update for 2.8.15

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
avcodec/dvdsub_parser: Allocate input padding
2018-07-15 20:29:24 +02:00 · 2018-07-15 20:08:57 +02:00 · 2018-07-15 20:07:51 +02:00 · 2018-07-15 19:25:36 +02:00 · 2018-07-13 18:36:55 +02:00 · 2018-07-10 10:51:01 +02:00
2352 changed files with 191284 additions and 36051 deletions
@@ -28,6 +28,7 @@
 /ffserver
 /config.*
 /coverage.info
+/avversion.h
 /doc/*.1
 /doc/*.3
 /doc/*.html
@@ -36,6 +37,7 @@
 /doc/avoptions_codec.texi
 /doc/avoptions_format.texi
 /doc/doxy/html/
+/doc/examples/avio_dir_cmd
 /doc/examples/avio_reading
 /doc/examples/decoding_encoding
 /doc/examples/demuxing_decoding
@@ -61,6 +63,7 @@
 /libavutil/ffversion.h
 /tests/audiogen
 /tests/base64
+/tests/checkasm/checkasm
 /tests/data/
 /tests/pixfmts.mak
 /tests/rotozoom
@@ -16,6 +16,7 @@ Specifically, the GPL parts of FFmpeg are:
 - optional x86 optimizations in the files
  - `libavcodec/x86/flac_dsp_gpl.asm`
  - `libavcodec/x86/idct_mmx.c`
+  - `libavfilter/x86/vf_removegrain.asm`
 - libutvideo encoding/decoding wrappers in
  `libavcodec/libutvideo*.cpp`
 - the X11 grabber in `libavdevice/x11grab.c`
@@ -27,9 +28,11 @@ Specifically, the GPL parts of FFmpeg are:
    - `vf_blackframe.c`
    - `vf_boxblur.c`
    - `vf_colormatrix.c`
+    - `vf_cover_rect.c`
    - `vf_cropdetect.c`
    - `vf_delogo.c`
    - `vf_eq.c`
+    - `vf_find_rect.c`
    - `vf_fspp.c`
    - `vf_geq.c`
    - `vf_histeq.c`
@@ -14,7 +14,6 @@ patches and related discussions.
 Project Leader
 ==============

-Michael Niedermayer
  final design decisions


@@ -43,9 +42,9 @@ QuickTime faststart:
 Miscellaneous Areas
 ===================

-documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu
+documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu, Lou Logan
 build system (configure, makefiles)     Diego Biurrun, Mans Rullgard
-project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser
+project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Lou Logan
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
 release management                      Michael Niedermayer
@@ -57,9 +56,9 @@ Communication
 website                                 Deby Barbara Lepage
 fate.ffmpeg.org                         Timothy Gu
 Trac bug tracker                        Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan
-mailing lists                           Michael Niedermayer, Baptiste Coudurier, Lou Logan
+mailing lists                           Baptiste Coudurier, Lou Logan
 Google+                                 Paul B Mahol, Michael Niedermayer, Alexander Strasser
-Twitter                                 Lou Logan
+Twitter                                 Lou Logan, Reynaldo H. Verdejo Pinochet
 Launchpad                               Timothy Gu


@@ -138,6 +137,7 @@ Codecs:
  4xm.c                                 Michael Niedermayer
  8bps.c                                Roberto Togni
  8svx.c                                Jaikrishnan Menon
+  aacenc*, aaccoder.c                   Rostislav Pehlivanov
  aasc.c                                Kostya Shishkov
  ac3*                                  Justin Ruggles
  alacenc.c                             Jaikrishnan Menon
@@ -166,11 +166,12 @@ Codecs:
  dca.c                                 Kostya Shishkov, Benjamin Larsson
  dnxhd*                                Baptiste Coudurier
  dpcm.c                                Mike Melanson
-  dss_sp.c                              Oleksij Rempel, Michael Niedermayer
+  dss_sp.c                              Oleksij Rempel
  dv.c                                  Roman Shaposhnik
  dvbsubdec.c                           Anshul Maheshwari
  dxa.c                                 Kostya Shishkov
  eacmv*, eaidct*, eat*                 Peter Ross
+  evrc*                                 Paul B Mahol
  exif.c, exif.h                        Thilo Borgmann
  ffv1*                                 Michael Niedermayer
  ffwavesynth.c                         Nicolas George
@@ -200,6 +201,7 @@ Codecs:
  libcelt_dec.c                         Nicolas George
  libdirac*                             David Conrad
  libgsm.c                              Michel Bardiaux
+  libkvazaar.c                          Arttu Ylä-Outinen
  libopenjpeg.c                         Jaikrishnan Menon
  libopenjpegenc.c                      Michael Bradshaw
  libschroedinger*                      David Conrad
@@ -237,6 +239,7 @@ Codecs:
  qdm2.c, qdm2data.h                    Roberto Togni, Benjamin Larsson
  qdrw.c                                Kostya Shishkov
  qpeg.c                                Kostya Shishkov
+  qsv*                                  Ivan Uskov
  qtrle.c                               Mike Melanson
  ra144.c, ra144.h, ra288.c, ra288.h    Roberto Togni
  resample2.c                           Michael Niedermayer
@@ -298,11 +301,12 @@ Codecs:

 Hardware acceleration:
  crystalhd.c                           Philip Langdale
-  dxva2*                                Laurent Aimar
+  dxva2*                                Hendrik Leppkes, Laurent Aimar
  libstagefright.cpp                    Mohamed Naufal
  vaapi*                                Gwenole Beauchesne
  vda*                                  Sebastien Zwickert
-  vdpau*                                Carl Eugen Hoyos
+  vdpau*                                Philip Langdale, Carl Eugen Hoyos
+  videotoolbox*                         Sebastien Zwickert


 libavdevice
@@ -334,6 +338,7 @@ Generic parts:
  graphdump.c                           Nicolas George

 Filters:
+  f_drawgraph.c                         Paul B Mahol
  af_adelay.c                           Paul B Mahol
  af_aecho.c                            Paul B Mahol
  af_afade.c                            Paul B Mahol
@@ -344,14 +349,21 @@ Filters:
  af_astreamsync.c                      Nicolas George
  af_atempo.c                           Pavel Koshevoy
  af_biquads.c                          Paul B Mahol
+  af_chorus.c                           Paul B Mahol
  af_compand.c                          Paul B Mahol
  af_ladspa.c                           Paul B Mahol
  af_pan.c                              Nicolas George
+  af_sidechaincompress.c                Paul B Mahol
  af_silenceremove.c                    Paul B Mahol
+  avf_aphasemeter.c                     Paul B Mahol
  avf_avectorscope.c                    Paul B Mahol
  avf_showcqt.c                         Muhammad Faiz
  vf_blend.c                            Paul B Mahol
+  vf_colorchannelmixer.c                Paul B Mahol
  vf_colorbalance.c                     Paul B Mahol
+  vf_colorkey.c                         Timo Rothenpieler
+  vf_colorlevels.c                      Paul B Mahol
+  vf_deband.c                           Paul B Mahol
  vf_dejudder.c                         Nicholas Robbins
  vf_delogo.c                           Jean Delvare (CC <khali@linux-fr.org>)
  vf_drawbox.c/drawgrid                 Andrey Utkin
@@ -362,12 +374,16 @@ Filters:
  vf_il.c                               Paul B Mahol
  vf_lenscorrection.c                   Daniel Oberhoff
  vf_mergeplanes.c                      Paul B Mahol
+  vf_neighbor.c                         Paul B Mahol
  vf_psnr.c                             Paul B Mahol
+  vf_random.c                           Paul B Mahol
  vf_scale.c                            Michael Niedermayer
  vf_separatefields.c                   Paul B Mahol
+  vf_ssim.c                             Paul B Mahol
  vf_stereo3d.c                         Paul B Mahol
  vf_telecine.c                         Paul B Mahol
  vf_yadif.c                            Michael Niedermayer
+  vf_zoompan.c                          Paul B Mahol

 Sources:
  vsrc_mandelbrot.c                     Michael Niedermayer
@@ -384,6 +400,7 @@ Generic parts:

 Muxers/Demuxers:
  4xm.c                                 Mike Melanson
+  aadec.c                               Vesselin Bontchev (vesselin.bontchev at yandex dot com)
  adtsenc.c                             Robert Swain
  afc.c                                 Paul B Mahol
  aiffdec.c                             Baptiste Coudurier, Matthieu Bouron
@@ -402,7 +419,7 @@ Muxers/Demuxers:
  cdxl.c                                Paul B Mahol
  crc.c                                 Michael Niedermayer
  daud.c                                Reimar Doeffinger
-  dss.c                                 Oleksij Rempel, Michael Niedermayer
+  dss.c                                 Oleksij Rempel
  dtshddec.c                            Paul B Mahol
  dv.c                                  Roman Shaposhnik
  dxa.c                                 Kostya Shishkov
@@ -415,6 +432,7 @@ Muxers/Demuxers:
  gxf.c                                 Reimar Doeffinger
  gxfenc.c                              Baptiste Coudurier
  hls.c                                 Anssi Hannula
+  hls encryption (hlsenc.c)             Christian Suloway
  idcin.c                               Mike Melanson
  idroqdec.c                            Mike Melanson
  iff.c                                 Jaikrishnan Menon
@@ -437,7 +455,7 @@ Muxers/Demuxers:
  mgsts.c                               Paul B Mahol
  microdvd*                             Aurelien Jacobs
  mm.c                                  Peter Ross
-  mov.c                                 Michael Niedermayer, Baptiste Coudurier
+  mov.c                                 Baptiste Coudurier
  movenc.c                              Baptiste Coudurier, Matthieu Bouron
  mpc.c                                 Kostya Shishkov
  mpeg.c                                Michael Niedermayer
@@ -500,6 +518,7 @@ Muxers/Demuxers:
  wvenc.c                               Paul B Mahol

 Protocols:
+  async.c                               Zhang Rui
  bluray.c                              Petri Hintukainen
  ftp.c                                 Lukasz Marek
  http.c                                Ronald S. Bultje
@@ -535,20 +554,21 @@ Amiga / PowerPC                         Colin Ward
 Linux / PowerPC                         Luca Barbato
 Windows MinGW                           Alex Beregszaszi, Ramiro Polla
 Windows Cygwin                          Victor Paesa
-Windows MSVC                            Matthew Oliver
+Windows MSVC                            Matthew Oliver, Hendrik Leppkes
 Windows ICL                             Matthew Oliver
 ADI/Blackfin DSP                        Marc Hoffman
 Sparc                                   Roman Shaposhnik
-x86                                     Michael Niedermayer
+OS/2                                    KO Myung-Hun


 Releases
 ========

+2.8                                     Michael Niedermayer
+2.7                                     Michael Niedermayer
 2.6                                     Michael Niedermayer
 2.5                                     Michael Niedermayer
 2.4                                     Michael Niedermayer
-2.2                                     Michael Niedermayer

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

@@ -579,6 +599,7 @@ Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
 Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
 Panagiotis Issaris            6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
 Peter Ross                    A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
+Philip Langdale               5DC5 8D66 5FBA 3A43 18EC 045E F8D6 B194 6A75 682E
 Reimar Doeffinger             C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
 Reinhard Tartler              9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
 Reynaldo H. Verdejo Pinochet  6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
@@ -31,7 +31,10 @@ $(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog)-$(CONFIG_OPENCL) += cmdutils_o
 OBJS-ffmpeg                   += ffmpeg_opt.o ffmpeg_filter.o
 OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o
 OBJS-ffmpeg-$(HAVE_DXVA2_LIB) += ffmpeg_dxva2.o
-OBJS-ffmpeg-$(CONFIG_VDA)     += ffmpeg_vda.o
+ifndef CONFIG_VIDEOTOOLBOX
+OBJS-ffmpeg-$(CONFIG_VDA)     += ffmpeg_videotoolbox.o
+endif
+OBJS-ffmpeg-$(CONFIG_VIDEOTOOLBOX) += ffmpeg_videotoolbox.o
 OBJS-ffserver                 += ffserver_config.o

 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
@@ -60,6 +63,7 @@ include $(SRC_PATH)/common.mak

 FF_EXTRALIBS := $(FFEXTRALIBS)
 FF_DEP_LIBS  := $(DEP_LIBS)
+FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)

 all: $(AVPROGS)

@@ -80,8 +84,8 @@ SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS      \
               HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS            \
               ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS     \
               ALTIVEC-OBJS MMX-OBJS YASM-OBJS                           \
-               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS                \
-               OBJS SLIBOBJS HOSTOBJS TESTOBJS
+               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MSA-OBJS       \
+               MMI-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS

 define RESET
 $(1) :=
@@ -175,7 +179,7 @@ clean::

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

 config:
 	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
@@ -1 +1 @@
-2.5.git
+2.8.15
@@ -0,0 +1,15 @@
+
+              ┌────────────────────────────────────────┐
+              │ RELEASE NOTES for FFmpeg 2.8 "Feynman" │
+              └────────────────────────────────────────┘
+
+   The FFmpeg Project proudly presents FFmpeg 2.8 "Feynman", about 3
+   months after the release of FFmpeg 2.7.
+
+   A complete Changelog is available at the root of the project, and the
+   complete Git history on http://source.ffmpeg.org.
+
+   We hope you will like this release as much as we enjoyed working on it, and
+   as usual, if you have any questions about it, or any FFmpeg related topic,
+   feel free to join us on the #ffmpeg IRC channel (on irc.freenode.net) or ask
+   on the mailing-lists.
@@ -7,8 +7,11 @@ OBJS-$(HAVE_NEON)    += $(NEON-OBJS)    $(NEON-OBJS-yes)
 OBJS-$(HAVE_MIPSFPU)   += $(MIPSFPU-OBJS)    $(MIPSFPU-OBJS-yes)
 OBJS-$(HAVE_MIPSDSPR1) += $(MIPSDSPR1-OBJS)  $(MIPSDSPR1-OBJS-yes)
 OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS)  $(MIPSDSPR2-OBJS-yes)
+OBJS-$(HAVE_MSA)       += $(MSA-OBJS)        $(MSA-OBJS-yes)
+OBJS-$(HAVE_MMI)   += $(MMI-OBJS)   $(MMI-OBJS-yes)

 OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
+OBJS-$(HAVE_VSX)     += $(VSX-OBJS) $(VSX-OBJS-yes)

 OBJS-$(HAVE_MMX)     += $(MMX-OBJS)     $(MMX-OBJS-yes)
 OBJS-$(HAVE_YASM)    += $(YASM-OBJS)    $(YASM-OBJS-yes)
@@ -41,8 +41,10 @@
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
 #include "libavutil/bprint.h"
+#include "libavutil/display.h"
 #include "libavutil/mathematics.h"
 #include "libavutil/imgutils.h"
+#include "libavutil/libm.h"
 #include "libavutil/parseutils.h"
 #include "libavutil/pixdesc.h"
 #include "libavutil/eval.h"
@@ -58,10 +60,13 @@
 #include <sys/time.h>
 #include <sys/resource.h>
 #endif
+#if HAVE_SETDLLDIRECTORY
+#include <windows.h>
+#endif

 static int init_report(const char *env);

-struct SwsContext *sws_opts;
+AVDictionary *sws_dict;
 AVDictionary *swr_opts;
 AVDictionary *format_opts, *codec_opts, *resample_opts;

@@ -71,20 +76,13 @@ int hide_banner = 0;

 void init_opts(void)
 {
-
-    if(CONFIG_SWSCALE)
-        sws_opts = sws_getContext(16, 16, 0, 16, 16, 0, SWS_BICUBIC,
-                              NULL, NULL, NULL);
+    av_dict_set(&sws_dict, "flags", "bicubic", 0);
 }

 void uninit_opts(void)
 {
-#if CONFIG_SWSCALE
-    sws_freeContext(sws_opts);
-    sws_opts = NULL;
-#endif
-
    av_dict_free(&swr_opts);
+    av_dict_free(&sws_dict);
    av_dict_free(&format_opts);
    av_dict_free(&codec_opts);
    av_dict_free(&resample_opts);
@@ -111,6 +109,15 @@ static void log_callback_report(void *ptr, int level, const char *fmt, va_list v
    }
 }

+void init_dynload(void)
+{
+#if HAVE_SETDLLDIRECTORY
+    /* Calling SetDllDirectory with the empty string (but not NULL) removes the
+     * current working directory from the DLL search path as a security pre-caution. */
+    SetDllDirectory("");
+#endif
+}
+
 static void (*program_exit)(int ret);

 void register_exit(void (*cb)(int ret))
@@ -480,10 +487,22 @@ static void dump_argument(const char *a)
    fputc('"', report_file);
 }

+static void check_options(const OptionDef *po)
+{
+    while (po->name) {
+        if (po->flags & OPT_PERFILE)
+            av_assert0(po->flags & (OPT_INPUT | OPT_OUTPUT));
+        po++;
+    }
+}
+
 void parse_loglevel(int argc, char **argv, const OptionDef *options)
 {
    int idx = locate_option(argc, argv, options, "loglevel");
    const char *env;
+
+    check_options(options);
+
    if (!idx)
        idx = locate_option(argc, argv, options, "v");
    if (idx && argv[idx + 1])
@@ -515,7 +534,7 @@ static const AVOption *opt_find(void *obj, const char *name, const char *unit,
    return o;
 }

-#define FLAGS (o->type == AV_OPT_TYPE_FLAGS) ? AV_DICT_APPEND : 0
+#define FLAGS (o->type == AV_OPT_TYPE_FLAGS && (arg[0]=='-' || arg[0]=='+')) ? AV_DICT_APPEND : 0
 int opt_default(void *optctx, const char *opt, const char *arg)
 {
    const AVOption *o;
@@ -551,14 +570,24 @@ int opt_default(void *optctx, const char *opt, const char *arg)
    }
 #if CONFIG_SWSCALE
    sc = sws_get_class();
-    if (!consumed && opt_find(&sc, opt, NULL, 0,
-                         AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ)) {
-        // XXX we only support sws_flags, not arbitrary sws options
-        int ret = av_opt_set(sws_opts, opt, arg, 0);
+    if (!consumed && (o = opt_find(&sc, opt, NULL, 0,
+                         AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
+        struct SwsContext *sws = sws_alloc_context();
+        int ret = av_opt_set(sws, opt, arg, 0);
+        sws_freeContext(sws);
+        if (!strcmp(opt, "srcw") || !strcmp(opt, "srch") ||
+            !strcmp(opt, "dstw") || !strcmp(opt, "dsth") ||
+            !strcmp(opt, "src_format") || !strcmp(opt, "dst_format")) {
+            av_log(NULL, AV_LOG_ERROR, "Directly using swscale dimensions/format options is not supported, please use the -s or -pix_fmt options\n");
+            return AVERROR(EINVAL);
+        }
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Error setting option %s.\n", opt);
            return ret;
        }
+
+        av_dict_set(&sws_dict, opt, arg, FLAGS);
+
        consumed = 1;
    }
 #else
@@ -632,9 +661,7 @@ static void finish_group(OptionParseContext *octx, int group_idx,
    *g             = octx->cur_group;
    g->arg         = arg;
    g->group_def   = l->group_def;
-#if CONFIG_SWSCALE
-    g->sws_opts    = sws_opts;
-#endif
+    g->sws_dict    = sws_dict;
    g->swr_opts    = swr_opts;
    g->codec_opts  = codec_opts;
    g->format_opts = format_opts;
@@ -643,9 +670,7 @@ static void finish_group(OptionParseContext *octx, int group_idx,
    codec_opts  = NULL;
    format_opts = NULL;
    resample_opts = NULL;
-#if CONFIG_SWSCALE
-    sws_opts    = NULL;
-#endif
+    sws_dict    = NULL;
    swr_opts    = NULL;
    init_opts();

@@ -701,9 +726,8 @@ void uninit_parse_context(OptionParseContext *octx)
            av_dict_free(&l->groups[j].codec_opts);
            av_dict_free(&l->groups[j].format_opts);
            av_dict_free(&l->groups[j].resample_opts);
-#if CONFIG_SWSCALE
-            sws_freeContext(l->groups[j].sws_opts);
-#endif
+
+            av_dict_free(&l->groups[j].sws_dict);
            av_dict_free(&l->groups[j].swr_opts);
        }
        av_freep(&l->groups);
@@ -845,6 +869,7 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg)
        { "info"   , AV_LOG_INFO    },
        { "verbose", AV_LOG_VERBOSE },
        { "debug"  , AV_LOG_DEBUG   },
+        { "trace"  , AV_LOG_TRACE   },
    };
    char *tail;
    int level;
@@ -1309,12 +1334,12 @@ static void print_codec(const AVCodec *c)
    if (c->type == AVMEDIA_TYPE_VIDEO ||
        c->type == AVMEDIA_TYPE_AUDIO) {
        printf("    Threading capabilities: ");
-        switch (c->capabilities & (CODEC_CAP_FRAME_THREADS |
-                                   CODEC_CAP_SLICE_THREADS)) {
-        case CODEC_CAP_FRAME_THREADS |
-             CODEC_CAP_SLICE_THREADS: printf("frame and slice"); break;
-        case CODEC_CAP_FRAME_THREADS: printf("frame");           break;
-        case CODEC_CAP_SLICE_THREADS: printf("slice");           break;
+        switch (c->capabilities & (AV_CODEC_CAP_FRAME_THREADS |
+                                   AV_CODEC_CAP_SLICE_THREADS)) {
+        case AV_CODEC_CAP_FRAME_THREADS |
+             AV_CODEC_CAP_SLICE_THREADS: printf("frame and slice"); break;
+        case AV_CODEC_CAP_FRAME_THREADS: printf("frame");           break;
+        case AV_CODEC_CAP_SLICE_THREADS: printf("slice");           break;
        default:                      printf("no");              break;
        }
        printf("\n");
@@ -1488,11 +1513,11 @@ static void print_codecs(int encoder)

        while ((codec = next_codec_for_id(desc->id, codec, encoder))) {
            printf(" %c", get_media_type_char(desc->type));
-            printf((codec->capabilities & CODEC_CAP_FRAME_THREADS) ? "F" : ".");
-            printf((codec->capabilities & CODEC_CAP_SLICE_THREADS) ? "S" : ".");
-            printf((codec->capabilities & CODEC_CAP_EXPERIMENTAL)  ? "X" : ".");
-            printf((codec->capabilities & CODEC_CAP_DRAW_HORIZ_BAND)?"B" : ".");
-            printf((codec->capabilities & CODEC_CAP_DR1)           ? "D" : ".");
+            printf((codec->capabilities & AV_CODEC_CAP_FRAME_THREADS) ? "F" : ".");
+            printf((codec->capabilities & AV_CODEC_CAP_SLICE_THREADS) ? "S" : ".");
+            printf((codec->capabilities & AV_CODEC_CAP_EXPERIMENTAL)  ? "X" : ".");
+            printf((codec->capabilities & AV_CODEC_CAP_DRAW_HORIZ_BAND)?"B" : ".");
+            printf((codec->capabilities & AV_CODEC_CAP_DR1)           ? "D" : ".");

            printf(" %-20s %s", codec->name, codec->long_name ? codec->long_name : "");
            if (strcmp(codec->name, desc->name))
@@ -1535,10 +1560,10 @@ int show_protocols(void *optctx, const char *opt, const char *arg)
    printf("Supported file protocols:\n"
           "Input:\n");
    while ((name = avio_enum_protocols(&opaque, 0)))
-        printf("%s\n", name);
+        printf("  %s\n", name);
    printf("Output:\n");
    while ((name = avio_enum_protocols(&opaque, 1)))
-        printf("%s\n", name);
+        printf("  %s\n", name);
    return 0;
 }

