERROR("Image size must be a multiple of 16\n");
+                    }
                }
            }
        } else if (!av_strcasecmp(cmd, "VideoFrameRate")) {
@@ -4454,6 +4477,14 @@ static int parse_ffconfig(const char *filename)
                    video_enc.time_base.den = frame_rate.num;
                }
            }
+        } else if (!av_strcasecmp(cmd, "PixelFormat")) {
+            get_arg(arg, sizeof(arg), &p);
+            if (stream) {
+                video_enc.pix_fmt = av_get_pix_fmt(arg);
+                if (video_enc.pix_fmt == AV_PIX_FMT_NONE) {
+                    ERROR("Unknown pixel format: %s\n", arg);
+                }
+            }
        } else if (!av_strcasecmp(cmd, "VideoGopSize")) {
            get_arg(arg, sizeof(arg), &p);
            if (stream)
@@ -4471,7 +4502,6 @@ static int parse_ffconfig(const char *filename)
            }
        } else if (!av_strcasecmp(cmd, "AVOptionVideo") ||
                   !av_strcasecmp(cmd, "AVOptionAudio")) {
-            char arg2[1024];
            AVCodecContext *avctx;
            int type;
            get_arg(arg, sizeof(arg), &p);
@@ -4484,7 +4514,7 @@ static int parse_ffconfig(const char *filename)
                type = AV_OPT_FLAG_AUDIO_PARAM;
            }
            if (ffserver_opt_default(arg, arg2, avctx, type|AV_OPT_FLAG_ENCODING_PARAM)) {
-                ERROR("AVOption error: %s %s\n", arg, arg2);
+                ERROR("Error setting %s option to %s %s\n", cmd, arg, arg2);
            }
        } else if (!av_strcasecmp(cmd, "AVPresetVideo") ||
                   !av_strcasecmp(cmd, "AVPresetAudio")) {
@@ -4615,6 +4645,10 @@ static int parse_ffconfig(const char *filename)
                ERROR("Already in a tag\n");
            } else {
                redirect = av_mallocz(sizeof(FFStream));
+                if (!redirect) {
+                    ret = AVERROR(ENOMEM);
+                    goto end;
+                }
                *last_stream = redirect;
                last_stream = &redirect->next;

@@ -4637,21 +4671,19 @@ static int parse_ffconfig(const char *filename)
                redirect = NULL;
            }
        } else if (!av_strcasecmp(cmd, "LoadModule")) {
-            get_arg(arg, sizeof(arg), &p);
-#if HAVE_DLOPEN
-            load_module(arg);
-#else
-            ERROR("Module support not compiled into this version: '%s'\n", arg);
-#endif
+            ERROR("Loadable modules no longer supported\n");
        } else {
            ERROR("Incorrect keyword: '%s'\n", cmd);
        }
    }
 #undef ERROR

+end:
    fclose(f);
+    if (ret < 0)
+        return ret;
    if (errors)
-        return -1;
+        return AVERROR(EINVAL);
    else
        return 0;
 }
@@ -4706,6 +4738,9 @@ static const OptionDef options[] = {
 int main(int argc, char **argv)
 {
    struct sigaction sigact = { { 0 } };
+    int ret = 0;
+
+    config_filename = av_strdup("/etc/ffserver.conf");

    parse_loglevel(argc, argv, options);
    av_register_all();
@@ -4717,9 +4752,6 @@ 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());
@@ -4728,8 +4760,9 @@ int main(int argc, char **argv)
    sigact.sa_flags = SA_NOCLDSTOP | SA_RESTART;
    sigaction(SIGCHLD, &sigact, 0);

-    if (parse_ffconfig(config_filename) < 0) {
-        fprintf(stderr, "Incorrect config file - exiting.\n");
+    if ((ret = parse_ffconfig(config_filename)) < 0) {
+        fprintf(stderr, "Error reading configuration file '%s': %s\n",
+                config_filename, av_err2str(ret));
        exit(1);
    }
    av_freep(&config_filename);
--- a/libavcodec/012v.c
+++ b/libavcodec/012v.c
@@ -49,6 +49,12 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
        av_log(avctx, AV_LOG_ERROR, "Width 1 not supported.\n");
        return AVERROR_INVALIDDATA;
    }
+
+    if (   avctx->codec_tag == MKTAG('0', '1', '2', 'v')
+        && avpkt->size % avctx->height == 0
+        && avpkt->size / avctx->height * 3 >= width * 8)
+        stride = avpkt->size / avctx->height;
+
    if (avpkt->size < avctx->height * stride) {
        av_log(avctx, AV_LOG_ERROR, "Packet too small: %d instead of %d\n",
               avpkt->size, avctx->height * stride);
@@ -144,10 +150,10 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,

 AVCodec ff_zero12v_decoder = {
    .name           = "012v",
+    .long_name      = NULL_IF_CONFIG_SMALL("Uncompressed 4:2:2 10-bit"),
    .type           = AVMEDIA_TYPE_VIDEO,
    .id             = AV_CODEC_ID_012V,
    .init           = zero12v_decode_init,
    .decode         = zero12v_decode_frame,
    .capabilities   = CODEC_CAP_DR1,
-    .long_name      = NULL_IF_CONFIG_SMALL("Uncompressed 4:2:2 10-bit"),
 };
--- a/Show More
+++ b/Show More
@@ -1 +1 @@
 .0
 .2.11