@@ -1553,7 +1578,7 @@ int show_filters(void *optctx, const char *opt, const char *arg)
    printf("Filters:\n"
           "  T.. = Timeline support\n"
           "  .S. = Slice threading\n"
-           "  ..C = Commmand support\n"
+           "  ..C = Command support\n"
           "  A = Audio input/output\n"
           "  V = Video input/output\n"
           "  N = Dynamic number and/or type of input/output\n"
@@ -1566,10 +1591,10 @@ int show_filters(void *optctx, const char *opt, const char *arg)
                *(descr_cur++) = '>';
            }
            pad = i ? filter->outputs : filter->inputs;
-            for (j = 0; pad && pad[j].name; j++) {
+            for (j = 0; pad && avfilter_pad_get_name(pad, j); j++) {
                if (descr_cur >= descr + sizeof(descr) - 4)
                    break;
-                *(descr_cur++) = get_media_type_char(pad[j].type);
+                *(descr_cur++) = get_media_type_char(avfilter_pad_get_type(pad, j));
            }
            if (!j)
                *(descr_cur++) = ((!i && (filter->flags & AVFILTER_FLAG_DYNAMIC_INPUTS)) ||
@@ -1860,64 +1885,6 @@ int read_yesno(void)
    return yesno;
 }

-int cmdutils_read_file(const char *filename, char **bufptr, size_t *size)
-{
-    int64_t ret;
-    FILE *f = av_fopen_utf8(filename, "rb");
-
-    if (!f) {
-        ret = AVERROR(errno);
-        av_log(NULL, AV_LOG_ERROR, "Cannot read file '%s': %s\n", filename,
-               strerror(errno));
-        return ret;
-    }
-
-    ret = fseek(f, 0, SEEK_END);
-    if (ret == -1) {
-        ret = AVERROR(errno);
-        goto out;
-    }
-
-    ret = ftell(f);
-    if (ret < 0) {
-        ret = AVERROR(errno);
-        goto out;
-    }
-    *size = ret;
-
-    ret = fseek(f, 0, SEEK_SET);
-    if (ret == -1) {
-        ret = AVERROR(errno);
-        goto out;
-    }
-
-    *bufptr = av_malloc(*size + 1);
-    if (!*bufptr) {
-        av_log(NULL, AV_LOG_ERROR, "Could not allocate file buffer\n");
-        ret = AVERROR(ENOMEM);
-        goto out;
-    }
-    ret = fread(*bufptr, 1, *size, f);
-    if (ret < *size) {
-        av_free(*bufptr);
-        if (ferror(f)) {
-            ret = AVERROR(errno);
-            av_log(NULL, AV_LOG_ERROR, "Error while reading file '%s': %s\n",
-                   filename, strerror(errno));
-        } else
-            ret = AVERROR_EOF;
-    } else {
-        ret = 0;
-        (*bufptr)[(*size)++] = '\0';
-    }
-
-out:
-    if (ret < 0)
-        av_log(NULL, AV_LOG_ERROR, "IO error: %s\n", av_err2str(ret));
-    fclose(f);
-    return ret;
-}
-
 FILE *get_preset_file(char *filename, size_t filename_size,
                      const char *preset_name, int is_path,
                      const char *codec_name)
@@ -2072,6 +2039,33 @@ void *grow_array(void *array, int elem_size, int *size, int new_size)
    return array;
 }

+double get_rotation(AVStream *st)
+{
+    AVDictionaryEntry *rotate_tag = av_dict_get(st->metadata, "rotate", NULL, 0);
+    uint8_t* displaymatrix = av_stream_get_side_data(st,
+                                                     AV_PKT_DATA_DISPLAYMATRIX, NULL);
+    double theta = 0;
+
+    if (rotate_tag && *rotate_tag->value && strcmp(rotate_tag->value, "0")) {
+        char *tail;
+        theta = av_strtod(rotate_tag->value, &tail);
+        if (*tail)
+            theta = 0;
+    }
+    if (displaymatrix && !theta)
+        theta = -av_display_rotation_get((int32_t*) displaymatrix);
+
+    theta -= 360*floor(theta/360 + 0.9/360);
+
+    if (fabs(theta - 90*round(theta/90)) > 2)
+        av_log(NULL, AV_LOG_WARNING, "Odd rotation angle.\n"
+               "If you want to help, upload a sample "
+               "of this file to ftp://upload.ffmpeg.org/incoming/ "
+               "and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)");
+
+    return theta;
+}
+
 #if CONFIG_AVDEVICE
 static int print_device_sources(AVInputFormat *fmt, AVDictionary *opts)
 {
@@ -2081,7 +2075,7 @@ static int print_device_sources(AVInputFormat *fmt, AVDictionary *opts)
    if (!fmt || !fmt->priv_class  || !AV_IS_INPUT_DEVICE(fmt->priv_class->category))
        return AVERROR(EINVAL);

-    printf("Audo-detected sources for %s:\n", fmt->name);
+    printf("Auto-detected sources for %s:\n", fmt->name);
    if (!fmt->get_device_list) {
        ret = AVERROR(ENOSYS);
        printf("Cannot list sources. Not implemented.\n");
@@ -2111,7 +2105,7 @@ static int print_device_sinks(AVOutputFormat *fmt, AVDictionary *opts)
    if (!fmt || !fmt->priv_class  || !AV_IS_OUTPUT_DEVICE(fmt->priv_class->category))
        return AVERROR(EINVAL);

-    printf("Audo-detected sinks for %s:\n", fmt->name);
+    printf("Auto-detected sinks for %s:\n", fmt->name);
    if (!fmt->get_device_list) {
        ret = AVERROR(ENOSYS);
        printf("Cannot list sinks. Not implemented.\n");
@@ -2228,4 +2222,5 @@ int show_sinks(void *optctx, const char *opt, const char *arg)
    av_log_set_level(error_level);
    return ret;
 }
+
 #endif
@@ -46,7 +46,7 @@ extern const int program_birth_year;

 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
-extern struct SwsContext *sws_opts;
+extern AVDictionary *sws_dict;
 extern AVDictionary *swr_opts;
 extern AVDictionary *format_opts, *codec_opts, *resample_opts;
 extern int hide_banner;
@@ -61,6 +61,11 @@ void register_exit(void (*cb)(int ret));
 */
 void exit_program(int ret) av_noreturn;

+/**
+ * Initialize dynamic library loading
+ */
+void init_dynload(void);
+
 /**
 * Initialize the cmdutils option system, in particular
 * allocate the *_opts contexts.
@@ -277,7 +282,7 @@ typedef struct OptionGroup {
    AVDictionary *codec_opts;
    AVDictionary *format_opts;
    AVDictionary *resample_opts;
-    struct SwsContext *sws_opts;
+    AVDictionary *sws_dict;
    AVDictionary *swr_opts;
 } OptionGroup;

@@ -445,13 +450,13 @@ int show_devices(void *optctx, const char *opt, const char *arg);

 #if CONFIG_AVDEVICE
 /**
- * Print a listing containing audodetected sinks of the output device.
+ * Print a listing containing autodetected sinks of the output device.
 * Device name with options may be passed as an argument to limit results.
 */
 int show_sinks(void *optctx, const char *opt, const char *arg);

 /**
- * Print a listing containing audodetected sources of the input device.
+ * Print a listing containing autodetected sources of the input device.
 * Device name with options may be passed as an argument to limit results.
 */
 int show_sources(void *optctx, const char *opt, const char *arg);
@@ -529,18 +534,6 @@ int show_colors(void *optctx, const char *opt, const char *arg);
 */
 int read_yesno(void);

-/**
- * Read the file with name filename, and put its content in a newly
- * allocated 0-terminated buffer.
- *
- * @param filename file to read from
- * @param bufptr location where pointer to buffer is returned
- * @param size   location where size of buffer is returned
- * @return >= 0 in case of success, a negative value corresponding to an
- * AVERROR error code in case of failure.
- */
-int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
-
 /**
 * Get a file corresponding to a preset file.
 *
@@ -597,4 +590,6 @@ void *grow_array(void *array, int elem_size, int *size, int new_size);
    char name[128];\
    av_get_channel_layout_string(name, sizeof(name), 0, ch_layout);

+double get_rotation(AVStream *st);
+
 #endif /* CMDUTILS_H */
@@ -22,6 +22,7 @@
 #include "libavutil/time.h"
 #include "libavutil/log.h"
 #include "libavutil/opencl.h"
+#include "libavutil/avstring.h"
 #include "cmdutils.h"

 typedef struct {
@@ -238,7 +239,8 @@ int opt_opencl_bench(void *optctx, const char *opt, const char *arg)
                devices[count].platform_idx = i;
                devices[count].device_idx = j;
                devices[count].runtime = score;
-                strcpy(devices[count].device_name, device_node->device_name);
+                av_strlcpy(devices[count].device_name, device_node->device_name,
+                           sizeof(devices[count].device_name));
                count++;
            }
        }
@@ -7,7 +7,7 @@ all: all-yes

 DEFAULT_YASMD=.dbg

-ifeq (1, DBG)
+ifeq ($(DBG),1)
 YASMD=$(DEFAULT_YASMD)
 else
 YASMD=
@@ -118,8 +118,9 @@ TOOLOBJS  := $(TOOLS:%=tools/%.o)
 TOOLS     := $(TOOLS:%=tools/%$(EXESUF))
 HEADERS   += $(HEADERS-yes)

-PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
-DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))
+PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(2)LIBNAME))
+DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib),$(CONFIG_SHARED:yes=S)))
+STATIC_DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))

 SRC_DIR    := $(SRC_PATH)/lib$(NAME)
 ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
@@ -38,40 +38,9 @@
 #ifndef __AVISYNTH_C__
 #define __AVISYNTH_C__

-#ifdef __cplusplus
-#  define EXTERN_C extern "C"
-#else
-#  define EXTERN_C
-#endif
-
-#define AVSC_USE_STDCALL 1
-
-#ifndef AVSC_USE_STDCALL
-#  define AVSC_CC __cdecl
-#else
-#  define AVSC_CC __stdcall
-#endif
-
-#define AVSC_INLINE static __inline
-
-#ifdef AVISYNTH_C_EXPORTS
-#  define AVSC_EXPORT EXTERN_C
-#  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
-#else
-#  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
-#  ifndef AVSC_NO_DECLSPEC
-#    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
-#  else
-#    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
-#  endif
-#endif
-
-typedef unsigned char BYTE;
-#ifdef __GNUC__
-typedef long long int INT64;
-#else
-typedef __int64 INT64;
-#endif
+#include "avs/config.h"
+#include "avs/capi.h"
+#include "avs/types.h"


 /////////////////////////////////////////////////////////////////////
@@ -79,8 +48,8 @@ typedef __int64 INT64;
 // Constants
 //

-#ifndef __AVISYNTH_H__
-enum { AVISYNTH_INTERFACE_VERSION = 4 };
+#ifndef __AVISYNTH_6_H__
+enum { AVISYNTH_INTERFACE_VERSION = 6 };
 #endif

 enum {AVS_SAMPLE_INT8  = 1<<0,
@@ -112,8 +81,8 @@ enum {AVS_CS_BGR = 1<<28,
      AVS_CS_PLANAR = 1<<31,

      AVS_CS_SHIFT_SUB_WIDTH   = 0,
-      AVS_CS_SHIFT_SUB_HEIGHT  = 1 << 3,
-      AVS_CS_SHIFT_SAMPLE_BITS = 1 << 4,
+      AVS_CS_SHIFT_SUB_HEIGHT  = 8,
+      AVS_CS_SHIFT_SAMPLE_BITS = 16,

      AVS_CS_SUB_WIDTH_MASK    = 7 << AVS_CS_SHIFT_SUB_WIDTH,
      AVS_CS_SUB_WIDTH_1       = 3 << AVS_CS_SHIFT_SUB_WIDTH, // YV24
@@ -180,15 +149,66 @@ enum {  //SUBTYPES
  AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};

 enum {
-  AVS_CACHE_NOTHING=0,
-  AVS_CACHE_RANGE=1,
-  AVS_CACHE_ALL=2,
-  AVS_CACHE_AUDIO=3,
-  AVS_CACHE_AUDIO_NONE=4,
-  AVS_CACHE_AUDIO_AUTO=5
+  // New 2.6 explicitly defined cache hints.
+  AVS_CACHE_NOTHING=10, // Do not cache video.
+  AVS_CACHE_WINDOW=11, // Hard protect upto X frames within a range of X from the current frame N.
+  AVS_CACHE_GENERIC=12, // LRU cache upto X frames.
+  AVS_CACHE_FORCE_GENERIC=13, // LRU cache upto X frames, override any previous CACHE_WINDOW.
+
+  AVS_CACHE_GET_POLICY=30, // Get the current policy.
+  AVS_CACHE_GET_WINDOW=31, // Get the current window h_span.
+  AVS_CACHE_GET_RANGE=32, // Get the current generic frame range.
+
+  AVS_CACHE_AUDIO=50, // Explicitly do cache audio, X byte cache.
+  AVS_CACHE_AUDIO_NOTHING=51, // Explicitly do not cache audio.
+  AVS_CACHE_AUDIO_NONE=52, // Audio cache off (auto mode), X byte intial cache.
+  AVS_CACHE_AUDIO_AUTO=53, // Audio cache on (auto mode), X byte intial cache.
+
+  AVS_CACHE_GET_AUDIO_POLICY=70, // Get the current audio policy.
+  AVS_CACHE_GET_AUDIO_SIZE=71, // Get the current audio cache size.
+
+  AVS_CACHE_PREFETCH_FRAME=100, // Queue request to prefetch frame N.
+  AVS_CACHE_PREFETCH_GO=101, // Action video prefetches.
+
+  AVS_CACHE_PREFETCH_AUDIO_BEGIN=120, // Begin queue request transaction to prefetch audio (take critical section).
+  AVS_CACHE_PREFETCH_AUDIO_STARTLO=121, // Set low 32 bits of start.
+  AVS_CACHE_PREFETCH_AUDIO_STARTHI=122, // Set high 32 bits of start.
+  AVS_CACHE_PREFETCH_AUDIO_COUNT=123, // Set low 32 bits of length.
+  AVS_CACHE_PREFETCH_AUDIO_COMMIT=124, // Enqueue request transaction to prefetch audio (release critical section).
+  AVS_CACHE_PREFETCH_AUDIO_GO=125, // Action audio prefetches.
+
+  AVS_CACHE_GETCHILD_CACHE_MODE=200, // Cache ask Child for desired video cache mode.
+  AVS_CACHE_GETCHILD_CACHE_SIZE=201, // Cache ask Child for desired video cache size.
+  AVS_CACHE_GETCHILD_AUDIO_MODE=202, // Cache ask Child for desired audio cache mode.
+  AVS_CACHE_GETCHILD_AUDIO_SIZE=203, // Cache ask Child for desired audio cache size.
+
+  AVS_CACHE_GETCHILD_COST=220, // Cache ask Child for estimated processing cost.
+    AVS_CACHE_COST_ZERO=221, // Child response of zero cost (ptr arithmetic only).
+    AVS_CACHE_COST_UNIT=222, // Child response of unit cost (less than or equal 1 full frame blit).
+    AVS_CACHE_COST_LOW=223, // Child response of light cost. (Fast)
+    AVS_CACHE_COST_MED=224, // Child response of medium cost. (Real time)
+    AVS_CACHE_COST_HI=225, // Child response of heavy cost. (Slow)
+
+  AVS_CACHE_GETCHILD_THREAD_MODE=240, // Cache ask Child for thread safetyness.
+    AVS_CACHE_THREAD_UNSAFE=241, // Only 1 thread allowed for all instances. 2.5 filters default!
+    AVS_CACHE_THREAD_CLASS=242, // Only 1 thread allowed for each instance. 2.6 filters default!
+    AVS_CACHE_THREAD_SAFE=243, //  Allow all threads in any instance.
+    AVS_CACHE_THREAD_OWN=244, // Safe but limit to 1 thread, internally threaded.
+
+  AVS_CACHE_GETCHILD_ACCESS_COST=260, // Cache ask Child for preferred access pattern.
+    AVS_CACHE_ACCESS_RAND=261, // Filter is access order agnostic.
+    AVS_CACHE_ACCESS_SEQ0=262, // Filter prefers sequential access (low cost)
+    AVS_CACHE_ACCESS_SEQ1=263, // Filter needs sequential access (high cost)
  };

-#define AVS_FRAME_ALIGN 16
+#ifdef BUILDING_AVSCORE
+struct AVS_ScriptEnvironment {
+        IScriptEnvironment * env;
+        const char * error;
+        AVS_ScriptEnvironment(IScriptEnvironment * e = 0)
+                : env(e), error(0) {}
+};
+#endif

 typedef struct AVS_Clip AVS_Clip;
 typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
@@ -238,29 +258,23 @@ AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
 AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
        { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }

-AVSC_INLINE int avs_is_yv24(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV24  & AVS_CS_PLANAR_FILTER); }
+AVSC_API(int, avs_is_yv24)(const AVS_VideoInfo * p);

-AVSC_INLINE int avs_is_yv16(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV16  & AVS_CS_PLANAR_FILTER); }
+AVSC_API(int, avs_is_yv16)(const AVS_VideoInfo * p);

-AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV12  & AVS_CS_PLANAR_FILTER); }
+AVSC_API(int, avs_is_yv12)(const AVS_VideoInfo * p) ;

-AVSC_INLINE int avs_is_yv411(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV411 & AVS_CS_PLANAR_FILTER); }
+AVSC_API(int, avs_is_yv411)(const AVS_VideoInfo * p);

-AVSC_INLINE int avs_is_y8(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_Y8    & AVS_CS_PLANAR_FILTER); }
+AVSC_API(int, avs_is_y8)(const AVS_VideoInfo * p);

 AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
-        { return ((p->pixel_type & property)==property ); }
+        { return ((p->image_type & property)==property ); }

 AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
        { return !!(p->pixel_type & AVS_CS_PLANAR); }

-AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
-        { return avs_is_planar(p) ? ((p->pixel_type & AVS_CS_PLANAR_MASK) == (c_space & AVS_CS_PLANAR_FILTER)) : ((p->pixel_type & c_space) == c_space); }
+AVSC_API(int, avs_is_color_space)(const AVS_VideoInfo * p, int c_space);

 AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
        { return !!(p->image_type & AVS_IT_FIELDBASED); }
@@ -274,25 +288,18 @@ AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
 AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
        { return !!(p->image_type & AVS_IT_TFF); }

-AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
-{
-  switch (p->pixel_type) {
-      case AVS_CS_BGR24: return 24;
-      case AVS_CS_BGR32: return 32;
-      case AVS_CS_YUY2:  return 16;
-      case AVS_CS_YV12:
-      case AVS_CS_I420:  return 12;
-      default:           return 0;
-    }
-}
-AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
-        { return pixels * (avs_bits_per_pixel(p)>>3); }   // Will work on planar images, but will return only luma planes
+AVSC_API(int, avs_get_plane_width_subsampling)(const AVS_VideoInfo * p, int plane);

-AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
-        { return avs_bytes_from_pixels(p,p->width); }  // Also only returns first plane on planar images
+AVSC_API(int, avs_get_plane_height_subsampling)(const AVS_VideoInfo * p, int plane);

-AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
-        { if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p;  } return vi->height * ((avs_row_size(vi)+3) & ~3); }
+
+AVSC_API(int, avs_bits_per_pixel)(const AVS_VideoInfo * p);
+
+AVSC_API(int, avs_bytes_from_pixels)(const AVS_VideoInfo * p, int pixels);
+
+AVSC_API(int, avs_row_size)(const AVS_VideoInfo * p, int plane);
+
+AVSC_API(int, avs_bmp_size)(const AVS_VideoInfo * vi);

 AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
        { return p->audio_samples_per_second; }
@@ -350,11 +357,13 @@ AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned den
    p->fps_denominator = denominator/x;
 }

+#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
 AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
 {
        return (x->pixel_type == y->pixel_type)
                || (avs_is_yv12(x) && avs_is_yv12(y));
 }
+#endif

 /////////////////////////////////////////////////////////////////////
 //
@@ -391,89 +400,38 @@ typedef struct AVS_VideoFrame {
 } AVS_VideoFrame;

 // Access functions for AVS_VideoFrame
-AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
-        return p->pitch;}
+AVSC_API(int, avs_get_pitch_p)(const AVS_VideoFrame * p, int plane);

-AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
-  switch (plane) {
-  case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
-  return p->pitch;}
+#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
+AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
+        return avs_get_pitch_p(p, 0);}
+#endif
+
+AVSC_API(int, avs_get_row_size_p)(const AVS_VideoFrame * p, int plane);

 AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
        return p->row_size; }

-AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
-        int r;
-    switch (plane) {
-    case AVS_PLANAR_U: case AVS_PLANAR_V:
-                if (p->pitchUV) return p->row_sizeUV;
-                else            return 0;
-    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
-                if (p->pitchUV) {
-                        r = (p->row_sizeUV+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-                        if (r < p->pitchUV)
-                                return r;
-                        return p->row_sizeUV;
-                } else return 0;
-    case AVS_PLANAR_Y_ALIGNED:
-                r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-                if (r <= p->pitch)
-                        return r;
-                return p->row_size;
-    }
-    return p->row_size;
-}
+AVSC_API(int, avs_get_height_p)(const AVS_VideoFrame * p, int plane);

 AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
        return p->height;}

-AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
-        switch (plane) {
-                case AVS_PLANAR_U: case AVS_PLANAR_V:
-                        if (p->pitchUV) return p->heightUV;
-                        return 0;
-        }
-        return p->height;}
+AVSC_API(const BYTE *, avs_get_read_ptr_p)(const AVS_VideoFrame * p, int plane);

+#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
 AVSC_INLINE const BYTE* avs_get_read_ptr(const AVS_VideoFrame * p) {
-        return p->vfb->data + p->offset;}
+        return avs_get_read_ptr_p(p, 0);}
+#endif

-AVSC_INLINE const BYTE* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
-{
-        switch (plane) {
-                case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
-                case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
-                default:           return p->vfb->data + p->offset;}
-}
+AVSC_API(int, avs_is_writable)(const AVS_VideoFrame * p);

-AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
-        return (p->refcount == 1 && p->vfb->refcount == 1);}
-
-AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p)
-{
-        if (avs_is_writable(p)) {
-                ++p->vfb->sequence_number;
-                return p->vfb->data + p->offset;
-        } else
-                return 0;
-}
-
-AVSC_INLINE BYTE* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
-{
-        if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
-                ++p->vfb->sequence_number;
-                return p->vfb->data + p->offset;
-        } else if (plane==AVS_PLANAR_Y) {
-                return 0;
-        } else {
-                switch (plane) {
-                        case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
-                        case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
-                        default:       return p->vfb->data + p->offset;
-                }
-        }
-}
+AVSC_API(BYTE *, avs_get_write_ptr_p)(const AVS_VideoFrame * p, int plane);

+#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
+AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p) {
+        return avs_get_write_ptr_p(p, 0);}
+#endif

 AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
 // makes a shallow copy of a video frame
@@ -658,12 +616,16 @@ enum {
  AVS_CPUF_SSSE3      = 0x200,   //  Core 2
  AVS_CPUF_SSE4       = 0x400,   //  Penryn, Wolfdale, Yorkfield
  AVS_CPUF_SSE4_1     = 0x400,
-  AVS_CPUF_SSE4_2     = 0x800,   //  Nehalem
+//AVS_CPUF_AVX        = 0x800,   //  Sandy Bridge, Bulldozer
+  AVS_CPUF_SSE4_2    = 0x1000,   //  Nehalem
+//AVS_CPUF_AVX2      = 0x2000,   //  Haswell
+//AVS_CPUF_AVX512    = 0x4000,   //  Knights Landing
 };

+
 AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error

-AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
+AVSC_API(int, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
 AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);

 AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
@@ -700,12 +662,12 @@ AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
 AVSC_INLINE
 AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
                                     const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
+  {return avs_new_video_frame_a(env,vi,FRAME_ALIGN);}

 AVSC_INLINE
 AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
                               const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
+  {return avs_new_video_frame_a(env,vi,FRAME_ALIGN);}
 #endif


@@ -773,7 +735,6 @@ struct AVS_Library {
  AVSC_DECLARE_FUNC(avs_function_exists);
  AVSC_DECLARE_FUNC(avs_get_audio);
  AVSC_DECLARE_FUNC(avs_get_cpu_flags);
-  AVSC_DECLARE_FUNC(avs_get_error);
  AVSC_DECLARE_FUNC(avs_get_frame);
  AVSC_DECLARE_FUNC(avs_get_parity);
  AVSC_DECLARE_FUNC(avs_get_var);
@@ -798,6 +759,27 @@ struct AVS_Library {
  AVSC_DECLARE_FUNC(avs_subframe_planar);
  AVSC_DECLARE_FUNC(avs_take_clip);
  AVSC_DECLARE_FUNC(avs_vsprintf);
+
+  AVSC_DECLARE_FUNC(avs_get_error);
+  AVSC_DECLARE_FUNC(avs_is_yv24);
+  AVSC_DECLARE_FUNC(avs_is_yv16);
+  AVSC_DECLARE_FUNC(avs_is_yv12);
+  AVSC_DECLARE_FUNC(avs_is_yv411);
+  AVSC_DECLARE_FUNC(avs_is_y8);
+  AVSC_DECLARE_FUNC(avs_is_color_space);
+
+  AVSC_DECLARE_FUNC(avs_get_plane_width_subsampling);
+  AVSC_DECLARE_FUNC(avs_get_plane_height_subsampling);
+  AVSC_DECLARE_FUNC(avs_bits_per_pixel);
+  AVSC_DECLARE_FUNC(avs_bytes_from_pixels);
+  AVSC_DECLARE_FUNC(avs_row_size);
+  AVSC_DECLARE_FUNC(avs_bmp_size);
+  AVSC_DECLARE_FUNC(avs_get_pitch_p);
+  AVSC_DECLARE_FUNC(avs_get_row_size_p);
+  AVSC_DECLARE_FUNC(avs_get_height_p);
+  AVSC_DECLARE_FUNC(avs_get_read_ptr_p);
+  AVSC_DECLARE_FUNC(avs_is_writable);
+  AVSC_DECLARE_FUNC(avs_get_write_ptr_p);
 };

 #undef AVSC_DECLARE_FUNC
@@ -805,7 +787,7 @@ struct AVS_Library {

 AVSC_INLINE AVS_Library * avs_load_library() {
  AVS_Library *library = (AVS_Library *)malloc(sizeof(AVS_Library));
-  if (!library)
+  if (library == NULL)
    return NULL;
  library->handle = LoadLibrary("avisynth");
  if (library->handle == NULL)
@@ -832,7 +814,6 @@ AVSC_INLINE AVS_Library * avs_load_library() {
  AVSC_LOAD_FUNC(avs_function_exists);
  AVSC_LOAD_FUNC(avs_get_audio);
  AVSC_LOAD_FUNC(avs_get_cpu_flags);
-  AVSC_LOAD_FUNC(avs_get_error);
  AVSC_LOAD_FUNC(avs_get_frame);
  AVSC_LOAD_FUNC(avs_get_parity);
  AVSC_LOAD_FUNC(avs_get_var);
@@ -858,6 +839,27 @@ AVSC_INLINE AVS_Library * avs_load_library() {
  AVSC_LOAD_FUNC(avs_take_clip);
  AVSC_LOAD_FUNC(avs_vsprintf);

+  AVSC_LOAD_FUNC(avs_get_error);
+  AVSC_LOAD_FUNC(avs_is_yv24);
+  AVSC_LOAD_FUNC(avs_is_yv16);
+  AVSC_LOAD_FUNC(avs_is_yv12);
+  AVSC_LOAD_FUNC(avs_is_yv411);
+  AVSC_LOAD_FUNC(avs_is_y8);
+  AVSC_LOAD_FUNC(avs_is_color_space);
+
+  AVSC_LOAD_FUNC(avs_get_plane_width_subsampling);
+  AVSC_LOAD_FUNC(avs_get_plane_height_subsampling);
+  AVSC_LOAD_FUNC(avs_bits_per_pixel);
+  AVSC_LOAD_FUNC(avs_bytes_from_pixels);
+  AVSC_LOAD_FUNC(avs_row_size);
+  AVSC_LOAD_FUNC(avs_bmp_size);
+  AVSC_LOAD_FUNC(avs_get_pitch_p);
+  AVSC_LOAD_FUNC(avs_get_row_size_p);
+  AVSC_LOAD_FUNC(avs_get_height_p);
+  AVSC_LOAD_FUNC(avs_get_read_ptr_p);
+  AVSC_LOAD_FUNC(avs_is_writable);
+  AVSC_LOAD_FUNC(avs_get_write_ptr_p);
+
 #undef __AVSC_STRINGIFY
 #undef AVSC_STRINGIFY
 #undef AVSC_LOAD_FUNC
@@ -870,7 +872,7 @@ fail:
 }

 AVSC_INLINE void avs_free_library(AVS_Library *library) {
-  if (!library)
+  if (library == NULL)
    return;
  FreeLibrary(library->handle);
  free(library);
@@ -1,68 +0,0 @@
-//  Copyright (c) 2011 FFmpegSource Project
-//
-//  Permission is hereby granted, free of charge, to any person obtaining a copy
-//  of this software and associated documentation files (the "Software"), to deal
-//  in the Software without restriction, including without limitation the rights
-//  to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
-//  copies of the Software, and to permit persons to whom the Software is
-//  furnished to do so, subject to the following conditions:
-//
-//  The above copyright notice and this permission notice shall be included in
-//  all copies or substantial portions of the Software.
-//
-//  THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
-//  IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
-//  FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
-//  AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
-//  LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
-//  OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
-//  THE SOFTWARE.
-
-/* these are defines/functions that are used and were changed in the switch to 2.6
- * and are needed to maintain full compatility with 2.5 */
-
-enum {
-  AVS_CS_YV12_25 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-v-u, planar
-  AVS_CS_I420_25 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-u-v, planar
-};
-
-AVSC_INLINE int avs_get_height_p_25(const AVS_VideoFrame * p, int plane) {
-    switch (plane)
-    {
-        case AVS_PLANAR_U: case AVS_PLANAR_V:
-            if (p->pitchUV)
-                return p->height>>1;
-            return 0;
-    }
-    return p->height;}
-
-AVSC_INLINE int avs_get_row_size_p_25(const AVS_VideoFrame * p, int plane) {
-    int r;
-    switch (plane)
-    {
-    case AVS_PLANAR_U: case AVS_PLANAR_V:
-        if (p->pitchUV)
-            return p->row_size>>1;
-        else
-            return 0;
-    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
-        if (p->pitchUV)
-        {
-            r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize
-            if (r < p->pitchUV)
-                return r;
-            return p->row_size>>1;
-        }
-        else
-            return 0;
-    case AVS_PLANAR_Y_ALIGNED:
-        r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-        if (r <= p->pitch)
-            return r;
-        return p->row_size;
-    }
-    return p->row_size;
-}
-
-AVSC_INLINE int avs_is_yv12_25(const AVS_VideoInfo * p)
-    { return ((p->pixel_type & AVS_CS_YV12_25) == AVS_CS_YV12_25)||((p->pixel_type & AVS_CS_I420_25) == AVS_CS_I420_25); }
@@ -0,0 +1,62 @@
+// Avisynth C Interface Version 0.20
+// Copyright 2003 Kevin Atkinson
+
+// This program is free software; you can redistribute it and/or modify
+// it under the terms of the GNU General Public License as published by
+// the Free Software Foundation; either version 2 of the License, or
+// (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+// GNU General Public License for more details.
+//
+// You should have received a copy of the GNU General Public License
+// along with this program; if not, write to the Free Software
+// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
+// http://www.gnu.org/copyleft/gpl.html .
+//
+// As a special exception, I give you permission to link to the
+// Avisynth C interface with independent modules that communicate with
+// the Avisynth C interface solely through the interfaces defined in
+// avisynth_c.h, regardless of the license terms of these independent
+// modules, and to copy and distribute the resulting combined work
+// under terms of your choice, provided that every copy of the
+// combined work is accompanied by a complete copy of the source code
+// of the Avisynth C interface and Avisynth itself (with the version
+// used to produce the combined work), being distributed under the
+// terms of the GNU General Public License plus this exception.  An
+// independent module is a module which is not derived from or based
+// on Avisynth C Interface, such as 3rd-party filters, import and
+// export plugins, or graphical user interfaces.
+
+#ifndef AVS_CAPI_H
+#define AVS_CAPI_H
+
+#ifdef __cplusplus
+#  define EXTERN_C extern "C"
+#else
+#  define EXTERN_C
+#endif
+
+#ifndef AVSC_USE_STDCALL
+#  define AVSC_CC __cdecl
+#else
+#  define AVSC_CC __stdcall
+#endif
+
+#define AVSC_INLINE static __inline
+
+#ifdef BUILDING_AVSCORE
+#  define AVSC_EXPORT EXTERN_C
+#  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
+#else
+#  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
+#  ifndef AVSC_NO_DECLSPEC
+#    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
+#  else
+#    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
+#  endif
+#endif
+
+#endif //AVS_CAPI_H
@@ -0,0 +1,55 @@
+// Avisynth C Interface Version 0.20
+// Copyright 2003 Kevin Atkinson
+
+// This program is free software; you can redistribute it and/or modify
+// it under the terms of the GNU General Public License as published by
+// the Free Software Foundation; either version 2 of the License, or
+// (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+// GNU General Public License for more details.
+//
+// You should have received a copy of the GNU General Public License
+// along with this program; if not, write to the Free Software
+// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
+// http://www.gnu.org/copyleft/gpl.html .
+//
+// As a special exception, I give you permission to link to the
+// Avisynth C interface with independent modules that communicate with
+// the Avisynth C interface solely through the interfaces defined in
+// avisynth_c.h, regardless of the license terms of these independent
+// modules, and to copy and distribute the resulting combined work
+// under terms of your choice, provided that every copy of the
+// combined work is accompanied by a complete copy of the source code
+// of the Avisynth C interface and Avisynth itself (with the version
+// used to produce the combined work), being distributed under the
+// terms of the GNU General Public License plus this exception.  An
+// independent module is a module which is not derived from or based
+// on Avisynth C Interface, such as 3rd-party filters, import and
+// export plugins, or graphical user interfaces.
+
+#ifndef AVS_CONFIG_H
+#define AVS_CONFIG_H
+
+// Undefine this to get cdecl calling convention
+#define AVSC_USE_STDCALL 1
+
+// NOTE TO PLUGIN AUTHORS:
+// Because FRAME_ALIGN can be substantially higher than the alignment
+// a plugin actually needs, plugins should not use FRAME_ALIGN to check for
+// alignment. They should always request the exact alignment value they need.
+// This is to make sure that plugins work over the widest range of AviSynth
+// builds possible.
+#define FRAME_ALIGN 32
+
+#if   defined(_M_AMD64) || defined(__x86_64)
+#   define X86_64
+#elif defined(_M_IX86) || defined(__i386__)
+#   define X86_32
+#else
+#   error Unsupported CPU architecture.
+#endif
+
+#endif //AVS_CONFIG_H
@@ -0,0 +1,51 @@
+// Avisynth C Interface Version 0.20
+// Copyright 2003 Kevin Atkinson
+
+// This program is free software; you can redistribute it and/or modify
+// it under the terms of the GNU General Public License as published by
+// the Free Software Foundation; either version 2 of the License, or
+// (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+// GNU General Public License for more details.
+//
+// You should have received a copy of the GNU General Public License
+// along with this program; if not, write to the Free Software
+// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
+// http://www.gnu.org/copyleft/gpl.html .
+//
+// As a special exception, I give you permission to link to the
+// Avisynth C interface with independent modules that communicate with
+// the Avisynth C interface solely through the interfaces defined in
+// avisynth_c.h, regardless of the license terms of these independent
+// modules, and to copy and distribute the resulting combined work
+// under terms of your choice, provided that every copy of the
+// combined work is accompanied by a complete copy of the source code
+// of the Avisynth C interface and Avisynth itself (with the version
+// used to produce the combined work), being distributed under the
+// terms of the GNU General Public License plus this exception.  An
+// independent module is a module which is not derived from or based
+// on Avisynth C Interface, such as 3rd-party filters, import and
+// export plugins, or graphical user interfaces.
+
+#ifndef AVS_TYPES_H
+#define AVS_TYPES_H
+
+// Define all types necessary for interfacing with avisynth.dll
+
+// Raster types used by VirtualDub & Avisynth
+typedef unsigned int    Pixel32;
+typedef unsigned char   BYTE;
+
+// Audio Sample information
+typedef float SFLOAT;
+
+#ifdef __GNUC__
+typedef long long int INT64;
+#else
+typedef __int64 INT64;
+#endif
+
+#endif //AVS_TYPES_H
@@ -66,6 +66,8 @@ static av_always_inline int pthread_create(pthread_t *thread, const pthread_attr
    struct thread_arg *thread_arg;

    thread_arg = av_mallocz(sizeof(struct thread_arg));
+    if (!thread_arg)
+        return ENOMEM;

    thread_arg->start_routine = start_routine;
    thread_arg->arg = arg;
@@ -0,0 +1,9 @@
+#!/bin/sh
+
+LINK_EXE_PATH=$(dirname "$(command -v cl)")/link
+if [ -x "$LINK_EXE_PATH" ]; then
+    "$LINK_EXE_PATH" $@
+else
+    link $@
+fi
+exit $?
@@ -15,6 +15,98 @@ libavutil:     2014-08-09

 API changes, most recent first:

+-------- 8< --------- FFmpeg 2.8 was cut here -------- 8< ---------
+
+2015-08-27 - 1dd854e1 - lavc 56.58.100 - vaapi.h
+  Deprecate old VA-API context (vaapi_context) fields that were only
+  set and used by libavcodec. They are all managed internally now.
+
+2015-08-19 - 9f8e57ef - lavu 54.31.100 - pixfmt.h
+  Add a unique pixel format for VA-API (AV_PIX_FMT_VAAPI) that
+  indicates the nature of the underlying storage: a VA surface. This
+  yields the same value as AV_PIX_FMT_VAAPI_VLD.
+  Deprecate old VA-API related pixel formats: AV_PIX_FMT_VAAPI_MOCO,
+  AV_PIX_FMT_VAAPI_IDCT, AV_PIX_FMT_VAAPI_VLD.
+
+2015-08-02 - lavu 54.30.100 / 54.17.0
+  9ed59f1 / 7a7df34c -  Add av_blowfish_alloc().
+  a130ec9 / ae365453 -  Add av_rc4_alloc().
+  9ca1997 / 5d8bea3b -  Add av_xtea_alloc().
+  3cf08e9 / d9e8b47e -  Add av_des_alloc().
+
+2015-07-27 - lavc 56.56.100 / 56.35.0 - avcodec.h
+  94d68a4 / 7c6eb0a1 - Rename CODEC_FLAG* defines to AV_CODEC_FLAG*.
+  444e987 / def97856 - Rename CODEC_CAP_* defines to AV_CODEC_CAP_*.
+  29d147c / 059a9348 - Rename FF_INPUT_BUFFER_PADDING_SIZE and FF_MIN_BUFFER_SIZE
+              to AV_INPUT_BUFFER_PADDING_SIZE and AV_INPUT_BUFFER_MIN_SIZE.
+
+2015-07-22 - c40ecff - lavc 56.51.100 - avcodec.h
+  Add AV_PKT_DATA_QUALITY_STATS to export the quality value, PSNR, and pict_type
+  of an AVPacket.
+
+2015-07-16 - 8dad213 - lavc 56.49.100
+  Add av_codec_get_codec_properties(), FF_CODEC_PROPERTY_LOSSLESS
+  and FF_CODEC_PROPERTY_CLOSED_CAPTIONS
+
+2015-07-03 - d563e13 / 83212943 - lavu 54.28.100 / 56.15.0
+  Add av_version_info().
+
+-------- 8< --------- FFmpeg 2.7 was cut here -------- 8< ---------
+
+2015-06-04 - cc17b43 - lswr  1.2.100
+  Add swr_get_out_samples()
+
+2015-05-27 - c312bfa - lavu 54.26.100 - cpu.h
+  Add AV_CPU_FLAG_AVXSLOW.
+
+2015-05-26 - 1fb9b2a - lavu 54.25.100 - rational.h
+  Add av_q2intfloat().
+
+2015-05-13 - cc48409 / e7c5e17 - lavc 56.39.100 / 56.23.0
+  Add av_vda_default_init2.
+
+2015-05-11 - 541d75f - lavf 56.33.100 - avformat.h
+  Add AVOpenCallback AVFormatContext.open_cb
+
+2015-05-07 - a7dd933 - 56.38.100 - avcodec.h
+  Add av_packet_side_data_name().
+
+2015-05-07 - 01e59d4 - 56.37.102 - avcodec.h
+  Add FF_PROFILE_VP9_2 and FF_PROFILE_VP9_3.
+
+2015-05-04 - 079b7f6 - 56.37.100 - avcodec.h
+  Add FF_PROFILE_VP9_0 and FF_PROFILE_VP9_1.
+
+2015-04-22 - 748d481 - lavf 56.31.100 - avformat.h
+  Add AVFMT_FLAG_FAST_SEEK flag. Some formats (initially mp3) use it to enable
+  fast, but inaccurate seeking.
+
+2015-04-20 - 8e8219e / c253340 - lavu 54.23.100 / 54.12.0 - log.h
+  Add AV_LOG_TRACE for extremely verbose debugging.
+
+2015-04-02 - 26e0e393 - lavf 56.29.100 - avio.h
+  Add AVIODirEntryType.AVIO_ENTRY_SERVER.
+  Add AVIODirEntryType.AVIO_ENTRY_SHARE.
+  Add AVIODirEntryType.AVIO_ENTRY_WORKGROUP.
+
+2015-03-31 - 3188696 - lavu 54.22.100 - avstring.h
+  Add av_append_path_component()
+
+2015-03-27 - 184084c - lavf 56.27.100 - avio.h url.h
+  New directory listing API.
+
+  Add AVIODirEntryType enum.
+  Add AVIODirEntry, AVIODirContext structures.
+  Add avio_open_dir(), avio_read_dir(), avio_close_dir(), avio_free_directory_entry().
+  Add ff_alloc_dir_entry().
+  Extend URLProtocol with url_open_dir(), url_read_dir(), url_close_dir().
+
+2015-03-29 - 268ff17 / c484561 - lavu 54.21.100 / 54.10.0 - pixfmt.h
+  Add AV_PIX_FMT_MMAL for MMAL hardware acceleration.
+
+2015-03-19 - 11fe56c - 56.29.100 / lavc 56.22.0
+  Add FF_PROFILE_DTS_EXPRESS.
+
 -------- 8< --------- FFmpeg 2.6 was cut here -------- 8< ---------

 2015-03-04 - cca4476 - lavf 56.25.100
@@ -140,7 +232,7 @@ API changes, most recent first:
  Add av_opt_get_dict_val/set_dict_val with AV_OPT_TYPE_DICT to support
  dictionary types being set as options.

-2014-08-13 - afbd4b8 - lavf 56.01.0 - avformat.h
+2014-08-13 - afbd4b7e09 - lavf 56.01.0 - avformat.h
  Add AVFormatContext.event_flags and AVStream.event_flags for signaling to
  the user when events happen in the file/stream.

@@ -157,7 +249,7 @@ API changes, most recent first:
 2014-08-08 - 5c3c671 - lavf 55.53.100 - avio.h
  Add avio_feof() and deprecate url_feof().

-2014-08-07 - bb78903 - lsws 2.1.3 - swscale.h
+2014-08-07 - bb789016d4 - lsws 2.1.3 - swscale.h
  sws_getContext is not going to be removed in the future.

 2014-08-07 - a561662 / ad1ee5f - lavc 55.73.101 / 55.57.3 - avcodec.h
@@ -636,6 +728,9 @@ API changes, most recent first:
   av_ripemd_update()
   av_ripemd_final()

+2013-06-10 - 82ef670 - lavu 52.35.101 - hmac.h
+  Add AV_HMAC_SHA224, AV_HMAC_SHA256, AV_HMAC_SHA384, AV_HMAC_SHA512
+
 2013-06-04 - 30b491f / fc962d4 - lavu 52.35.100 / 52.13.0 - mem.h
  Add av_realloc_array and av_reallocp_array

@@ -31,7 +31,7 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.

-PROJECT_NUMBER         =
+PROJECT_NUMBER         = 2.8.15

 # 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
@@ -36,6 +36,7 @@ DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
 DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
 DOCS = $(DOCS-yes)

+DOC_EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE)      += avio_dir_cmd
 DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
 DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE)           += avcodec
 DOC_EXAMPLES-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding
@@ -139,6 +139,26 @@ ffmpeg -i frame_%d.jpg -c:v copy rotated.avi

@section mp3_header_decompress

+@section mpeg4_unpack_bframes
+
+Unpack DivX-style packed B-frames.
+
+DivX-style packed B-frames are not valid MPEG-4 and were only a
+workaround for the broken Video for Windows subsystem.
+They use more space, can cause minor AV sync issues, require more
+CPU power to decode (unless the player has some decoded picture queue
+to compensate the 2,0,2,0 frame per packet style) and cause
+trouble if copied into a standard container like mp4 or mpeg-ps/ts,
+because MPEG-4 decoders may not be able to decode them, since they are
+not valid MPEG-4.
+
+For example to fix an AVI file containing an MPEG-4 stream with
+DivX-style packed B-frames using @command{ffmpeg}, you can use the command:
+
+@example
+ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi
+@end example
+
@section noise

 Damages the contents of packets without damaging the container. Can be
@@ -16,6 +16,9 @@ DESTDIR
    Destination directory for the install targets, useful to prepare packages
    or install FFmpeg in cross-environments.

+GEN
+    Set to ‘1’ to generate the missing or mismatched references.
+
 Makefile targets:

 all
@@ -475,6 +475,9 @@ per-block quantization parameter (QP)
 motion vector
@item dct_coeff

+@item green_metadata
+display complexity metadata for the upcoming frame, GoP or for a given duration.
+
@item skip

@item startcode
@@ -1042,7 +1045,11 @@ Possible values:
@item color_primaries @var{integer} (@emph{decoding/encoding,video})
@item color_trc @var{integer} (@emph{decoding/encoding,video})
@item colorspace @var{integer} (@emph{decoding/encoding,video})
+
@item color_range @var{integer} (@emph{decoding/encoding,video})
+If used as input parameter, it serves as a hint to the decoder, which
+color_range the input has.
+
@item chroma_sample_location @var{integer} (@emph{decoding/encoding,video})

@item log_level_offset @var{integer}
@@ -25,6 +25,13 @@ enabled decoders.
 A description of some of the currently available video decoders
 follows.

+@section hevc
+
+HEVC / H.265 decoder.
+
+Note: the @option{skip_loop_filter} option has effect only at level
+@code{all}.
+
@section rawvideo

 Raw video decoder.
@@ -83,6 +90,23 @@ Loud sounds are fully compressed.  Soft sounds are enhanced.

@end table

+@section flac
+
+FLAC audio decoder.
+
+This decoder aims to implement the complete FLAC specification from Xiph.
+
+@subsection FLAC Decoder options
+
+@table @option
+
+@item -use_buggy_lpc
+The lavc FLAC encoder used to produce buggy streams with high lpc values
+(like the default value). This option makes it possible to decode such streams
+correctly by using lavc's old buggy lpc logic for decoding.
+
+@end table
+
@section ffwavesynth

 Internal wave synthetizer.
@@ -171,6 +195,25 @@ without this library.
@chapter Subtitles Decoders
@c man begin SUBTILES DECODERS

+@section dvbsub
+
+@subsection Options
+
+@table @option
+@item compute_clut
+@table @option
+@item -1
+Compute clut if no matching CLUT is in the stream.
+@item 0
+Never compute CLUT
+@item 1
+Always compute CLUT and override the one provided in the stream.
+@end table
+@item dvb_substream
+Selects the dvb substream, or all substreams if -1 which is default.
+
+@end table
+
@section dvdsub

 This codec decodes the bitmap subtitles used in DVDs; the same subtitles can
@@ -18,6 +18,12 @@ enabled demuxers.

 The description of some of the currently available demuxers follows.

+@section aa
+
+Audible Format 2, 3, and 4 demuxer.
+
+This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
+
@section applehttp

 Apple HTTP Live Streaming demuxer.
@@ -98,7 +104,7 @@ All subsequent file-related directives apply to that file.

@item @code{ffconcat version 1.0}
 Identify the script type and version. It also sets the @option{safe} option
-to 1 if it was to its default -1.
+to 1 if it was -1.

 To make FFmpeg recognize the format automatically, this directive must
 appears exactly as is (no extra space or byte-order-mark) on the very first
@@ -112,6 +118,47 @@ file is not available or accurate.
 If the duration is set for all files, then it is possible to seek in the
 whole concatenated video.

+@item @code{inpoint @var{timestamp}}
+In point of the file. When the demuxer opens the file it instantly seeks to the
+specified timestamp. Seeking is done so that all streams can be presented
+successfully at In point.
+
+This directive works best with intra frame codecs, because for non-intra frame
+ones you will usually get extra packets before the actual In point and the
+decoded content will most likely contain frames before In point too.
+
+For each file, packets before the file In point will have timestamps less than
+the calculated start timestamp of the file (negative in case of the first
+file), and the duration of the files (if not specified by the @code{duration}
+directive) will be reduced based on their specified In point.
+
+Because of potential packets before the specified In point, packet timestamps
+may overlap between two concatenated files.
+
+@item @code{outpoint @var{timestamp}}
+Out point of the file. When the demuxer reaches the specified decoding
+timestamp in any of the streams, it handles it as an end of file condition and
+skips the current and all the remaining packets from all streams.
+
+Out point is exclusive, which means that the demuxer will not output packets
+with a decoding timestamp greater or equal to Out point.
+
+This directive works best with intra frame codecs and formats where all streams
+are tightly interleaved. For non-intra frame codecs you will usually get
+additional packets with presentation timestamp after Out point therefore the
+decoded content will most likely contain frames after Out point too. If your
+streams are not tightly interleaved you may not get all the packets from all
+streams before Out point and you may only will be able to decode the earliest
+stream until Out point.
+
+The duration of the files (if not specified by the @code{duration}
+directive) will be reduced based on their specified Out point.
+
+@item @code{file_packet_metadata @var{key=value}}
+Metadata of the packets of the file. The specified metadata will be set for
+each file packet. You can specify this directive multiple times to add multiple
+metadata entries.
+
@item @code{stream}
 Introduce a stream in the virtual file.
 All subsequent stream-related directives apply to the last introduced
@@ -145,12 +192,15 @@ component.

 If set to 0, any file name is accepted.

-The default is -1, it is equivalent to 1 if the format was automatically
+The default is 1.
+
+-1 is equivalent to 1 if the format was automatically
 probed and 0 otherwise.

@item auto_convert
 If set to 1, try to perform automatic conversions on packet data to make the
 streams concatenable.
+The default is 1.

 Currently, the only conversion is adding the h264_mp4toannexb bitstream
 filter to H.264 streams in MP4 format. This is necessary in particular if
@@ -205,6 +255,11 @@ It accepts the following options:
 Set the minimum valid delay between frames in hundredths of seconds.
 Range is 0 to 6000. Default value is 2.

+@item max_gif_delay
+Set the maximum valid delay between frames in hundredth of seconds.
+Range is 0 to 65535. Default value is 65535 (nearly eleven minutes),
+the maximum value allowed by the specification.
+
@item default_delay
 Set the default delay between frames in hundredths of seconds.
 Range is 0 to 6000. Default value is 10.
@@ -228,6 +283,24 @@ used to end the output video at the length of the shortest input file,
 which in this case is @file{input.mp4} as the GIF in this example loops
 infinitely.

+@section hls
+
+HLS demuxer
+
+It accepts the following options:
+
+@table @option
+@item live_start_index
+segment index to start live streams at (negative values are from the end).
+
+@item allowed_extensions
+',' separated list of file extensions that hls is allowed to access.
+
+@item max_reload
+Maximum number of times a insufficient list is attempted to be reloaded.
+Default value is 1000.
+@end table
+
@section image2

 Image file demuxer.
@@ -253,6 +326,10 @@ Select the pattern type used to interpret the provided filename.

@var{pattern_type} accepts one of the following values.
@table @option
+@item none
+Disable pattern matching, therefore the video will only contain the specified
+image. You should use this option if you do not want to create sequences from
+multiple images and your filenames may contain special pattern characters.
@item sequence
 Select a sequence pattern type, used to specify a sequence of files
 indexed by sequential numbers.
@@ -359,17 +436,47 @@ ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
@end example
@end itemize

+@section mov/mp4/3gp/Quicktme
+
+Quicktime / MP4 demuxer.
+
+This demuxer accepts the following options:
+@table @option
+@item enable_drefs
+Enable loading of external tracks, disabled by default.
+Enabling this can theoretically leak information in some use cases.
+
+@item use_absolute_path
+Allows loading of external tracks via absolute paths, disabled by default.
+Enabling this poses a security risk. It should only be enabled if the source
+is known to be non malicious.
+
+@end table
+
@section mpegts

 MPEG-2 transport stream demuxer.

+This demuxer accepts the following options:
@table @option
+@item resync_size
+Set size limit for looking up a new synchronization. Default value is
+65536.

@item fix_teletext_pts
-Overrides teletext packet PTS and DTS values with the timestamps calculated
+Override teletext packet PTS and DTS values with the timestamps calculated
 from the PCR of the first program which the teletext stream is part of and is
 not discarded. Default value is 1, set this option to 0 if you want your
 teletext packet PTS and DTS values untouched.
+
+@item ts_packetsize
+Output option carrying the raw packet size in bytes.
+Show the detected raw packet size, cannot be set by the user.
+
+@item scan_all_pmts
+Scan and combine all PMTs. The value is an integer with value from -1
+to 1 (-1 means automatic setting, 1 means enabled, 0 means
+disabled). Default value is -1.
@end table

@section rawvideo
@@ -228,7 +228,7 @@ autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
@end example

 For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
-@example
+@lisp
 (c-add-style "ffmpeg"
             '("k&r"
               (c-basic-offset . 4)
@@ -239,7 +239,7 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
               )
             )
 (setq c-default-style "ffmpeg")
-@end example
+@end lisp

@section Development Policy

@@ -400,6 +400,35 @@ finding a new maintainer and also don't forget updating the @file{MAINTAINERS} f

 We think our rules are not too hard. If you have comments, contact us.

+@section Code of conduct
+
+Be friendly and respectful towards others and third parties.
+Treat others the way you yourself want to be treated.
+
+Be considerate. Not everyone shares the same viewpoint and priorities as you do.
+Different opinions and interpretations help the project.
+Looking at issues from a different perspective assists development.
+
+Do not assume malice for things that can be attributed to incompetence. Even if
+it is malice, it's rarely good to start with that as initial assumption.
+
+Stay friendly even if someone acts contrarily. Everyone has a bad day
+once in a while.
+If you yourself have a bad day or are angry then try to take a break and reply
+once you are calm and without anger if you have to.
+
+Try to help other team members and cooperate if you can.
+
+The goal of software development is to create technical excellence, not for any
+individual to be better and "win" against the others. Large software projects
+are only possible and successful through teamwork.
+
+If someone struggles do not put them down. Give them a helping hand
+instead and point them in the right direction.
+
+Finally, keep in mind the immortal words of Bill and Ted,
+"Be excellent to each other."
+
@anchor{Submitting patches}
@section Submitting patches

@@ -543,6 +572,10 @@ tools/trasher, the noise bitstream filter, and
 should not crash, end in a (near) infinite loop, or allocate ridiculous
 amounts of memory when fed damaged data.

+@item
+Did you test your decoder or demuxer against sample files?
+Samples may be obtained at @url{http://samples.ffmpeg.org}.
+
@item
 Does the patch not mix functional and cosmetic changes?

@@ -633,6 +666,10 @@ not related to the comments received during review. Such patches will
 be rejected. Instead, submit significant changes or new features as
 separate patches.

+Everyone is welcome to review patches. Also if you are waiting for your patch
+to be reviewed, please consider helping to review other patches, that is a great
+way to get everyone's patches reviewed sooner.
+
@anchor{Regression tests}
@section Regression tests

@@ -494,6 +494,85 @@ Selected by Encoder (default)

@end table

+@anchor{flac}
+@section flac
+
+FLAC (Free Lossless Audio Codec) Encoder
+
+@subsection Options
+
+The following options are supported by FFmpeg's flac encoder.
+
+@table @option
+@item compression_level
+Sets the compression level, which chooses defaults for many other options
+if they are not set explicitly.
+
+@item frame_size
+Sets the size of the frames in samples per channel.
+
+@item lpc_coeff_precision
+Sets the LPC coefficient precision, valid values are from 1 to 15, 15 is the
+default.
+
+@item lpc_type
+Sets the first stage LPC algorithm
+@table @samp
+@item none
+LPC is not used
+
+@item fixed
+fixed LPC coefficients
+
+@item levinson
+
+@item cholesky
+@end table
+
+@item lpc_passes
+Number of passes to use for Cholesky factorization during LPC analysis
+
+@item min_partition_order
+The minimum partition order
+
+@item max_partition_order
+The maximum partition order
+
+@item prediction_order_method
+@table @samp
+@item estimation
+@item 2level
+@item 4level
+@item 8level
+@item search
+Bruteforce search
+@item log
+@end table
+
+@item ch_mode
+Channel mode
+@table @samp
+@item auto
+The mode is chosen automatically for each frame
+@item indep
+Chanels are independently coded
+@item left_side
+@item right_side
+@item mid_side
+@end table
+
+@item exact_rice_parameters
+Chooses if rice parameters are calculated exactly or approximately.
+if set to 1 then they are chosen exactly, which slows the code down slightly and
+improves compression slightly.
+
+@item multi_dim_quant
+Multi Dimensional Quantization. If set to 1 then a 2nd stage LPC algorithm is
+applied after the first stage to finetune the coefficients. This is quite slow
+and slightly improves compression.
+
+@end table
+
@anchor{libfaac}
@section libfaac

@@ -1263,6 +1342,30 @@ disabled
 A description of some of the currently available video encoders
 follows.

+@section jpeg2000
+
+The native jpeg 2000 encoder is lossy by default, the @code{-q:v}
+option can be used to set the encoding quality. Lossless encoding
+can be selected with @code{-pred 1}.
+
+@subsection Options
+
+@table @option
+@item format
+Can be set to either @code{j2k} or @code{jp2} (the default) that
+makes it possible to store non-rgb pix_fmts.
+
+@end table
+
+@section snow
+
+@subsection Options
+
+@table @option
+@item iterative_dia_size
+dia size for the iterative motion estimation
+@end table
+
@section libtheora

 libtheora Theora encoder wrapper.
@@ -1337,113 +1440,153 @@ You need to explicitly configure the build with @code{--enable-libvpx}.

@subsection Options

-Mapping from FFmpeg to libvpx options with conversion notes in parentheses.
+The following options are supported by the libvpx wrapper. The
+@command{vpxenc}-equivalent options or values are listed in parentheses
+for easy migration.
+
+To reduce the duplication of documentation, only the private options
+and some others requiring special attention are documented here. For
+the documentation of the undocumented generic options, see
+@ref{codec-options,,the Codec Options chapter}.
+
+To get more documentation of the libvpx options, invoke the command
+@command{ffmpeg -h encoder=libvpx}, @command{ffmpeg -h encoder=libvpx-vp9} or
+@command{vpxenc --help}. Further information is available in the libvpx API
+documentation.

@table @option

-@item threads
-g_threads
+@item b (@emph{target-bitrate})
+Set bitrate in bits/s. Note that FFmpeg's @option{b} option is
+expressed in bits/s, while @command{vpxenc}'s @option{target-bitrate} is in
+kilobits/s.

-@item profile
-g_profile
+@item g (@emph{kf-max-dist})

-@item vb
-rc_target_bitrate
+@item keyint_min (@emph{kf-min-dist})

-@item g
-kf_max_dist
+@item qmin (@emph{min-q})

-@item keyint_min
-kf_min_dist
+@item qmax (@emph{max-q})

-@item qmin
-rc_min_quantizer
+@item bufsize (@emph{buf-sz}, @emph{buf-optimal-sz})
+Set ratecontrol buffer size (in bits). Note @command{vpxenc}'s options are
+specified in milliseconds, the libvpx wrapper converts this value as follows:
+@code{buf-sz = bufsize * 1000 / bitrate},
+@code{buf-optimal-sz = bufsize * 1000 / bitrate * 5 / 6}.

-@item qmax
-rc_max_quantizer
+@item rc_init_occupancy (@emph{buf-initial-sz})
+Set number of bits which should be loaded into the rc buffer before decoding
+starts. Note @command{vpxenc}'s option is specified in milliseconds, the libvpx
+wrapper converts this value as follows:
+@code{rc_init_occupancy * 1000 / bitrate}.

-@item bufsize, vb
-rc_buf_sz
-@code{(bufsize * 1000 / vb)}
+@item undershoot-pct
+Set datarate undershoot (min) percentage of the target bitrate.

-rc_buf_optimal_sz
-@code{(bufsize * 1000 / vb * 5 / 6)}
+@item overshoot-pct
+Set datarate overshoot (max) percentage of the target bitrate.

-@item rc_init_occupancy, vb
-rc_buf_initial_sz
-@code{(rc_init_occupancy * 1000 / vb)}
+@item skip_threshold (@emph{drop-frame})

-@item rc_buffer_aggressivity
-rc_undershoot_pct
+@item qcomp (@emph{bias-pct})

-@item skip_threshold
-rc_dropframe_thresh
+@item maxrate (@emph{maxsection-pct})
+Set GOP max bitrate in bits/s. Note @command{vpxenc}'s option is specified as a
+percentage of the target bitrate, the libvpx wrapper converts this value as
+follows: @code{(maxrate * 100 / bitrate)}.

-@item qcomp
-rc_2pass_vbr_bias_pct
+@item minrate (@emph{minsection-pct})
+Set GOP min bitrate in bits/s. Note @command{vpxenc}'s option is specified as a
+percentage of the target bitrate, the libvpx wrapper converts this value as
+follows: @code{(minrate * 100 / bitrate)}.

-@item maxrate, vb
-rc_2pass_vbr_maxsection_pct
-@code{(maxrate * 100 / vb)}
+@item minrate, maxrate, b @emph{end-usage=cbr}
+@code{(minrate == maxrate == bitrate)}.

-@item minrate, vb
-rc_2pass_vbr_minsection_pct
-@code{(minrate * 100 / vb)}
+@item crf (@emph{end-usage=cq}, @emph{cq-level})

-@item minrate, maxrate, vb
-@code{VPX_CBR}
-@code{(minrate == maxrate == vb)}
-
-@item crf
-@code{VPX_CQ}, @code{VP8E_SET_CQ_LEVEL}
-
-@item quality
-@table @option
-@item @var{best}
-@code{VPX_DL_BEST_QUALITY}
-@item @var{good}
-@code{VPX_DL_GOOD_QUALITY}
-@item @var{realtime}
-@code{VPX_DL_REALTIME}
+@item quality, deadline (@emph{deadline})
+@table @samp
+@item best
+Use best quality deadline. Poorly named and quite slow, this option should be
+avoided as it may give worse quality output than good.
+@item good
+Use good quality deadline. This is a good trade-off between speed and quality
+when used with the @option{cpu-used} option.
+@item realtime
+Use realtime quality deadline.
@end table

-@item speed
-@code{VP8E_SET_CPUUSED}
+@item speed, cpu-used (@emph{cpu-used})
+Set quality/speed ratio modifier. Higher values speed up the encode at the cost
+of quality.

-@item nr
-@code{VP8E_SET_NOISE_SENSITIVITY}
+@item nr (@emph{noise-sensitivity})

-@item mb_threshold
-@code{VP8E_SET_STATIC_THRESHOLD}
+@item static-thresh
+Set a change threshold on blocks below which they will be skipped by the
+encoder.

-@item slices
-@code{VP8E_SET_TOKEN_PARTITIONS}
+@item slices (@emph{token-parts})
+Note that FFmpeg's @option{slices} option gives the total number of partitions,
+while @command{vpxenc}'s @option{token-parts} is given as
+@code{log2(partitions)}.

@item max-intra-rate
-@code{VP8E_SET_MAX_INTRA_BITRATE_PCT}
+Set maximum I-frame bitrate as a percentage of the target bitrate. A value of 0
+means unlimited.

@item force_key_frames
@code{VPX_EFLAG_FORCE_KF}

@item Alternate reference frame related
@table @option
-@item vp8flags altref
-@code{VP8E_SET_ENABLEAUTOALTREF}
-@item @var{arnr_max_frames}
-@code{VP8E_SET_ARNR_MAXFRAMES}
-@item @var{arnr_type}
-@code{VP8E_SET_ARNR_TYPE}
-@item @var{arnr_strength}
-@code{VP8E_SET_ARNR_STRENGTH}
-@item @var{rc_lookahead}
-g_lag_in_frames
+@item auto-alt-ref
+Enable use of alternate reference frames (2-pass only).
+@item arnr-max-frames
+Set altref noise reduction max frame count.
+@item arnr-type
+Set altref noise reduction filter type: backward, forward, centered.
+@item arnr-strength
+Set altref noise reduction filter strength.
+@item rc-lookahead, lag-in-frames (@emph{lag-in-frames})
+Set number of frames to look ahead for frametype and ratecontrol.
@end table

-@item vp8flags error_resilient
-g_error_resilient
+@item error-resilient
+Enable error resiliency features.

-@item aq_mode
-@code{VP9E_SET_AQ_MODE}
+@item VP9-specific options
+@table @option
+@item lossless
+Enable lossless mode.
+@item tile-columns
+Set number of tile columns to use. Note this is given as
+@code{log2(tile_columns)}. For example, 8 tile columns would be requested by
+setting the @option{tile-columns} option to 3.
+@item tile-rows
+Set number of tile rows to use. Note this is given as @code{log2(tile_rows)}.
+For example, 4 tile rows would be requested by setting the @option{tile-rows}
+option to 2.
+@item frame-parallel
+Enable frame parallel decodability features.
+@item aq-mode
+Set adaptive quantization mode (0: off (default), 1: variance 2: complexity, 3:
+cyclic refresh).
+@item colorspace @emph{color-space}
+Set input color space. The VP9 bitstream supports signaling the following
+colorspaces:
+@table @option
+@item @samp{rgb} @emph{sRGB}
+@item @samp{bt709} @emph{bt709}
+@item @samp{unspecified} @emph{unknown}
+@item @samp{bt470bg} @emph{bt601}
+@item @samp{smpte170m} @emph{smpte170}
+@item @samp{smpte240m} @emph{smpte240}
+@item @samp{bt2020_ncl} @emph{bt2020}
+@end table
+@end table

@end table

@@ -2181,6 +2324,30 @@ Setting a higher @option{bits_per_mb} limit will improve the speed.
 For the fastest encoding speed set the @option{qscale} parameter (4 is the
 recommended value) and do not set a size constraint.

+@section libkvazaar
+
+Kvazaar H.265/HEVC encoder.
+
+Requires the presence of the libkvazaar headers and library during
+configuration. You need to explicitly configure the build with
+@option{--enable-libkvazaar}.
+
+@subsection Options
+
+@table @option
+
+@item b
+Set target video bitrate in bit/s and enable rate control.
+
+@item threads
+Set number of encoding threads.
+
+@item kvazaar-params
+Set kvazaar parameters as a list of @var{name}=@var{value} pairs separated
+by commas (,). See kvazaar documentation for a list of options.
+
+@end table
+
@c man end VIDEO ENCODERS

@chapter Subtitles Encoders
@@ -11,12 +11,14 @@ CFLAGS += -Wall -g
 CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
 LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)

-EXAMPLES=       avio_reading                       \
+EXAMPLES=       avio_dir_cmd                       \
+                avio_reading                       \
                decoding_encoding                  \
                demuxing_decoding                  \
                extract_mvs                        \
                filtering_video                    \
                filtering_audio                    \
+                http_multiclient                   \
                metadata                           \
                muxing                             \
                remuxing                           \
@@ -0,0 +1,180 @@
+/*
+ * Copyright (c) 2014 Lukasz Marek
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavformat/avio.h>
+
+static const char *type_string(int type)
+{
+    switch (type) {
+    case AVIO_ENTRY_DIRECTORY:
+        return "<DIR>";
+    case AVIO_ENTRY_FILE:
+        return "<FILE>";
+    case AVIO_ENTRY_BLOCK_DEVICE:
+        return "<BLOCK DEVICE>";
+    case AVIO_ENTRY_CHARACTER_DEVICE:
+        return "<CHARACTER DEVICE>";
+    case AVIO_ENTRY_NAMED_PIPE:
+        return "<PIPE>";
+    case AVIO_ENTRY_SYMBOLIC_LINK:
+        return "<LINK>";
+    case AVIO_ENTRY_SOCKET:
+        return "<SOCKET>";
+    case AVIO_ENTRY_SERVER:
+        return "<SERVER>";
+    case AVIO_ENTRY_SHARE:
+        return "<SHARE>";
+    case AVIO_ENTRY_WORKGROUP:
+        return "<WORKGROUP>";
+    case AVIO_ENTRY_UNKNOWN:
+    default:
+        break;
+    }
+    return "<UNKNOWN>";
+}
+
+static int list_op(const char *input_dir)
+{
+    AVIODirEntry *entry = NULL;
+    AVIODirContext *ctx = NULL;
+    int cnt, ret;
+    char filemode[4], uid_and_gid[20];
+
+    if ((ret = avio_open_dir(&ctx, input_dir, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot open directory: %s.\n", av_err2str(ret));
+        goto fail;
+    }
+
+    cnt = 0;
+    for (;;) {
+        if ((ret = avio_read_dir(ctx, &entry)) < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Cannot list directory: %s.\n", av_err2str(ret));
+            goto fail;
+        }
+        if (!entry)
+            break;
+        if (entry->filemode == -1) {
+            snprintf(filemode, 4, "???");
+        } else {
+            snprintf(filemode, 4, "%3"PRIo64, entry->filemode);
+        }
+        snprintf(uid_and_gid, 20, "%"PRId64"(%"PRId64")", entry->user_id, entry->group_id);
+        if (cnt == 0)
+            av_log(NULL, AV_LOG_INFO, "%-9s %12s %30s %10s %s %16s %16s %16s\n",
+                   "TYPE", "SIZE", "NAME", "UID(GID)", "UGO", "MODIFIED",
+                   "ACCESSED", "STATUS_CHANGED");
+        av_log(NULL, AV_LOG_INFO, "%-9s %12"PRId64" %30s %10s %s %16"PRId64" %16"PRId64" %16"PRId64"\n",
+               type_string(entry->type),
+               entry->size,
+               entry->name,
+               uid_and_gid,
+               filemode,
+               entry->modification_timestamp,
+               entry->access_timestamp,
+               entry->status_change_timestamp);
+        avio_free_directory_entry(&entry);
+        cnt++;
+    };
+
+  fail:
+    avio_close_dir(&ctx);
+    return ret;
+}
+
+static int del_op(const char *url)
+{
+    int ret = avpriv_io_delete(url);
+    if (ret < 0)
+        av_log(NULL, AV_LOG_ERROR, "Cannot delete '%s': %s.\n", url, av_err2str(ret));
+    return ret;
+}
+
+static int move_op(const char *src, const char *dst)
+{
+    int ret = avpriv_io_move(src, dst);
+    if (ret < 0)
+        av_log(NULL, AV_LOG_ERROR, "Cannot move '%s' into '%s': %s.\n", src, dst, av_err2str(ret));
+    return ret;
+}
+
+
+static void usage(const char *program_name)
+{
+    fprintf(stderr, "usage: %s OPERATION entry1 [entry2]\n"
+            "API example program to show how to manipulate resources "
+            "accessed through AVIOContext.\n"
+            "OPERATIONS:\n"
+            "list      list content of the directory\n"
+            "move      rename content in directory\n"
+            "del       delete content in directory\n",
+            program_name);
+}
+
+int main(int argc, char *argv[])
+{
+    const char *op = NULL;
+    int ret;
+
+    av_log_set_level(AV_LOG_DEBUG);
+
+    if (argc < 2) {
+        usage(argv[0]);
+        return 1;
+    }
+
+    /* register codecs and formats and other lavf/lavc components*/
+    av_register_all();
+    avformat_network_init();
+
+    op = argv[1];
+    if (strcmp(op, "list") == 0) {
+        if (argc < 3) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for list operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = list_op(argv[2]);
+        }
+    } else if (strcmp(op, "del") == 0) {
+        if (argc < 3) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for del operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = del_op(argv[2]);
+        }
+    } else if (strcmp(op, "move") == 0) {
+        if (argc < 4) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for move operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = move_op(argv[2], argv[3]);
+        }
+    } else {
+        av_log(NULL, AV_LOG_INFO, "Invalid operation %s\n", op);
+        ret = AVERROR(EINVAL);
+    }
+
+    avformat_network_deinit();
+
+    return ret < 0 ? 1 : 0;
+}
@@ -245,7 +245,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
    AVCodecContext *c= NULL;
    int len;
    FILE *f, *outfile;
-    uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
+    uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
    AVPacket avpkt;
    AVFrame *decoded_frame = NULL;

@@ -521,7 +521,7 @@ static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
        /* the picture is allocated by the decoder, no need to free it */
        snprintf(buf, sizeof(buf), outfilename, *frame_count);
        pgm_save(frame->data[0], frame->linesize[0],
-                 avctx->width, avctx->height, buf);
+                 frame->width, frame->height, buf);
        (*frame_count)++;
    }
    if (pkt->data) {
@@ -538,13 +538,13 @@ static void video_decode_example(const char *outfilename, const char *filename)
    int frame_count;
    FILE *f;
    AVFrame *frame;
-    uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
+    uint8_t inbuf[INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
    AVPacket avpkt;

    av_init_packet(&avpkt);

    /* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
-    memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
+    memset(inbuf + INBUF_SIZE, 0, AV_INPUT_BUFFER_PADDING_SIZE);

    printf("Decode video file %s to %s\n", filename, outfilename);

@@ -561,8 +561,8 @@ static void video_decode_example(const char *outfilename, const char *filename)
        exit(1);
    }

-    if(codec->capabilities&CODEC_CAP_TRUNCATED)
-        c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
+    if (codec->capabilities & AV_CODEC_CAP_TRUNCATED)
+        c->flags |= AV_CODEC_FLAG_TRUNCATED; // we do not send complete frames

    /* For some codecs, such as msmpeg4 and mpeg4, width and height
       MUST be initialized there because this information is not
@@ -81,22 +81,24 @@ static int decode_packet(int *got_frame, int cached)
            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
            return ret;
        }
-        if (video_dec_ctx->width != width || video_dec_ctx->height != height ||
-            video_dec_ctx->pix_fmt != pix_fmt) {
-            /* To handle this change, one could call av_image_alloc again and
-             * decode the following frames into another rawvideo file. */
-            fprintf(stderr, "Error: Width, height and pixel format have to be "
-                    "constant in a rawvideo file, but the width, height or "
-                    "pixel format of the input video changed:\n"
-                    "old: width = %d, height = %d, format = %s\n"
-                    "new: width = %d, height = %d, format = %s\n",
-                    width, height, av_get_pix_fmt_name(pix_fmt),
-                    video_dec_ctx->width, video_dec_ctx->height,
-                    av_get_pix_fmt_name(video_dec_ctx->pix_fmt));
-            return -1;
-        }

        if (*got_frame) {
+
+            if (frame->width != width || frame->height != height ||
+                frame->format != pix_fmt) {
+                /* To handle this change, one could call av_image_alloc again and
+                 * decode the following frames into another rawvideo file. */
+                fprintf(stderr, "Error: Width, height and pixel format have to be "
+                        "constant in a rawvideo file, but the width, height or "
+                        "pixel format of the input video changed:\n"
+                        "old: width = %d, height = %d, format = %s\n"
+                        "new: width = %d, height = %d, format = %s\n",
+                        width, height, av_get_pix_fmt_name(pix_fmt),
+                        frame->width, frame->height,
+                        av_get_pix_fmt_name(frame->format));
+                return -1;
+            }
+
            printf("video_frame%s n:%d coded_n:%d pts:%s\n",
                   cached ? "(cached)" : "",
                   video_frame_count++, frame->coded_picture_number,
@@ -38,7 +38,10 @@
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>

-const char *filter_descr = "scale=78:24";
+const char *filter_descr = "scale=78:24,transpose=cclock";
+/* other way:
+   scale=78:24 [scl]; [scl] transpose=cclock // assumes "[in]" and "[out]" to be input output pads respectively
+ */

 static AVFormatContext *fmt_ctx;
 static AVCodecContext *dec_ctx;
@@ -0,0 +1,155 @@
+/*
+ * Copyright (c) 2015 Stephan Holljes
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libavformat multi-client network API usage example.
+ *
+ * @example http_multiclient.c
+ * This example will serve a file without decoding or demuxing it over http.
+ * Multiple clients can connect and will receive the same file.
+ */
+
+#include <libavformat/avformat.h>
+#include <libavutil/opt.h>
+#include <unistd.h>
+
+void process_client(AVIOContext *client, const char *in_uri)
+{
+    AVIOContext *input = NULL;
+    uint8_t buf[1024];
+    int ret, n, reply_code;
+    char *resource = NULL;
+    while ((ret = avio_handshake(client)) > 0) {
+        av_opt_get(client, "resource", AV_OPT_SEARCH_CHILDREN, &resource);
+        // check for strlen(resource) is necessary, because av_opt_get()
+        // may return empty string.
+        if (resource && strlen(resource))
+            break;
+    }
+    if (ret < 0)
+        goto end;
+    av_log(client, AV_LOG_TRACE, "resource=%p\n", resource);
+    if (resource && resource[0] == '/' && !strcmp((resource + 1), in_uri)) {
+        reply_code = 200;
+    } else {
+        reply_code = AVERROR_HTTP_NOT_FOUND;
+    }
+    if ((ret = av_opt_set_int(client, "reply_code", reply_code, AV_OPT_SEARCH_CHILDREN)) < 0) {
+        av_log(client, AV_LOG_ERROR, "Failed to set reply_code: %s.\n", av_err2str(ret));
+        goto end;
+    }
+    av_log(client, AV_LOG_TRACE, "Set reply code to %d\n", reply_code);
+
+    while ((ret = avio_handshake(client)) > 0);
+
+    if (ret < 0)
+        goto end;
+
+    fprintf(stderr, "Handshake performed.\n");
+    if (reply_code != 200)
+        goto end;
+    fprintf(stderr, "Opening input file.\n");
+    if ((ret = avio_open2(&input, in_uri, AVIO_FLAG_READ, NULL, NULL)) < 0) {
+        av_log(input, AV_LOG_ERROR, "Failed to open input: %s: %s.\n", in_uri,
+               av_err2str(ret));
+        goto end;
+    }
+    for(;;) {
+        n = avio_read(input, buf, sizeof(buf));
+        if (n < 0) {
+            if (n == AVERROR_EOF)
+                break;
+            av_log(input, AV_LOG_ERROR, "Error reading from input: %s.\n",
+                   av_err2str(n));
+            break;
+        }
+        avio_write(client, buf, n);
+        avio_flush(client);
+    }
+end:
+    fprintf(stderr, "Flushing client\n");
+    avio_flush(client);
+    fprintf(stderr, "Closing client\n");
+    avio_close(client);
+    fprintf(stderr, "Closing input\n");
+    avio_close(input);
+}
+
+int main(int argc, char **argv)
+{
+    av_log_set_level(AV_LOG_TRACE);
+    AVDictionary *options = NULL;
+    AVIOContext *client = NULL, *server = NULL;
+    const char *in_uri, *out_uri;
+    int ret, pid;
+    if (argc < 3) {
+        printf("usage: %s input http://hostname[:port]\n"
+               "API example program to serve http to multiple clients.\n"
+               "\n", argv[0]);
+        return 1;
+    }
+
+    in_uri = argv[1];
+    out_uri = argv[2];
+
+    av_register_all();
+    avformat_network_init();
+
+    if ((ret = av_dict_set(&options, "listen", "2", 0)) < 0) {
+        fprintf(stderr, "Failed to set listen mode for server: %s\n", av_err2str(ret));
+        return ret;
+    }
+    if ((ret = avio_open2(&server, out_uri, AVIO_FLAG_WRITE, NULL, &options)) < 0) {
+        fprintf(stderr, "Failed to open server: %s\n", av_err2str(ret));
+        return ret;
+    }
+    fprintf(stderr, "Entering main loop.\n");
+    for(;;) {
+        if ((ret = avio_accept(server, &client)) < 0)
+            goto end;
+        fprintf(stderr, "Accepted client, forking process.\n");
+        // XXX: Since we don't reap our children and don't ignore signals
+        //      this produces zombie processes.
+        pid = fork();
+        if (pid < 0) {
+            perror("Fork failed");
+            ret = AVERROR(errno);
+            goto end;
+        }
+        if (pid == 0) {
+            fprintf(stderr, "In child.\n");
+            process_client(client, in_uri);
+            avio_close(server);
+            exit(0);
+        }
+        if (pid > 0)
+            avio_close(client);
+    }
+end:
+    avio_close(server);
+    if (ret < 0 && ret != AVERROR_EOF) {
+        fprintf(stderr, "Some errors occurred: %s\n", av_err2str(ret));
+        return 1;
+    }
+    return 0;
+}
@@ -172,7 +172,7 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,

    /* Some formats want stream headers to be separate. */
    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
-        c->flags |= CODEC_FLAG_GLOBAL_HEADER;
+        c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
 }

 /**************************************************************/
@@ -230,7 +230,7 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
    /* increment frequency by 110 Hz per second */
    ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;

-    if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
+    if (c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
        nb_samples = 10000;
    else
        nb_samples = c->frame_size;
@@ -405,7 +405,7 @@ int main(int argc, char **argv)
    decoder_ctx->codec_id = AV_CODEC_ID_H264;
    if (video_st->codec->extradata_size) {
        decoder_ctx->extradata = av_mallocz(video_st->codec->extradata_size +
-                                            FF_INPUT_BUFFER_PADDING_SIZE);
+                                            AV_INPUT_BUFFER_PADDING_SIZE);
        if (!decoder_ctx->extradata) {
            ret = AVERROR(ENOMEM);
            goto finish;
@@ -101,7 +101,7 @@ int main(int argc, char **argv)
        }
        out_stream->codec->codec_tag = 0;
        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-            out_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
+            out_stream->codec->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
    }
    av_dump_format(ofmt_ctx, 0, out_filename, 1);

@@ -192,7 +192,7 @@ static int open_output_file(const char *filename,
     * Mark the encoder so that it behaves accordingly.
     */
    if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
-        (*output_codec_context)->flags |= CODEC_FLAG_GLOBAL_HEADER;
+        (*output_codec_context)->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;

    /** Open the encoder for the audio stream to use it later. */
    if ((error = avcodec_open2(*output_codec_context, output_codec, NULL)) < 0) {
@@ -117,7 +117,7 @@ static int open_output_file(const char *filename)
            /* in this example, we choose transcoding to same codec */
            encoder = avcodec_find_encoder(dec_ctx->codec_id);
            if (!encoder) {
-                av_log(NULL, AV_LOG_FATAL, "Neccessary encoder not found\n");
+                av_log(NULL, AV_LOG_FATAL, "Necessary encoder not found\n");
                return AVERROR_INVALIDDATA;
            }

@@ -161,7 +161,7 @@ static int open_output_file(const char *filename)
        }

        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-            enc_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
+            enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;

    }
    av_dump_format(ofmt_ctx, 0, filename, 1);
@@ -449,7 +449,7 @@ static int flush_encoder(unsigned int stream_index)
    int got_frame;

    if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
-                CODEC_CAP_DELAY))
+                AV_CODEC_CAP_DELAY))
        return 0;

    while (1) {
@@ -349,7 +349,7 @@ FFmpeg has a @url{http://ffmpeg.org/ffmpeg-protocols.html#concat,
@code{concat}} protocol designed specifically for that, with examples in the
 documentation.

-A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate
+A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow one to concatenate
 video by merely concatenating the files containing them.

 Hence you may concatenate your multimedia files by first transcoding them to
@@ -13,36 +13,36 @@

@chapter Introduction

-  FATE is an extended regression suite on the client-side and a means
+FATE is an extended regression suite on the client-side and a means
 for results aggregation and presentation on the server-side.

-  The first part of this document explains how you can use FATE from
+The first part of this document explains how you can use FATE from
 your FFmpeg source directory to test your ffmpeg binary. The second
 part describes how you can run FATE to submit the results to FFmpeg's
 FATE server.

-  In any way you can have a look at the publicly viewable FATE results
+In any way you can have a look at the publicly viewable FATE results
 by visiting this website:

-  @url{http://fate.ffmpeg.org/}
+@url{http://fate.ffmpeg.org/}

-  This is especially recommended for all people contributing source
+This is especially recommended for all people contributing source
 code to FFmpeg, as it can be seen if some test on some platform broke
 with their recent contribution. This usually happens on the platforms
 the developers could not test on.

-  The second part of this document describes how you can run FATE to
+The second part of this document describes how you can run FATE to
 submit your results to FFmpeg's FATE server. If you want to submit your
 results be sure to check that your combination of CPU, OS and compiler
 is not already listed on the above mentioned website.

-  In the third part you can find a comprehensive listing of FATE makefile
+In the third part you can find a comprehensive listing of FATE makefile
 targets and variables.


@chapter Using FATE from your FFmpeg source directory

-  If you want to run FATE on your machine you need to have the samples
+If you want to run FATE on your machine you need to have the samples
 in place. You can get the samples via the build target fate-rsync.
 Use this command from the top-level source directory:

@@ -51,11 +51,11 @@ make fate-rsync SAMPLES=fate-suite/
 make fate       SAMPLES=fate-suite/
@end example

-  The above commands set the samples location by passing a makefile
+The above commands set the samples location by passing a makefile
 variable via command line. It is also possible to set the samples
 location at source configuration time by invoking configure with
-`--samples=<path to the samples directory>'. Afterwards you can
-invoke the makefile targets without setting the SAMPLES makefile
+@option{--samples=<path to the samples directory>}. Afterwards you can
+invoke the makefile targets without setting the @var{SAMPLES} makefile
 variable. This is illustrated by the following commands:

@example
@@ -64,7 +64,7 @@ make fate-rsync
 make fate
@end example

-  Yet another way to tell FATE about the location of the sample
+Yet another way to tell FATE about the location of the sample
 directory is by making sure the environment variable FATE_SAMPLES
 contains the path to your samples directory. This can be achieved
 by e.g. putting that variable in your shell profile or by setting
@@ -85,7 +85,7 @@ To use a custom wrapper to run the test, pass @option{--target-exec} to

@chapter Submitting the results to the FFmpeg result aggregation server

-  To submit your results to the server you should run fate through the
+To submit your results to the server you should run fate through the
 shell script @file{tests/fate.sh} from the FFmpeg sources. This script needs
 to be invoked with a configuration file as its first argument.

@@ -93,23 +93,23 @@ to be invoked with a configuration file as its first argument.
 tests/fate.sh /path/to/fate_config
@end example

-  A configuration file template with comments describing the individual
+A configuration file template with comments describing the individual
 configuration variables can be found at @file{doc/fate_config.sh.template}.

@ifhtml
-  The mentioned configuration template is also available here:
+The mentioned configuration template is also available here:
@verbatiminclude fate_config.sh.template
@end ifhtml

-  Create a configuration that suits your needs, based on the configuration
-template. The `slot' configuration variable can be any string that is not
+Create a configuration that suits your needs, based on the configuration
+template. The @env{slot} configuration variable can be any string that is not
 yet used, but it is suggested that you name it adhering to the following
-pattern <arch>-<os>-<compiler>-<compiler version>. The configuration file
-itself will be sourced in a shell script, therefore all shell features may
-be used. This enables you to setup the environment as you need it for your
-build.
+pattern @samp{@var{arch}-@var{os}-@var{compiler}-@var{compiler version}}. The
+configuration file itself will be sourced in a shell script, therefore all
+shell features may be used. This enables you to setup the environment as you
+need it for your build.

-  For your first test runs the `fate_recv' variable should be empty or
+For your first test runs the @env{fate_recv} variable should be empty or
 commented out. This will run everything as normal except that it will omit
 the submission of the results to the server. The following files should be
 present in $workdir as specified in the configuration file:
@@ -122,29 +122,29 @@ present in $workdir as specified in the configuration file:
    @item version
@end itemize

-  When you have everything working properly you can create an SSH key pair
+When you have everything working properly you can create an SSH key pair
 and send the public key to the FATE server administrator who can be contacted
 at the email address @email{fate-admin@@ffmpeg.org}.

-  Configure your SSH client to use public key authentication with that key
+Configure your SSH client to use public key authentication with that key
 when connecting to the FATE server. Also do not forget to check the identity
 of the server and to accept its host key. This can usually be achieved by
 running your SSH client manually and killing it after you accepted the key.
 The FATE server's fingerprint is:

-@table @option
+@table @samp
@item RSA
   d3:f1:83:97:a4:75:2b:a6:fb:d6:e8:aa:81:93:97:51
@item ECDSA
   76:9f:68:32:04:1e:d5:d4:ec:47:3f:dc:fc:18:17:86
@end table

-  If you have problems connecting to the FATE server, it may help to try out
+If you have problems connecting to the FATE server, it may help to try out
 the @command{ssh} command with one or more @option{-v} options. You should
 get detailed output concerning your SSH configuration and the authentication
 process.

-  The only thing left is to automate the execution of the fate.sh script and
+The only thing left is to automate the execution of the fate.sh script and
 the synchronisation of the samples directory.


@@ -165,7 +165,7 @@ Run the FATE test suite (requires the fate-suite dataset).

@section Makefile variables

-@table @option
+@table @env
@item V
 Verbosity level, can be set to 0, 1 or 2.
    @itemize
@@ -183,20 +183,20 @@ Specify how many threads to use while running regression tests, it is
 quite useful to detect thread-related regressions.

@item THREAD_TYPE
-Specify which threading strategy test, either @var{slice} or @var{frame},
-by default @var{slice+frame}
+Specify which threading strategy test, either @samp{slice} or @samp{frame},
+by default @samp{slice+frame}

@item CPUFLAGS
 Specify CPU flags.

@item TARGET_EXEC
 Specify or override the wrapper used to run the tests.
-The @var{TARGET_EXEC} option provides a way to run FATE wrapped in
+The @env{TARGET_EXEC} option provides a way to run FATE wrapped in
@command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
 through @command{ssh}.

@item GEN
-Set to @var{1} to generate the missing or mismatched references.
+Set to @samp{1} to generate the missing or mismatched references.
@end table

@section Examples
@@ -1,5 +1,6 @@
 slot=                                    # some unique identifier
 repo=git://source.ffmpeg.org/ffmpeg.git  # the source repository
+#branch=release/2.6                       # the branch to test
 samples=                                 # path to samples directory
 workdir=                                 # directory in which to do all the work
 #fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report
@@ -12,7 +12,7 @@

@chapter Synopsis

-ffmpeg [@var{global_options}] @{[@var{input_file_options}] -i @file{input_file}@} ... @{[@var{output_file_options}] @file{output_file}@} ...
+ffmpeg [@var{global_options}] @{[@var{input_file_options}] -i @file{input_url}@} ... @{[@var{output_file_options}] @file{output_url}@} ...

@chapter Description
@c man begin DESCRIPTION
@@ -24,10 +24,10 @@ rates and resize video on the fly with a high quality polyphase filter.
@command{ffmpeg} reads from an arbitrary number of input "files" (which can be regular
 files, pipes, network streams, grabbing devices, etc.), specified by the
@code{-i} option, and writes to an arbitrary number of output "files", which are
-specified by a plain output filename. Anything found on the command line which
-cannot be interpreted as an option is considered to be an output filename.
+specified by a plain output url. Anything found on the command line which
+cannot be interpreted as an option is considered to be an output url.

-Each input or output file can, in principle, contain any number of streams of
+Each input or output url can, in principle, contain any number of streams of
 different types (video/audio/subtitle/attachment/data). The allowed number and/or
 types of streams may be limited by the container format. Selecting which
 streams from which inputs will go into which output is either done automatically
@@ -80,7 +80,7 @@ The format option may be needed for raw input files.
 The transcoding process in @command{ffmpeg} for each output can be described by
 the following diagram:

-@example
+@verbatim
 _______              ______________
 |       |            |              |
 | input |  demuxer   | encoded data |   decoder
@@ -99,7 +99,7 @@ the following diagram:
 |________|           |______________|


-@end example
+@end verbatim

@command{ffmpeg} calls the libavformat library (containing demuxers) to read
 input files and get packets containing encoded data from them. When there are
@@ -124,7 +124,7 @@ Simple filtergraphs are those that have exactly one input and output, both of
 the same type. In the above diagram they can be represented by simply inserting
 an additional step between decoding and encoding:

-@example
+@verbatim
 _________                        ______________
 |         |                      |              |
 | decoded |                      | encoded data |
@@ -136,19 +136,19 @@ an additional step between decoding and encoding:
                | frames   |
                |__________|

-@end example
+@end verbatim

 Simple filtergraphs are configured with the per-stream @option{-filter} option
 (with @option{-vf} and @option{-af} aliases for video and audio respectively).
 A simple filtergraph for video can look for example like this:

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

-@end example
+@end verbatim

 Note that some filters change frame properties but not frame contents. E.g. the
@code{fps} filter in the example above changes number of frames, but does not
@@ -161,7 +161,7 @@ processing chain applied to one stream. This is the case, for example, when the
 more than one input and/or output, or when output stream type is different from
 input. They can be represented with the following diagram:

-@example
+@verbatim
 _________
 |         |
 | input 0 |\                    __________
@@ -179,7 +179,7 @@ input. They can be represented with the following diagram:
 | input 2 |/
 |_________|

-@end example
+@end verbatim

 Complex filtergraphs are configured with the @option{-filter_complex} option.
 Note that this option is global, since a complex filtergraph, by its nature,
@@ -198,14 +198,14 @@ step for the specified stream, so it does only demuxing and muxing. It is useful
 for changing the container format or modifying container-level metadata. The
 diagram above will, in this case, simplify to this:

-@example
+@verbatim
 _______              ______________            ________
 |       |            |              |          |        |
 | input |  demuxer   | encoded data |  muxer   | output |
 | file  | ---------> | packets      | -------> | file   |
 |_______|            |______________|          |________|

-@end example
+@end verbatim

 Since there is no decoding or encoding, it is very fast and there is no quality
 loss. However, it might not work in some cases because of many factors. Applying
@@ -243,8 +243,8 @@ Force input or output file format. The format is normally auto detected for inpu
 files and guessed from the file extension for output files, so this option is not
 needed in most cases.

-@item -i @var{filename} (@emph{input})
-input file name
+@item -i @var{url} (@emph{input})
+input file url

@item -y (@emph{global})
 Overwrite output files without asking.
@@ -277,16 +277,18 @@ libx264, and the 138th audio, which will be encoded with libvorbis.
 When used as an input option (before @code{-i}), limit the @var{duration} of
 data read from the input file.

-When used as an output option (before an output filename), stop writing the
+When used as an output option (before an output url), stop writing the
 output after its duration reaches @var{duration}.

-@var{duration} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.
+@var{duration} must be a time duration specification,
+see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.

 -to and -t are mutually exclusive and -t has priority.

@item -to @var{position} (@emph{output})
 Stop writing the output at @var{position}.
-@var{position} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.
+@var{position} must be a time duration specification,
+see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.

 -to and -t are mutually exclusive and -t has priority.

@@ -295,17 +297,23 @@ Set the file size limit, expressed in bytes.

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

-When used as an output option (before an output filename), decodes but discards
+When used as an output option (before an output url), decodes but discards
 input until the timestamps reach @var{position}.

-@var{position} may be either in seconds or in @code{hh:mm:ss[.xxx]} form.
+@var{position} must be a time duration specification,
+see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+
+@item -sseof @var{position} (@emph{input/output})
+
+Like the @code{-ss} option but relative to the "end of file". That is negative
+values are earlier in the file, 0 is at EOF.

@item -itsoffset @var{offset} (@emph{input})
 Set the input time offset.
@@ -320,7 +328,7 @@ the time duration specified in @var{offset}.
@item -timestamp @var{date} (@emph{output})
 Set the recording timestamp in the container.

-@var{date} must be a time duration specification,
+@var{date} must be a date specification,
 see @ref{date syntax,,the Date section in the ffmpeg-utils(1) manual,ffmpeg-utils}.

@item -metadata[:metadata_specifier] @var{key}=@var{value} (@emph{output,per-metadata})
@@ -462,6 +470,9 @@ Technical note -- attachments are implemented as codec extradata, so this
 option can actually be used to extract extradata from any stream, not just
 attachments.

+@item -noautorotate
+Disable automatically rotating video based on file metadata.
+
@end table

@section Video Options
@@ -687,6 +698,10 @@ is not specified, the value of the @var{DISPLAY} environment variable is used
 For DXVA2, this option should contain the number of the display adapter to use.
 If this option is not specified, the default adapter is used.
@end table
+
+@item -hwaccels
+List all hardware acceleration methods supported in this build of ffmpeg.
+
@end table

@section Audio Options
@@ -832,6 +847,14 @@ ffmpeg -i INPUT -map 0:m:language:eng OUTPUT

 Note that using this option disables the default mappings for this output file.

+@item -ignore_unknown
+Ignore input streams with unknown type instead of failing if copying
+such streams is attempted.
+
+@item -copy_unknown
+Allow input streams with unknown type to be copied instead of failing if copying
+such streams is attempted.
+
@item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][:@var{output_file_id}.@var{stream_specifier}]
 Map an audio channel from a given input to an output. If
@var{output_file_id}.@var{stream_specifier} is not set, the audio channel will
@@ -1071,7 +1094,7 @@ may be reassigned to a different value.
 For example, to set the stream 0 PID to 33 and the stream 1 PID to 36 for
 an output mpegts file:
@example
-ffmpeg -i infile -streamid 0:33 -streamid 1:36 out.ts
+ffmpeg -i inurl -streamid 0:33 -streamid 1:36 out.ts
@end example

@item -bsf[:@var{stream_specifier}] @var{bitstream_filters} (@emph{output,per-stream})
@@ -1158,7 +1181,14 @@ This option enables or disables accurate seeking in input files with the
 transcoding. Use @option{-noaccurate_seek} to disable it, which may be useful
 e.g. when copying some streams and transcoding the others.

-@item -thread_message_queue @var{size} (@emph{input})
+@item -seek_timestamp (@emph{input})
+This option enables or disables seeking by timestamp in input files with the
+@option{-ss} option. It is disabled by default. If enabled, the argument
+to the @option{-ss} option is considered an actual timestamp, and is not
+offset by the start time of the file. This matters only for files which do
+not start from timestamp 0, such as transport streams.
+
+@item -thread_queue_size @var{size} (@emph{input})
 This option sets the maximum number of queued packets when reading from the
 file or device. With low latency / high rate live streams, packets may be
 discarded if they are not read in a timely manner; raising this value can
@@ -1175,9 +1205,9 @@ The option is intended for cases where features are needed that cannot be
 specified to @command{ffserver} but can be to @command{ffmpeg}.

@item -sdp_file @var{file} (@emph{global})
-Print sdp information to @var{file}.
+Print sdp information for an output stream to @var{file}.
 This allows dumping sdp information when at least one output isn't an
-rtp stream.
+rtp stream. (Requires at least one of the output formats to be rtp).

@item -discard (@emph{input})
 Allows discarding specific streams or frames of streams at the demuxer.
@@ -1203,6 +1233,9 @@ Discard all frames excepts keyframes.
 Discard all frames.
@end table

+@item -xerror (@emph{global})
+Stop and exit on error
+
@end table

 As a special exception, you can use a bitmap subtitle stream as input: it
@@ -1278,47 +1311,6 @@ If no such file is found, then ffmpeg will search for a file named

@c man end OPTIONS

-@chapter Tips
-@c man begin TIPS
-
-@itemize
-@item
-For streaming at very low bitrates, use a low frame rate
-and a small GOP size. This is especially true for RealVideo where
-the Linux player does not seem to be very fast, so it can miss
-frames. An example is:
-
-@example
-ffmpeg -g 3 -r 3 -t 10 -b:v 50k -s qcif -f rv10 /tmp/b.rm
-@end example
-
-@item
-The parameter 'q' which is displayed while encoding is the current
-quantizer. The value 1 indicates that a very good quality could
-be achieved. The value 31 indicates the worst quality. If q=31 appears
-too often, it means that the encoder cannot compress enough to meet
-your bitrate. You must either increase the bitrate, decrease the
-frame rate or decrease the frame size.
-
-@item
-If your computer is not fast enough, you can speed up the
-compression at the expense of the compression ratio. You can use
-'-me zero' to speed up motion estimation, and '-g 0' to disable
-motion estimation completely (you have only I-frames, which means it
-is about as good as JPEG compression).
-
-@item
-To have very low audio bitrates, reduce the sampling frequency
-(down to 22050 Hz for MPEG audio, 22050 or 11025 for AC-3).
-
-@item
-To have a constant quality (but a variable bitrate), use the option
-'-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst
-quality).
-
-@end itemize
-@c man end TIPS
-
@chapter Examples
@c man begin EXAMPLES

@@ -1469,7 +1461,7 @@ combination with -ss to start extracting from a certain point in time.

 For creating a video from many images:
@example
-ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi
+ffmpeg -f image2 -framerate 12 -i foo-%03d.jpeg -s WxH foo.avi
@end example

 The syntax @code{foo-%03d.jpeg} specifies to use a decimal number
@@ -1484,7 +1476,7 @@ image2-specific @code{-pattern_type glob} option.
 For example, for creating a video from filenames matching the glob pattern
@code{foo-*.jpeg}:
@example
-ffmpeg -f image2 -pattern_type glob -i 'foo-*.jpeg' -r 12 -s WxH foo.avi
+ffmpeg -f image2 -pattern_type glob -framerate 12 -i 'foo-*.jpeg' -s WxH foo.avi
@end example

@item
@@ -12,7 +12,7 @@

@chapter Synopsis

-ffplay [@var{options}] [@file{input_file}]
+ffplay [@var{options}] [@file{input_url}]

@chapter Description
@c man begin DESCRIPTION
@@ -47,9 +47,17 @@ Disable video.
@item -sn
 Disable subtitles.
@item -ss @var{pos}
-Seek to a given position in seconds.
+Seek to @var{pos}. Note that in most formats it is not possible to seek
+exactly, so @command{ffplay} will seek to the nearest seek point to
+@var{pos}.
+
+@var{pos} must be a time duration specification,
+see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@item -t @var{duration}
-play <duration> seconds of audio/video
+Play @var{duration} seconds of audio/video.
+
+@var{duration} must be a time duration specification,
+see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@item -bytes
 Seek by bytes.
@item -nodisp
@@ -98,8 +106,8 @@ the input audio.
 Use the option "-filters" to show all the available filters (including
 sources and sinks).

-@item -i @var{input_file}
-Read @var{input_file}.
+@item -i @var{input_url}
+Read @var{input_url}.
@end table

@section Advanced options
@@ -161,7 +169,7 @@ Force a specific video decoder.
 Force a specific subtitle decoder.

@item -autorotate
-Automatically rotate the video according to presentation metadata. Enabled by
+Automatically rotate the video according to file metadata. Enabled by
 default, use @option{-noautorotate} to disable it.

@item -framedrop
@@ -12,7 +12,7 @@

@chapter Synopsis

-ffprobe [@var{options}] [@file{input_file}]
+ffprobe [@var{options}] [@file{input_url}]

@chapter Description
@c man begin DESCRIPTION
@@ -24,8 +24,8 @@ For example it can be used to check the format of the container used
 by a multimedia stream and the format and type of each media stream
 contained in it.

-If a filename is specified in input, ffprobe will try to open and
-probe the file content. If the file cannot be opened or recognized as
+If a url is specified in input, ffprobe will try to open and
+probe the url content. If the url cannot be opened or recognized as
 a multimedia file, a positive exit code is returned.

 ffprobe may be employed both as a standalone application or in
@@ -332,8 +332,8 @@ with name "PIXEL_FORMAT".
 Force bitexact output, useful to produce output which is not dependent
 on the specific build.

-@item -i @var{input_file}
-Read @var{input_file}.
+@item -i @var{input_url}
+Read @var{input_url}.

@end table
@c man end
@@ -447,17 +447,17 @@ writer).
 It can assume one of the following values:
@table @option
@item c
-Perform C-like escaping. Strings containing a newline ('\n'), carriage
-return ('\r'), a tab ('\t'), a form feed ('\f'), the escaping
-character ('\') or the item separator character @var{SEP} are escaped using C-like fashioned
-escaping, so that a newline is converted to the sequence "\n", a
-carriage return to "\r", '\' to "\\" and the separator @var{SEP} is
-converted to "\@var{SEP}".
+Perform C-like escaping. Strings containing a newline (@samp{\n}), carriage
+return (@samp{\r}), a tab (@samp{\t}), a form feed (@samp{\f}), the escaping
+character (@samp{\}) or the item separator character @var{SEP} are escaped
+using C-like fashioned escaping, so that a newline is converted to the
+sequence @samp{\n}, a carriage return to @samp{\r}, @samp{\} to @samp{\\} and
+the separator @var{SEP} is converted to @samp{\@var{SEP}}.

@item csv
 Perform CSV-like escaping, as described in RFC4180.  Strings
-containing a newline ('\n'), a carriage return ('\r'), a double quote
-('"'), or @var{SEP} are enclosed in double-quotes.
+containing a newline (@samp{\n}), a carriage return (@samp{\r}), a double quote
+(@samp{"}), or @var{SEP} are enclosed in double-quotes.

@item none
 Perform no escaping.
@@ -485,7 +485,7 @@ The description of the accepted options follows.
 Separator character used to separate the chapter, the section name, IDs and
 potential tags in the printed field key.

-Default value is '.'.
+Default value is @samp{.}.

@item hierarchical, h
 Specify if the section name specification should be hierarchical. If
@@ -507,21 +507,22 @@ The following conventions are adopted:
@item
 all key and values are UTF-8
@item
-'.' is the subgroup separator
+@samp{.} is the subgroup separator
@item
-newline, '\t', '\f', '\b' and the following characters are escaped
+newline, @samp{\t}, @samp{\f}, @samp{\b} and the following characters are
+escaped
@item
-'\' is the escape character
+@samp{\} is the escape character
@item
-'#' is the comment indicator
+@samp{#} is the comment indicator
@item
-'=' is the key/value separator
+@samp{=} is the key/value separator
@item
-':' is not used but usually parsed as key/value separator
+@samp{:} is not used but usually parsed as key/value separator
@end itemize

 This writer accepts options as a list of @var{key}=@var{value} pairs,
-separated by ":".
+separated by @samp{:}.

 The description of the accepted options follows.

@@ -48,6 +48,11 @@
    </xsd:complexType>

    <xsd:complexType name="packetType">
+      <xsd:sequence>
+        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
+        <xsd:element name="side_data_list" type="ffprobe:packetSideDataListType"   minOccurs="0" maxOccurs="1" />
+      </xsd:sequence>
+
      <xsd:attribute name="codec_type"    type="xsd:string" use="required" />
      <xsd:attribute name="stream_index"  type="xsd:int" use="required" />
      <xsd:attribute name="pts"           type="xsd:long"  />
@@ -65,6 +70,16 @@
      <xsd:attribute name="data_hash"     type="xsd:string" />
    </xsd:complexType>

+    <xsd:complexType name="packetSideDataListType">
+        <xsd:sequence>
+            <xsd:element name="side_data" type="ffprobe:packetSideDataType" minOccurs="1" maxOccurs="unbounded"/>
+        </xsd:sequence>
+    </xsd:complexType>
+    <xsd:complexType name="packetSideDataType">
+        <xsd:attribute name="side_data_type"              type="xsd:string"/>
+        <xsd:attribute name="side_data_size"              type="xsd:int"   />
+    </xsd:complexType>
+
    <xsd:complexType name="frameType">
      <xsd:sequence>
            <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
@@ -72,6 +87,7 @@
      </xsd:sequence>

      <xsd:attribute name="media_type"    type="xsd:string" use="required"/>
+      <xsd:attribute name="stream_index"  type="xsd:int"  />
      <xsd:attribute name="key_frame"     type="xsd:int"    use="required"/>
      <xsd:attribute name="pts"           type="xsd:long" />
      <xsd:attribute name="pts_time"      type="xsd:float"/>
@@ -155,6 +171,7 @@
      <xsd:sequence>
        <xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
+        <xsd:element name="side_data_list" type="ffprobe:packetSideDataListType"   minOccurs="0" maxOccurs="1" />
      </xsd:sequence>

      <xsd:attribute name="index"            type="xsd:int" use="required"/>
@@ -171,6 +188,8 @@
      <!-- video attributes -->
      <xsd:attribute name="width"                type="xsd:int"/>
      <xsd:attribute name="height"               type="xsd:int"/>
+      <xsd:attribute name="coded_width"          type="xsd:int"/>
+      <xsd:attribute name="coded_height"         type="xsd:int"/>
      <xsd:attribute name="has_b_frames"         type="xsd:int"/>
      <xsd:attribute name="sample_aspect_ratio"  type="xsd:string"/>
      <xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
@@ -254,6 +273,8 @@
    <xsd:complexType name="programVersionType">
      <xsd:attribute name="version"          type="xsd:string" use="required"/>
      <xsd:attribute name="copyright"        type="xsd:string" use="required"/>
+      <xsd:attribute name="build_date"       type="xsd:string"/>
+      <xsd:attribute name="build_time"       type="xsd:string"/>
      <xsd:attribute name="compiler_ident"   type="xsd:string" use="required"/>
      <xsd:attribute name="configuration"    type="xsd:string" use="required"/>
    </xsd:complexType>
@@ -82,6 +82,7 @@ Feed feed1.ffm
 # ra         : RealNetworks-compatible stream. Audio only.
 # mpjpeg     : Multipart JPEG (works with Netscape without any plugin)
 # jpeg       : Generate a single JPEG image.
+# mjpeg      : Generate a M-JPEG stream.
 # asf        : ASF compatible streaming (Windows Media Player format).
 # swf        : Macromedia Flash compatible stream
 # avi        : AVI format (MPEG-4 video, MPEG audio sound)
@@ -72,7 +72,7 @@ the HTTP server (configured through the @option{HTTPPort} option), and
 configuration file.

 Each feed is associated to a file which is stored on disk. This stored
-file is used to allow to send pre-recorded data to a player as fast as
+file is used to send pre-recorded data to a player as fast as
 possible when new content is added in real-time to the stream.

 A "live-stream" or "stream" is a resource published by
@@ -118,7 +118,8 @@ Multiple streams can be connected to the same feed.

 For example, you can have a situation described by the following
 graph:
-@example
+
+@verbatim
               _________       __________
              |         |     |          |
 ffmpeg 1 -----| feed 1  |-----| stream 1 |
@@ -143,7 +144,8 @@ ffmpeg 2 -----| feed 3  |-----| stream 4 |
              |         |     |          |
              | file 1  |-----| stream 5 |
              |_________|     |__________|
-@end example
+
+@end verbatim

@anchor{FFM}
@section FFM, FFM2 formats
@@ -36,8 +36,10 @@ Possible forms of stream specifiers are:
 Matches the stream with this index. E.g. @code{-threads:1 4} would set the
 thread count for the second stream to 4.
@item @var{stream_type}[:@var{stream_index}]
-@var{stream_type} is one of following: 'v' for video, 'a' for audio, 's' for subtitle,
-'d' for data, and 't' for attachments. If @var{stream_index} is given, then it matches
+@var{stream_type} is one of following: 'v' or 'V' for video, 'a' for audio, 's'
+for subtitle, 'd' for data, and 't' for attachments. 'v' matches all video
+streams, 'V' only matches video streams which are not attached pictures, video
+thumbnails or cover arts.  If @var{stream_index} is given, then it matches
 stream number @var{stream_index} of this type. Otherwise, it matches all
 streams of this type.
@item p:@var{program_id}[:@var{stream_index}]
@@ -50,6 +52,9 @@ Match the stream by stream id (e.g. PID in MPEG-TS container).
 Matches streams with the metadata tag @var{key} having the specified value. If
@var{value} is not given, matches streams that contain the given tag with any
 value.
+@item u
+Matches streams with usable configuration, the codec must be defined and the
+essential information such as video dimension or audio sample rate must be present.

 Note that in @command{ffmpeg}, matching by metadata will only work properly for
 input files.
@@ -187,6 +192,7 @@ warnings and errors. This is the default value.
 Same as @code{info}, except more verbose.
@item debug, 48
 Show everything, including debugging information.
+@item trace, 56
@end table

 By default the program logs to stderr, if coloring is supported by the
@@ -261,10 +267,14 @@ Possible flags for this option are:
@item sse4.1
@item sse4.2
@item avx
+@item avx2
@item xop
+@item fma3
@item fma4
@item 3dnow
@item 3dnowext
+@item bmi1
+@item bmi2
@item cmov
@end table
@item ARM
@@ -275,6 +285,13 @@ Possible flags for this option are:
@item vfp
@item vfpv3
@item neon
+@item setend
+@end table
+@item AArch64
+@table @samp
+@item armv8
+@item vfp
+@item neon
@end table
@item PowerPC
@table @samp
@@ -98,7 +98,7 @@ Buffer references ownership and permissions
    The AVFilterLink structure has a few AVFilterBufferRef fields. The
    cur_buf and out_buf were used with the deprecated
    start_frame/draw_slice/end_frame API and should no longer be used.
-    src_buf, cur_buf_copy and partial_buf are used by libavfilter internally
+    src_buf and partial_buf are used by libavfilter internally
    and must not be accessed by filters.

  Reference permissions
@@ -23,7 +23,7 @@ Reduce buffering.

@item probesize @var{integer} (@emph{input})
 Set probing size in bytes, i.e. the size of the data to analyze to get
-stream information. A higher value will allow to detect more
+stream information. A higher value will enable detecting more
 information in case it is dispersed into the stream, but will increase
 latency. Must be an integer not lesser than 32. It is 5000000 by default.

@@ -37,6 +37,8 @@ Possible values:
@table @samp
@item ignidx
 Ignore index.
+@item fastseek
+Enable fast, but inaccurate seeks for some formats.
@item genpts
 Generate PTS.
@item nofillin
@@ -67,7 +69,7 @@ Default is 0.

@item analyzeduration @var{integer} (@emph{input})
 Specify how many microseconds are analyzed to probe the input. A
-higher value will allow to detect more accurate information, but will
+higher value will enable detecting more accurate information, but will
 increase latency. It defaults to 5,000,000 microseconds = 5 seconds.

@item cryptokey @var{hexadecimal string} (@emph{input})
@@ -125,6 +127,25 @@ Consider all spec non compliancies as errors.
 Consider things that a sane encoder should not do as an error.
@end table

+@item max_interleave_delta @var{integer} (@emph{output})
+Set maximum buffering duration for interleaving. The duration is
+expressed in microseconds, and defaults to 1000000 (1 second).
+
+To ensure all the streams are interleaved correctly, libavformat will
+wait until it has at least one packet for each stream before actually
+writing any packets to the output file. When some streams are
+"sparse" (i.e. there are large gaps between successive packets), this
+can result in excessive buffering.
+
+This field specifies the maximum difference between the timestamps of the
+first and the last packet in the muxing queue, above which libavformat
+will output a packet regardless of whether it has queued a packet for all
+the streams.
+
+If set to 0, libavformat will continue buffering packets until it has
+a packet for each stream, regardless of the maximum timestamp
+difference between the buffered packets.
+
@item use_wallclock_as_timestamps @var{integer} (@emph{input})
 Use wallclock as timestamps.

@@ -184,6 +205,10 @@ For example to separate the fields with newlines and indention:
 ffprobe -dump_separator "
                          "  -i ~/videos/matrixbench_mpeg2.mpg
@end example
+
+@item max_streams @var{integer} (@emph{input})
+Specifies the maximum number of streams. This can be used to reject files that
+would require too many resources due to a large number of streams.
@end table

@c man end FORMAT OPTIONS
@@ -145,6 +145,14 @@ x265 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 kvazaar
+
+FFmpeg can make use of the kvazaar library for HEVC encoding.
+
+Go to @url{https://github.com/ultravideo/kvazaar} and follow the
+instructions for installing the library. Then pass
+@code{--enable-libkvazaar} to configure to enable it.
+
@section libilbc

 iLBC is a narrowband speech codec that has been made freely available
@@ -179,8 +187,8 @@ included in compat/avisynth/, which allows the user to enable support
 without needing to search for these headers themselves.

 For Windows, supported AviSynth variants are
-@url{http://avisynth.nl, AviSynth 2.5 or 2.6} for 32-bit builds and
-@url{http://avs-plus.net, AviSynth+ 0.1} for 32-bit and 64-bit builds.
+@url{http://avisynth.nl, AviSynth 2.6 RC1 or higher} for 32-bit builds and
+@url{http://avs-plus.net, AviSynth+ r1718 or higher} for 32-bit and 64-bit builds.

 For Linux and OS X, the supported AviSynth variant is
@url{https://github.com/avxsynth/avxsynth, AvxSynth}.
@@ -192,6 +200,17 @@ end user having AviSynth or AvxSynth installed - they'll only need to be
 installed to use AviSynth scripts (obviously).
@end float

+@section Intel QuickSync Video
+
+FFmpeg can use Intel QuickSync Video (QSV) for accelerated encoding and decoding
+of multiple codecs. To use QSV, FFmpeg must be linked against the @code{libmfx}
+dispatcher, which loads the actual decoding libraries.
+
+The dispatcher is open source and can be downloaded from
+@url{https://github.com/lu-zero/mfx_dispatch.git}. FFmpeg needs to be configured
+with the @code{--enable-libmfx} option and @code{pkg-config} needs to be able to
+locate the dispatcher's @code{.pc} files.
+

@chapter Supported File Formats, Codecs or Features

@@ -207,6 +226,10 @@ library:
@item 4xm                       @tab   @tab X
    @tab 4X Technologies format, used in some games.
@item 8088flex TMV              @tab   @tab X
+@item AAX                       @tab   @tab X
+    @tab Audible Enhanced Audio format, used in audiobooks.
+@item AA                        @tab   @tab X
+    @tab Audible Format 2, 3, and 4, used in audiobooks.
@item ACT Voice                 @tab   @tab X
    @tab contains G.729 audio
@item Adobe Filmstrip           @tab X @tab X
@@ -222,6 +245,7 @@ library:
    @tab Audio format used on the Nintendo Gamecube.
@item AFC                       @tab   @tab X
    @tab Audio format used on the Nintendo Gamecube.
+@item APNG                      @tab X @tab X
@item ASF                       @tab X @tab X
@item AST                       @tab X @tab X
    @tab Audio format used on the Nintendo Wii.
@@ -242,6 +266,8 @@ library:
    @tab Used in Z and Z95 games.
@item Brute Force & Ignorance   @tab   @tab X
    @tab Used in the game Flash Traffic: City of Angels.
+@item BFSTM                     @tab   @tab X
+    @tab Audio format used on the Nintendo WiiU (based on BRSTM).
@item BRSTM                     @tab   @tab X
    @tab Audio format used on the Nintendo Wii.
@item BWF                       @tab X @tab X
@@ -253,6 +279,8 @@ library:
@item Delphine Software International CIN @tab   @tab X
    @tab Multimedia format used by Delphine Software games.
@item Digital Speech Standard (DSS) @tab   @tab X
+@item Canopus HQ                @tab   @tab X
+@item Canopus HQA               @tab   @tab X
@item Canopus HQX               @tab   @tab X
@item CD+G                      @tab   @tab X
    @tab Video format used by CD+G karaoke disks
@@ -270,6 +298,7 @@ library:
@item Deluxe Paint Animation    @tab   @tab X
@item DFA                       @tab   @tab X
    @tab This format is used in Chronomaster game
+@item DirectDraw Surface        @tab   @tab X
@item DSD Stream File (DSF)     @tab   @tab X
@item DV video                  @tab X @tab X
@item DXA                       @tab   @tab X
@@ -466,6 +495,7 @@ library:
@item SoX native format         @tab X @tab X
@item SUN AU format             @tab X @tab X
@item SUP raw PGS subtitles     @tab   @tab X
+@item TDSC                      @tab   @tab X
@item Text files                @tab   @tab X
@item THP                       @tab   @tab X
    @tab Used on the Nintendo GameCube.
@@ -473,6 +503,7 @@ library:
    @tab Tiertex .seq files used in the DOS CD-ROM version of the game Flashback.
@item True Audio                @tab   @tab X
@item VC-1 test bitstream       @tab X @tab X
+@item Vidvox Hap                @tab X @tab X
@item Vivo                      @tab   @tab X
@item WAV                       @tab X @tab X
@item WavPack                   @tab X @tab X
@@ -507,6 +538,7 @@ following image formats are supported:
@item Alias PIX    @tab X @tab X
    @tab Alias/Wavefront PIX image format
@item animated GIF @tab X @tab X
+@item APNG         @tab X @tab X
@item BMP          @tab X @tab X
    @tab Microsoft BMP image
@item BRender PIX  @tab   @tab X
@@ -658,6 +690,8 @@ following image formats are supported:
    @tab Sorenson H.263 used in Flash
@item Forward Uncompressed   @tab     @tab  X
@item Fraps                  @tab     @tab  X
+@item Go2Meeting             @tab     @tab  X
+    @tab fourcc: G2M2, G2M3
@item Go2Webinar             @tab     @tab  X
    @tab fourcc: G2M4
@item H.261                  @tab  X  @tab  X
@@ -666,7 +700,7 @@ following image formats are supported:
@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10  @tab  E  @tab  X
    @tab encoding supported through external library libx264 and OpenH264
@item HEVC                   @tab  X  @tab  X
-    @tab encoding supported through the external library libx265
+    @tab encoding supported through external library libx265 and libkvazaar
@item HNM version 4          @tab     @tab  X
@item HuffYUV                @tab  X  @tab  X
@item HuffYUV FFmpeg variant @tab  X  @tab  X
@@ -831,7 +865,7 @@ following image formats are supported:
@item Name @tab Encoding @tab Decoding @tab Comments
@item 8SVX exponential       @tab     @tab  X
@item 8SVX fibonacci         @tab     @tab  X
-@item AAC+                   @tab  E  @tab  X
+@item AAC+                   @tab  E  @tab  IX
    @tab encoding supported through external library libaacplus
@item AAC                    @tab  E  @tab  X
    @tab encoding supported through external library libfaac and libvo-aacenc
@@ -871,7 +905,7 @@ following image formats are supported:
@item ADPCM MS IMA           @tab  X  @tab  X
@item ADPCM Nintendo Gamecube AFC  @tab     @tab  X
@item ADPCM Nintendo Gamecube DTK  @tab     @tab  X
-@item ADPCM Nintendo Gamecube THP  @tab     @tab  X
+@item ADPCM Nintendo THP  @tab     @tab  X
@item ADPCM QT IMA           @tab  X  @tab  X
@item ADPCM SEGA CRI ADX     @tab  X  @tab  X
    @tab Used in Sega Dreamcast games.
@@ -905,6 +939,7 @@ following image formats are supported:
@item COOK                   @tab     @tab  X
    @tab All versions except 5.1 are supported.
@item DCA (DTS Coherent Acoustics)  @tab  X  @tab  X
+    @tab supported extensions: XCh, XLL (partially)
@item DPCM id RoQ            @tab  X  @tab  X
    @tab Used in Quake III, Jedi Knight 2 and other computer games.
@item DPCM Interplay         @tab     @tab  X
@@ -948,8 +983,8 @@ following image formats are supported:
@item Musepack SV8           @tab     @tab  X
@item Nellymoser Asao        @tab  X  @tab  X
@item On2 AVC (Audio for Video Codec) @tab     @tab  X
-@item Opus                   @tab  E  @tab  E
-    @tab supported through external library libopus
+@item Opus                   @tab  E  @tab  X
+    @tab encoding supported through external library libopus
@item PCM A-law              @tab  X  @tab  X
@item PCM mu-law             @tab  X  @tab  X
@item PCM signed 8-bit planar  @tab  X  @tab  X
@@ -1,10 +1,10 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8

-@settitle Using git to develop FFmpeg
+@settitle Using Git to develop FFmpeg

@titlepage
-@center @titlefont{Using git to develop FFmpeg}
+@center @titlefont{Using Git to develop FFmpeg}
@end titlepage

@top
@@ -13,9 +13,9 @@

@chapter Introduction

-This document aims in giving some quick references on a set of useful git
+This document aims in giving some quick references on a set of useful Git
 commands. You should always use the extensive and detailed documentation
-provided directly by git:
+provided directly by Git:

@example
 git --help
@@ -32,22 +32,21 @@ man git-<command>
 shows information about the subcommand <command>.

 Additional information could be found on the
-@url{http://gitref.org, Git Reference} website
+@url{http://gitref.org, Git Reference} website.

 For more information about the Git project, visit the
-
-@url{http://git-scm.com/, Git website}
+@url{http://git-scm.com/, Git website}.

 Consult these resources whenever you have problems, they are quite exhaustive.

 What follows now is a basic introduction to Git and some FFmpeg-specific
-guidelines to ease the contribution to the project
+guidelines to ease the contribution to the project.

@chapter Basics Usage

-@section Get GIT
+@section Get Git

-You can get git from @url{http://git-scm.com/}
+You can get Git from @url{http://git-scm.com/}
 Most distribution and operating system provide a package for it.


@@ -75,6 +74,7 @@ git config --global core.autocrlf false
@end example


+@anchor{Updating the source tree to the latest revision}
@section Updating the source tree to the latest revision

@example
@@ -107,7 +107,7 @@ git add [-A] <filename/dirname>
 git rm [-r] <filename/dirname>
@end example

-GIT needs to get notified of all changes you make to your working
+Git needs to get notified of all changes you make to your working
 directory that makes files appear or disappear.
 Line moves across files are automatically tracked.

@@ -127,8 +127,8 @@ will show all local modifications in your working directory as unified diff.
 git log <filename(s)>
@end example

-You may also use the graphical tools like gitview or gitk or the web
-interface available at http://source.ffmpeg.org/
+You may also use the graphical tools like @command{gitview} or @command{gitk}
+or the web interface available at @url{http://source.ffmpeg.org/}.

@section Checking source tree status

@@ -149,6 +149,7 @@ git diff --check
 to double check your changes before committing them to avoid trouble later
 on. All experienced developers do this on each and every commit, no matter
 how small.
+
 Every one of them has been saved from looking like a fool by this many times.
 It's very easy for stray debug output or cosmetic modifications to slip in,
 please avoid problems through this extra level of scrutiny.
@@ -171,14 +172,14 @@ to make sure you don't have untracked files or deletions.
 git add [-i|-p|-A] <filenames/dirnames>
@end example

-Make sure you have told git your name and email address
+Make sure you have told Git your name and email address

@example
 git config --global user.name "My Name"
 git config --global user.email my@@email.invalid
@end example

-Use @var{--global} to set the global configuration for all your git checkouts.
+Use @option{--global} to set the global configuration for all your Git checkouts.

 Git will select the changes to the files for commit. Optionally you can use
 the interactive or the patch mode to select hunk by hunk what should be
@@ -209,7 +210,7 @@ include filenames in log messages, Git provides that information.

 Possibly make the commit message have a terse, descriptive first line, an
 empty line and then a full description. The first line will be used to name
-the patch by git format-patch.
+the patch by @command{git format-patch}.

@section Preparing a patchset

@@ -325,12 +326,14 @@ faulty commit disappear from the history.
@section Pushing changes to remote trees

@example
-git push
+git push origin master --dry-run
@end example

-Will push the changes to the default remote (@var{origin}).
+Will simulate a push of the local master branch to the default remote
+(@var{origin}). And list which branches and ranges or commits would have been
+pushed.
 Git will prevent you from pushing changes if the local and remote trees are
-out of sync. Refer to and to sync the local tree.
+out of sync. Refer to @ref{Updating the source tree to the latest revision}.

@example
 git remote add <name> <url>
@@ -349,23 +352,24 @@ branches matching the local ones.

@section Finding a specific svn revision

-Since version 1.7.1 git supports @var{:/foo} syntax for specifying commits
+Since version 1.7.1 Git supports @samp{:/foo} syntax for specifying commits
 based on a regular expression. see man gitrevisions

@example
 git show :/'as revision 23456'
@end example

-will show the svn changeset @var{r23456}. With older git versions searching in
+will show the svn changeset @samp{r23456}. With older Git versions searching in
 the @command{git log} output is the easiest option (especially if a pager with
 search capabilities is used).
+
 This commit can be checked out with

@example
 git checkout -b svn_23456 :/'as revision 23456'
@end example

-or for git < 1.7.1 with
+or for Git < 1.7.1 with

@example
 git checkout -b svn_23456 $SHA1
@@ -374,7 +378,7 @@ git checkout -b svn_23456 $SHA1
 where @var{$SHA1} is the commit hash from the @command{git log} output.


-@chapter pre-push checklist
+@chapter Pre-push checklist

 Once you have a set of commits that you feel are ready for pushing,
 work through the following checklist to doublecheck everything is in
@@ -385,7 +389,7 @@ Apply your common sense, but if in doubt, err on the side of caution.
 First, make sure that the commits and branches you are going to push
 match what you want pushed and that nothing is missing, extraneous or
 wrong. You can see what will be pushed by running the git push command
-with --dry-run first. And then inspecting the commits listed with
+with @option{--dry-run} first. And then inspecting the commits listed with
@command{git log -p 1234567..987654}. The @command{git status} command
 may help in finding local changes that have been forgotten to be added.

@@ -394,7 +398,7 @@ Next let the code pass through a full run of our testsuite.
@itemize
@item @command{make distclean}
@item @command{/path/to/ffmpeg/configure}
-@item @command{make check}
+@item @command{make fate}
@item if fate fails due to missing samples run @command{make fate-rsync} and retry
@end itemize

@@ -412,5 +416,5 @@ recommended.

@chapter Server Issues

-Contact the project admins @email{root@@ffmpeg.org} if you have technical
-problems with the GIT server.
+Contact the project admins at @email{root@@ffmpeg.org} if you have technical
+problems with the Git server.
@@ -1,7 +1,7 @@
@chapter Input Devices
@c man begin INPUT DEVICES

-Input devices are configured elements in FFmpeg which allow to access
+Input devices are configured elements in FFmpeg which enable accessing
 the data coming from a multimedia device attached to your system.

 When you configure your FFmpeg build, all the supported input devices
@@ -51,6 +51,18 @@ ffmpeg -f alsa -i hw:0 alsaout.wav
 For more information see:
@url{http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html}

+@subsection Options
+
+@table @option
+
+@item sample_rate
+Set the sample rate in Hz. Default is 48000.
+
+@item channels
+Set the number of channels. Default is 2.
+
+@end table
+
@section avfoundation

 AVFoundation input device.
@@ -114,6 +126,19 @@ und the first one in this list is used instead. Available pixel formats are:
 bgr48be, uyvy422, yuva444p, yuva444p16le, yuv444p, yuv422p16, yuv422p10, yuv444p10,
 yuv420p, nv12, yuyv422, gray}

+@item -framerate
+Set the grabbing frame rate. Default is @code{ntsc}, corresponding to a
+frame rate of @code{30000/1001}.
+
+@item -video_size
+Set the video frame size.
+
+@item -capture_cursor
+Capture the mouse pointer. Default is 0.
+
+@item -capture_mouse_clicks
+Capture the screen mouse clicks. Default is 0.
+
@end table

@subsection Examples
@@ -150,6 +175,99 @@ $ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi

 BSD video input device.

+@subsection Options
+
+@table @option
+
+@item framerate
+Set the frame rate.
+
+@item video_size
+Set the video frame size. Default is @code{vga}.
+
+@item standard
+
+Available values are:
+@table @samp
+@item pal
+
+@item ntsc
+
+@item secam
+
+@item paln
+
+@item palm
+
+@item ntscj
+
+@end table
+
+@end table
+
+@section decklink
+
+The decklink input device provides capture capabilities for Blackmagic
+DeckLink devices.
+
+To enable this input device, you need the Blackmagic DeckLink SDK and you
+need to configure with the appropriate @code{--extra-cflags}
+and @code{--extra-ldflags}.
+On Windows, you need to run the IDL files through @command{widl}.
+
+DeckLink is very picky about the formats it supports. Pixel format is
+uyvy422 or v210, framerate and video size must be determined for your device with
+@command{-list_formats 1}. Audio sample rate is always 48 kHz and the number
+of channels can be 2, 8 or 16.
+
+@subsection Options
+
+@table @option
+
+@item list_devices
+If set to @option{true}, print a list of devices and exit.
+Defaults to @option{false}.
+
+@item list_formats
+If set to @option{true}, print a list of supported formats and exit.
+Defaults to @option{false}.
+
+@item bm_v210
+If set to @samp{1}, video is captured in 10 bit v210 instead
+of uyvy422. Not all Blackmagic devices support this option.
+
+@end table
+
+@subsection Examples
+
+@itemize
+
+@item
+List input devices:
+@example
+ffmpeg -f decklink -list_devices 1 -i dummy
+@end example
+
+@item
+List supported formats:
+@example
+ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'
+@end example
+
+@item
+Capture video clip at 1080i50 (format 11):
+@example
+ffmpeg -f decklink -i 'Intensity Pro@@11' -acodec copy -vcodec copy output.avi
+@end example
+
+@item
+Capture video clip at 1080i50 10 bit:
+@example
+ffmpeg -bm_v210 1 -f decklink -i 'UltraStudio Mini Recorder@@11' -acodec copy -vcodec copy output.avi
+@end example
+
+@end itemize
+
@section dshow

 Windows DirectShow input device.
@@ -200,11 +318,11 @@ If set to @option{true}, print a list of selected device's options
 and exit.

@item video_device_number
-Set video device number for devices with same name (starts at 0,
+Set video device number for devices with the same name (starts at 0,
 defaults to 0).

@item audio_device_number
-Set audio device number for devices with same name (starts at 0,
+Set audio device number for devices with the same name (starts at 0,
 defaults to 0).

@item pixel_format
@@ -275,6 +393,30 @@ If set to @option{true}, before capture starts, popup a display
 dialog to the end user, allowing them to manually
 modify TV audio (like mono vs. stereo, Language A,B or C).

+@item audio_device_load
+Load an audio capture filter device from file instead of searching
+it by name. It may load additional parameters too, if the filter
+supports the serialization of its properties to.
+To use this an audio capture source has to be specified, but it can
+be anything even fake one.
+
+@item audio_device_save
+Save the currently used audio capture filter device and its
+parameters (if the filter supports it) to a file.
+If a file with the same name exists it will be overwritten.
+
+@item video_device_load
+Load a video capture filter device from file instead of searching
+it by name. It may load additional parameters too, if the filter
+supports the serialization of its properties to.
+To use this a video capture source has to be specified, but it can
+be anything even fake one.
+
+@item video_device_save
+Save the currently used video capture filter device and its
+parameters (if the filter supports it) to a file.
+If a file with the same name exists it will be overwritten.
+
@end table

@subsection Examples
@@ -330,6 +472,27 @@ $ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_numbe

 Linux DV 1394 input device.

+@subsection Options
+
+@table @option
+
+@item framerate
+Set the frame rate. Default is 25.
+
+@item standard
+
+Available values are:
+@table @samp
+@item pal
+
+@item ntsc
+
+@end table
+
+Default value is @code{ntsc}.
+
+@end table
+
@section fbdev

 Linux framebuffer input device.
@@ -342,18 +505,27 @@ console. It is accessed through a file device node, usually
 For more detailed information read the file
 Documentation/fb/framebuffer.txt included in the Linux source tree.

+See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
+
 To record from the framebuffer device @file{/dev/fb0} with
@command{ffmpeg}:
@example
-ffmpeg -f fbdev -r 10 -i /dev/fb0 out.avi
+ffmpeg -f fbdev -framerate 10 -i /dev/fb0 out.avi
@end example

 You can take a single screenshot image with the command:
@example
-ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg
+ffmpeg -f fbdev -framerate 1 -i /dev/fb0 -frames:v 1 screenshot.jpeg
@end example

-See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
+@subsection Options
+
+@table @option
+
+@item framerate
+Set the frame rate. Default is 25.
+
+@end table

@section gdigrab

@@ -539,6 +711,15 @@ $ jack_connect metro:120_bpm ffmpeg:input_1
 For more information read:
@url{http://jackaudio.org/}

+@subsection Options
+
+@table @option
+
+@item channels
+Set the number of channels. Default is 2.
+
+@end table
+
@section lavfi

 Libavfilter input virtual device.
@@ -579,6 +760,9 @@ Set the filename of the filtergraph to be read and sent to the other
 filters. Syntax of the filtergraph is the same as the one specified by
 the option @var{graph}.

+@item dumpgraph
+Dump graph to stderr.
+
@end table

@subsection Examples
@@ -780,6 +964,19 @@ ffmpeg -f oss -i /dev/dsp /tmp/oss.wav
 For more information about OSS see:
@url{http://manuals.opensound.com/usersguide/dsp.html}

+@subsection Options
+
+@table @option
+
+@item sample_rate
+Set the sample rate in Hz. Default is 48000.
+
+@item channels
+Set the number of channels. Default is 2.
+
+@end table
+
+
@section pulse

 PulseAudio input device.
@@ -820,6 +1017,10 @@ Specify the number of bytes per frame, by default it is set to 1024.
@item fragment_size
 Specify the minimal buffering fragment in PulseAudio, it will affect the
 audio latency. By default it is unset.
+
+@item wallclock
+Set the initial PTS using the current time. Default is 1.
+
@end table

@subsection Examples
@@ -855,6 +1056,22 @@ ffmpeg -f qtkit -i "default" out.mpg
 ffmpeg -f qtkit -list_devices true -i ""
@end example

+@subsection Options
+
+@table @option
+
+@item frame_rate
+Set frame rate. Default is 30.
+
+@item list_devices
+If set to @code{true}, print a list of devices and exit. Default is
+@code{false}.
+
+@item video_device_index
+Select the video device by index for devices with the same name (starts at 0).
+
+@end table
+
@section sndio

 sndio input device.
@@ -872,6 +1089,18 @@ command:
 ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
@end example

+@subsection Options
+
+@table @option
+
+@item sample_rate
+Set the sample rate in Hz. Default is 48000.
+
+@item channels
+Set the number of channels. Default is 2.
+
+@end table
+
@section video4linux2, v4l2

 Video4Linux2 input video device.
@@ -904,6 +1133,12 @@ conversion into the real time clock.
 Some usage examples of the video4linux2 device with @command{ffmpeg}
 and @command{ffplay}:
@itemize
+@item
+List supported formats for a video4linux2 device:
+@example
+ffplay -f video4linux2 -list_formats all /dev/video0
+@end example
+
@item
 Grab and show the input of a video4linux2 device:
@example
@@ -988,6 +1223,10 @@ Force conversion from monotonic to absolute timestamps.
@end table

 Default value is @code{default}.
+
+@item use_libv4l2
+Use libv4l2 (v4l-utils) conversion functions. Default is 0.
+
@end table

@section vfwcap
@@ -998,6 +1237,19 @@ The filename passed as input is the capture driver number, ranging from
 0 to 9. You may use "list" as filename to print a list of drivers. Any
 other filename will be interpreted as device number 0.

+@subsection Options
+
+@table @option
+
+@item video_size
+Set the video frame size.
+
+@item framerate
+Set the grabbing frame rate. Default value is @code{ntsc},
+corresponding to a frame rate of @code{30000/1001}.
+
+@end table
+
@section x11grab

 X11 video input device.
@@ -1109,89 +1361,8 @@ The syntax is:
 -grab_x @var{x_offset} -grab_y @var{y_offset}
@end example

-Set the grabing region coordinates. The are expressed as offset from the top left
+Set the grabbing region coordinates. They are expressed as offset from the top left
 corner of the X11 window. The default value is 0.

-@section decklink
-
-The decklink input device provides capture capabilities for Blackmagic
-DeckLink devices.
-
-To enable this input device, you need the Blackmagic DeckLink SDK and you
-need to configure with the appropriate @code{--extra-cflags}
-and @code{--extra-ldflags}.
-On Windows, you need to run the IDL files through @command{widl}.
-
-DeckLink is very picky about the formats it supports. Pixel format is
-uyvy422 or v210, framerate and video size must be determined for your device with
-@command{-list_formats 1}. Audio sample rate is always 48 kHz and the number
-of channels can be 2, 8 or 16.
-
-@subsection Options
-
-@table @option
-
-@item list_devices
-If set to @option{true}, print a list of devices and exit.
-Defaults to @option{false}.
-
-@item list_formats
-If set to @option{true}, print a list of supported formats and exit.
-Defaults to @option{false}.
-
-@item bm_v210
-If set to @samp{1}, video is captured in 10 bit v210 instead
-of uyvy422. Not all Blackmagic devices support this option.
-
-@item bm_channels <CHANNELS>
-Number of audio channels, can be 2, 8 or 16
-
-@item bm_audiodepth <BITDEPTH>
-Audio bit depth, can be 16 or 32.
-
-@end table
-
-@subsection Examples
-
-@itemize
-
-@item
-List input devices:
-@example
-ffmpeg -f decklink -list_devices 1 -i dummy
-@end example
-
-@item
-List supported formats:
-@example
-ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'
-@end example
-
-@item
-Capture video clip at 1080i50 (format 11):
-@example
-ffmpeg -f decklink -i 'Intensity Pro@@11' -acodec copy -vcodec copy output.avi
-@end example
-
-@item
-Capture video clip at 1080i50 10 bit:
-@example
-ffmpeg -bm_v210 1 -f decklink -i 'UltraStudio Mini Recorder@@11' -acodec copy -vcodec copy output.avi
-@end example
-
-@item
-Capture video clip at 720p50 with 32bit audio:
-@example
-ffmpeg -bm_audiodepth 32 -f decklink -i 'UltraStudio Mini Recorder@@14' -acodec copy -vcodec copy output.avi
-@end example
-
-@item
-Capture video clip at 576i50 with 8 audio channels:
-@example
-ffmpeg -bm_channels 8 -f decklink -i 'UltraStudio Mini Recorder@@3' -acodec copy -vcodec copy output.avi
-@end example
-
-@end itemize
-

@c man end INPUT DEVICES
@@ -12,10 +12,10 @@ A file consists of a header and a number of metadata tags divided into sections,
 each on its own line.

@item
-The header is a ';FFMETADATA' string, followed by a version number (now 1).
+The header is a @samp{;FFMETADATA} string, followed by a version number (now 1).

@item
-Metadata tags are of the form 'key=value'
+Metadata tags are of the form @samp{key=value}

@item
 Immediately after header follows global metadata
@@ -26,26 +26,30 @@ metadata.

@item
 A section starts with the section name in uppercase (i.e. STREAM or CHAPTER) in
-brackets ('[', ']') and ends with next section or end of file.
+brackets (@samp{[}, @samp{]}) and ends with next section or end of file.

@item
 At the beginning of a chapter section there may be an optional timebase to be
-used for start/end values. It must be in form 'TIMEBASE=num/den', where num and
-den are integers. If the timebase is missing then start/end times are assumed to
+used for start/end values. It must be in form
+@samp{TIMEBASE=@var{num}/@var{den}}, where @var{num} and @var{den} are
+integers. If the timebase is missing then start/end times are assumed to
 be in milliseconds.
+
 Next a chapter section must contain chapter start and end times in form
-'START=num', 'END=num', where num is a positive integer.
+@samp{START=@var{num}}, @samp{END=@var{num}}, where @var{num} is a positive
+integer.

@item
-Empty lines and lines starting with ';' or '#' are ignored.
+Empty lines and lines starting with @samp{;} or @samp{#} are ignored.

@item
-Metadata keys or values containing special characters ('=', ';', '#', '\' and a
-newline) must be escaped with a backslash '\'.
+Metadata keys or values containing special characters (@samp{=}, @samp{;},
+@samp{#}, @samp{\} and a newline) must be escaped with a backslash @samp{\}.

@item
-Note that whitespace in metadata (e.g. foo = bar) is considered to be a part of
-the tag (in the example above key is 'foo ', value is ' bar').
+Note that whitespace in metadata (e.g. @samp{foo = bar}) is considered to be
+a part of the tag (in the example above key is @samp{foo }, value is
+@samp{ bar}).
@end enumerate

 A ffmetadata file might look like this:
@@ -47,12 +47,16 @@ Files that have MIPS copyright notice in them:
 * libavutil/mips/
      float_dsp_mips.c
      libm_mips.h
+      softfloat_tables.h
 * libavcodec/
      fft_fixed_32.c
      fft_init_table.c
      fft_table.h
      mdct_fixed_32.c
 * libavcodec/mips/
+      aacdec_fixed.c
+      aacsbr_fixed.c
+      aacsbr_template.c
      aaccoder_mips.c
      aacpsy_mips.h
      ac3dsp_mips.c
@@ -54,7 +54,7 @@ thread.
 If the codec allocates writable tables in its init(), add an init_thread_copy()
 which re-allocates them for other threads.

-Add CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little
+Add AV_CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little
 speed gain at this point but it should work.

 If there are inter-frame dependencies, so the codec calls
@@ -263,6 +263,62 @@ ffmpeg in.nut -hls_segment_filename 'file%03d.ts' out.m3u8
 This example will produce the playlist, @file{out.m3u8}, and segment files:
@file{file000.ts}, @file{file001.ts}, @file{file002.ts}, etc.

+@item hls_key_info_file @var{key_info_file}
+Use the information in @var{key_info_file} for segment encryption. The first
+line of @var{key_info_file} specifies the key URI written to the playlist. The
+key URL is used to access the encryption key during playback. The second line
+specifies the path to the key file used to obtain the key during the encryption
+process. The key file is read as a single packed array of 16 octets in binary
+format. The optional third line specifies the initialization vector (IV) as a
+hexadecimal string to be used instead of the segment sequence number (default)
+for encryption. Changes to @var{key_info_file} will result in segment
+encryption with the new key/IV and an entry in the playlist for the new key
+URI/IV.
+
+Key info file format:
+@example
+@var{key URI}
+@var{key file path}
+@var{IV} (optional)
+@end example
+
+Example key URIs:
+@example
+http://server/file.key
+/path/to/file.key
+file.key
+@end example
+
+Example key file paths:
+@example
+file.key
+/path/to/file.key
+@end example
+
+Example IV:
+@example
+0123456789ABCDEF0123456789ABCDEF
+@end example
+
+Key info file example:
+@example
+http://server/file.key
+/path/to/file.key
+0123456789ABCDEF0123456789ABCDEF
+@end example
+
+Example shell script:
+@example
+#!/bin/sh
+BASE_URL=$@{1:-'.'@}
+openssl rand 16 > file.key
+echo $BASE_URL/file.key > file.keyinfo
+echo file.key >> file.keyinfo
+echo $(openssl rand -hex 16) >> file.keyinfo
+ffmpeg -f lavfi -re -i testsrc -c:v h264 -hls_flags delete_segments \
+  -hls_key_info_file file.keyinfo out.m3u8
+@end example
+
@item hls_flags single_file
 If this flag is set, the muxer will store all segments in a single MPEG-TS
 file, and will use byte ranges in the playlist. HLS playlists generated with
@@ -611,6 +667,13 @@ point on IIS with this muxer. Example:
 ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1)
@end example

+@subsection Audible AAX
+
+Audible AAX files are encrypted M4B files, and they can be decrypted by specifying a 4 byte activation secret.
+@example
+ffmpeg -activation_bytes 1CEB00DA -i test.aax -vn -c:a copy output.mp4
+@end example
+
@section mp3

 The MP3 muxer writes a raw MP3 stream with the following optional features:
@@ -703,6 +766,10 @@ Set a constant muxrate (default VBR).
@item -pcr_period @var{numer}
 Override the default PCR retransmission time (default 20ms), ignored
 if variable muxrate is selected.
+@item pat_period @var{number}
+Maximal time in seconds between PAT/PMT tables.
+@item sdt_period @var{number}
+Maximal time in seconds between SDT tables.
@item -pes_payload_size @var{number}
 Set minimum PES packet payload in bytes.
@item -mpegts_flags @var{flags}
@@ -754,6 +821,8 @@ Option mpegts_flags may take a set of such flags:
 Reemit PAT/PMT before writing the next packet.
@item latm
 Use LATM packetization for AAC.
+@item pat_pmt_at_frames
+Reemit PAT and PMT at each video frame.
@end table

@subsection Example
@@ -770,6 +839,21 @@ ffmpeg -i file.mpg -c copy \
     -y out.ts
@end example

+@section mxf, mxf_d10
+
+MXF muxer.
+
+@subsection Options
+
+The muxer options are:
+
+@table @option
+@item store_user_comments @var{bool}
+Set if user comments should be stored if available or never.
+IRT D-10 does not allow user comments. The default is thus to write them for
+mxf but not for mxf_d10
+@end table
+
@section null

 Null muxer.
@@ -908,13 +992,6 @@ Allow caching (only affects M3U8 list files).
 Allow live-friendly file generation.
@end table

-@item segment_list_type @var{type}
-Select the listing format.
-@table @option
-@item @var{flat} use a simple flat list of entries.
-@item @var{hls} use a m3u8-like structure.
-@end table
-
@item segment_list_size @var{size}
 Update the list file so that it contains at most @var{size}
 segments. If 0 the list file will contain all the segments. Default
@@ -924,6 +1001,9 @@ value is 0.
 Prepend @var{prefix} to each entry. Useful to generate absolute paths.
 By default no prefix is applied.

+@item segment_list_type @var{type}
+Select the listing format.
+
 The following values are recognized:
@table @samp
@item flat
@@ -1030,6 +1110,12 @@ segments to write. If this is selected, the output segment name must
 contain a @code{strftime} function template. Default value is
@code{0}.

+@item break_non_keyframes @var{1|0}
+If enabled, allow segments to start on frames other than keyframes. This
+improves behavior on some players when the time between keyframes is
+inconsistent, but may make things worse on others, and can cause some oddities
+during seeking. Defaults to @code{0}.
+
@item reset_timestamps @var{1|0}
 Reset timestamps at the begin of each segment, so that each segment
 will start with near-zero timestamps. It is meant to ease the playback
@@ -1210,7 +1296,17 @@ is the @option{global_header} flag.

 WebM DASH Manifest muxer.

-This muxer implements the WebM DASH Manifest specification to generate the DASH manifest XML.
+This muxer implements the WebM DASH Manifest specification to generate the DASH
+manifest XML. It also supports manifest generation for DASH live streams.
+
+For more information see:
+
+@itemize @bullet
+@item
+WebM DASH Specification: @url{https://sites.google.com/a/webmproject.org/wiki/adaptive-streaming/webm-dash-specification}
+@item
+ISO DASH Specification: @url{http://standards.iso.org/ittf/PubliclyAvailableStandards/c065274_ISO_IEC_23009-1_2014.zip}
+@end itemize

@subsection Options

@@ -1221,6 +1317,32 @@ This muxer supports the following options:
 This option has the following syntax: "id=x,streams=a,b,c id=y,streams=d,e" where x and y are the
 unique identifiers of the adaptation sets and a,b,c,d and e are the indices of the corresponding
 audio and video streams. Any number of adaptation sets can be added using this option.
+
+@item live
+Set this to 1 to create a live stream DASH Manifest. Default: 0.
+
+@item chunk_start_index
+Start index of the first chunk. This will go in the @samp{startNumber} attribute
+of the @samp{SegmentTemplate} element in the manifest. Default: 0.
+
+@item chunk_duration_ms
+Duration of each chunk in milliseconds. This will go in the @samp{duration}
+attribute of the @samp{SegmentTemplate} element in the manifest. Default: 1000.
+
+@item utc_timing_url
+URL of the page that will return the UTC timestamp in ISO format. This will go
+in the @samp{value} attribute of the @samp{UTCTiming} element in the manifest.
+Default: None.
+
+@item time_shift_buffer_depth
+Smallest time (in seconds) shifting buffer for which any Representation is
+guaranteed to be available. This will go in the @samp{timeShiftBufferDepth}
+attribute of the @samp{MPD} element. Default: 60.
+
+@item minimum_update_period
+Minimum update period (in seconds) of the manifest. This will go in the
+@samp{minimumUpdatePeriod} attribute of the @samp{MPD} element. Default: 0.
+
@end table

@subsection Example
@@ -1236,4 +1358,47 @@ ffmpeg -f webm_dash_manifest -i video1.webm \
       manifest.xml
@end example

+@section webm_chunk
+
+WebM Live Chunk Muxer.
+
+This muxer writes out WebM headers and chunks as separate files which can be
+consumed by clients that support WebM Live streams via DASH.
+
+@subsection Options
+
+This muxer supports the following options:
+
+@table @option
+@item chunk_start_index
+Index of the first chunk (defaults to 0).
+
+@item header
+Filename of the header where the initialization data will be written.
+
+@item audio_chunk_duration
+Duration of each audio chunk in milliseconds (defaults to 5000).
+@end table
+
+@subsection Example
+@example
+ffmpeg -f v4l2 -i /dev/video0 \
+       -f alsa -i hw:0 \
+       -map 0:0 \
+       -c:v libvpx-vp9 \
+       -s 640x360 -keyint_min 30 -g 30 \
+       -f webm_chunk \
+       -header webm_live_video_360.hdr \
+       -chunk_start_index 1 \
+       webm_live_video_360_%d.chk \
+       -map 1:0 \
+       -c:a libvorbis \
+       -b:a 128k \
+       -f webm_chunk \
+       -header webm_live_audio_128.hdr \
+       -chunk_start_index 1 \
+       -audio_chunk_duration 1000 \
+       webm_live_audio_128_%d.chk
+@end example
+
@c man end MUXERS
@@ -175,12 +175,6 @@ Notes:

@itemize

-@item It is possible that coreutils' @code{link.exe} conflicts with MSVC's linker.
-You can find out by running @code{which link} to see which @code{link.exe} you
-are using. If it is located at @code{/bin/link.exe}, then you have the wrong one
-in your @code{PATH}. Either move or remove that copy, or make sure MSVC's
-@code{link.exe} takes precedence in your @code{PATH} over coreutils'.
-
@item If you wish to build with zlib support, you will have to grab a compatible
 zlib binary from somewhere, with an MSVC import lib, or if you wish to link
 statically, you can follow the instructions below to build a compatible
@@ -19,6 +19,18 @@ supported protocols.

 A description of the currently available protocols follows.

+@section async
+
+Asynchronous data filling wrapper for input stream.
+
+Fill data in a background thread, to decouple I/O operation from demux thread.
+
+@example
+async:@var{URL}
+async:http://host/resource
+async:cache:http://host/resource
+@end example
+
@section bluray

 Read BluRay playlist.
@@ -63,7 +75,7 @@ cache:@var{URL}

 Physical concatenation protocol.

-Allow to read and seek from many resource in sequence as if they were
+Read and seek from many resources in sequence as if they were
 a unique resource.

 A URL accepted by this protocol has the syntax:
@@ -117,7 +129,7 @@ ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP/////////////

 File access protocol.

-Allow to read from or write to a file.
+Read from or write to a file.

 A file URL can have the form:
@example
@@ -155,7 +167,7 @@ time, which is valuable for files on slow medium.

 FTP (File Transfer Protocol).

-Allow to read from or write to remote resources using FTP protocol.
+Read from or write to remote resources using FTP protocol.

 Following syntax is required.
@example
@@ -277,6 +289,43 @@ Set initial byte offset.

@item end_offset
 Try to limit the request to bytes preceding this offset.
+
+@item method
+When used as a client option it sets the HTTP method for the request.
+
+When used as a server option it sets the HTTP method that is going to be
+expected from the client(s).
+If the expected and the received HTTP method do not match the client will
+be given a Bad Request response.
+When unset the HTTP method is not checked for now. This will be replaced by
+autodetection in the future.
+
+@item listen
+If set to 1 enables experimental HTTP server. This can be used to send data when
+used as an output option, or read data from a client with HTTP POST when used as
+an input option.
+If set to 2 enables experimental mutli-client HTTP server. This is not yet implemented
+in ffmpeg.c or ffserver.c and thus must not be used as a command line option.
+@example
+# Server side (sending):
+ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
+
+# Client side (receiving):
+ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
+
+# Client can also be done with wget:
+wget http://@var{server}:@var{port} -O somefile.ogg
+
+# Server side (receiving):
+ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
+
+# Client side (sending):
+ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
+
+# Client can also be done with wget:
+wget --post-file=somefile.ogg http://@var{server}:@var{port}
+@end example
+
@end table

@subsection HTTP Cookies
@@ -374,7 +423,7 @@ be seekable, so they will fail with the MD5 output protocol.

 UNIX pipe access protocol.

-Allow to read and write from UNIX pipes.
+Read and write from UNIX pipes.

 The accepted syntax is:
@example
@@ -614,7 +663,7 @@ For more information see: @url{http://www.samba.org/}.

 Secure File Transfer Protocol via libssh

-Allow to read from or write to remote resources using SFTP protocol.
+Read from or write to remote resources using SFTP protocol.

 Following syntax is required.

@@ -1055,7 +1104,9 @@ subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
@end example

 Play an AVI file directly from a TAR archive:
+@example
 subfile,,start,183241728,end,366490624,,:archive.tar
+@end example

@section tcp

@@ -122,6 +122,22 @@ a_dither).

@end table

+@item alphablend
+Set the alpha blending to use when the input has alpha but the output does not.
+Default value is @samp{none}.
+
+@table @samp
+@item uniform_color
+Blend onto a uniform background color
+
+@item checkerboard
+Blend onto a checkerboard
+
+@item none
+No blending
+
+@end table
+
@end table

@c man end SCALER OPTIONS
@@ -166,7 +166,7 @@ INF: while(<$inf>) {
        if ($ended =~ /^(?:ifset|ifclear|ignore|menu|iftex|ifhtml|ifnothtml)$/) {
            $skipping = pop @skstack;
            next;
-        } elsif ($ended =~ /^(?:example|smallexample|display)$/) {
+        } elsif ($ended =~ /^(?:example|smallexample|verbatim|display)$/) {
            $shift = "";
            $_ = "";        # need a paragraph break
        } elsif ($ended =~ /^(?:itemize|enumerate|(?:multi|[fv])?table)$/) {
@@ -290,7 +290,7 @@ INF: while(<$inf>) {
        $_ = "\n=over 4\n";
    };

-    /^\@((?:small)?example|display)/ and do {
+    /^\@((?:small)?example|verbatim|display)/ and do {
        push @endwstack, $endw;
        $endw = $1;
        $shift = "\t";
@@ -384,7 +384,7 @@ sub postprocess
    # @* is also impossible in .pod; we discard it and any newline that
    # follows it.  Similarly, our macro @gol must be discarded.

-    s/\@anchor{(?:[^\}]*)\}//g;
+    s/\@anchor\{(?:[^\}]*)\}//g;
    s/\(?\@xref\{(?:[^\}]*)\}(?:[^.<]|(?:<[^<>]*>))*\.\)?//g;
    s/\s+\(\@pxref\{(?:[^\}]*)\}\)//g;
    s/;\s+\@pxref\{(?:[^\}]*)\}//g;
@@ -12,17 +12,17 @@ explicitly specified. The following rules are applied:

@itemize
@item
-@code{'} and @code{\} are special characters (respectively used for
+@samp{'} and @samp{\} are special characters (respectively used for
 quoting and escaping). In addition to them, there might be other
 special characters depending on the specific syntax where the escaping
 and quoting are employed.

@item
-A special character is escaped by prefixing it with a '\'.
+A special character is escaped by prefixing it with a @samp{\}.

@item
-All characters enclosed between '' are included literally in the
-parsed string. The quote character @code{'} itself cannot be quoted,
+All characters enclosed between @samp{''} are included literally in the
+parsed string. The quote character @samp{'} itself cannot be quoted,
 so you may need to close the quote and escape it.

@item
@@ -71,7 +71,7 @@ Escaping and quoting can be mixed together:
@end example

@item
-To include a literal @code{\} you can use either escaping or quoting:
+To include a literal @samp{\} you can use either escaping or quoting:
@example
 'c:\foo' can be written as c:\\foo
@end example
@@ -238,6 +238,14 @@ The following abbreviations are recognized:
 480x320
@item qhd
 960x540
+@item 2kdci
+2048x1080
+@item 4kdci
+4096x2160
+@item uhd2160
+3840x2160
+@item uhd4320
+7680x4320
@end table

@anchor{video rate syntax}
@@ -844,7 +852,7 @@ Return 1.0 if @var{x} is +/-INFINITY, 0.0 otherwise.
 Return 1.0 if @var{x} is NAN, 0.0 otherwise.

@item ld(var)
-Allow to load the value of the internal variable with number
+Load the value of the internal variable with number
@var{var}, which was previously stored with st(@var{var}, @var{expr}).
 The function returns the loaded value.

@@ -861,7 +869,7 @@ Return 1 if @var{x} is lesser than or equal to @var{y}, 0 otherwise.
 Return the maximum between @var{x} and @var{y}.

@item min(x, y)
-Return the maximum between @var{x} and @var{y}.
+Return the minimum between @var{x} and @var{y}.

@item mod(x, y)
 Compute the remainder of division of @var{x} by @var{y}.
@@ -912,7 +920,7 @@ Compute the square root of @var{expr}. This is equivalent to
 Compute expression @code{1/(1 + exp(4*x))}.

@item st(var, expr)
-Allow to store the value of the expression @var{expr} in an internal
+Store the value of the expression @var{expr} in an internal
 variable. @var{var} specifies the number of the variable where to
 store the value, and it is a value ranging from 0 to 9. The function
 returns the value stored in the internal variable.
@@ -63,6 +63,7 @@ enum HWAccelID {
    HWACCEL_VDPAU,
    HWACCEL_DXVA2,
    HWACCEL_VDA,
+    HWACCEL_VIDEOTOOLBOX,
 };

 typedef struct HWAccel {
@@ -92,6 +93,8 @@ typedef struct OptionsContext {

    /* input/output options */
    int64_t start_time;
+    int64_t start_time_eof;
+    int seek_timestamp;
    const char *format;

    SpecifierOpt *codec_names;
@@ -121,6 +124,8 @@ typedef struct OptionsContext {
    int        nb_hwaccels;
    SpecifierOpt *hwaccel_devices;
    int        nb_hwaccel_devices;
+    SpecifierOpt *autorotate;
+    int        nb_autorotate;

    /* output options */
    StreamMap *stream_maps;
@@ -226,6 +231,7 @@ typedef struct OutputFilter {

    /* temporary storage until stream maps are processed */
    AVFilterInOut       *out_tmp;
+    enum AVMediaType     type;
 } OutputFilter;

 typedef struct FilterGraph {
@@ -275,6 +281,7 @@ typedef struct InputStream {
    int top_field_first;
    int guess_layout_max;

+    int autorotate;
    int resample_height;
    int resample_width;
    int resample_pix_fmt;
@@ -339,6 +346,7 @@ typedef struct InputFile {
    int64_t ts_offset;
    int64_t last_ts;
    int64_t start_time;   /* user-specified start time in AV_TIME_BASE or AV_NOPTS_VALUE */
+    int seek_timestamp;
    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() */
@@ -394,11 +402,13 @@ typedef struct OutputStream {
    AVFrame *filtered_frame;
    AVFrame *last_frame;
    int last_droped;
+    int last_nb0_frames[3];

    /* video only */
    AVRational frame_rate;
    int force_fps;
    int top_field_first;
+    int rotate_overridden;

    AVRational frame_aspect_ratio;

@@ -422,8 +432,8 @@ typedef struct OutputStream {
    char *filters;         ///< filtergraph associated to the -filter option
    char *filters_script;  ///< filtergraph script associated to the -filter_script option

-    int64_t sws_flags;
    AVDictionary *encoder_opts;
+    AVDictionary *sws_dict;
    AVDictionary *swr_opts;
    AVDictionary *resample_opts;
    AVDictionary *bsf_args;
@@ -448,6 +458,15 @@ typedef struct OutputStream {
    // number of frames/samples sent to the encoder
    uint64_t frames_encoded;
    uint64_t samples_encoded;
+
+    /* packet quality factor */
+    int quality;
+
+    /* packet picture type */
+    int pict_type;
+
+    /* frame encode sum of squared error values */
+    int64_t error[4];
 } OutputStream;

 typedef struct OutputFile {
@@ -502,6 +521,7 @@ extern int frame_bits_per_raw_sample;
 extern AVIOContext *progress_avio;
 extern float max_error_rate;
 extern int vdpau_api_ver;
+extern char *videotoolbox_pixfmt;

 extern const AVIOInterruptCB int_cb;

@@ -529,11 +549,13 @@ int configure_filtergraph(FilterGraph *fg);
 int configure_output_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out);
 int ist_in_filtergraph(FilterGraph *fg, InputStream *ist);
 FilterGraph *init_simple_filtergraph(InputStream *ist, OutputStream *ost);
+int init_complex_filtergraph(FilterGraph *fg);

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

 int vdpau_init(AVCodecContext *s);
 int dxva2_init(AVCodecContext *s);
 int vda_init(AVCodecContext *s);
+int videotoolbox_init(AVCodecContext *s);

 #endif /* FFMPEG_H */
@@ -364,7 +364,7 @@ static int dxva2_alloc(AVCodecContext *s)
    d3dpp.SwapEffect       = D3DSWAPEFFECT_DISCARD;
    d3dpp.Flags            = D3DPRESENTFLAG_VIDEO;

-    hr = IDirect3D9_CreateDevice(ctx->d3d9, adapter, D3DDEVTYPE_HAL, GetShellWindow(),
+    hr = IDirect3D9_CreateDevice(ctx->d3d9, adapter, D3DDEVTYPE_HAL, GetDesktopWindow(),
                                 D3DCREATE_SOFTWARE_VERTEXPROCESSING | D3DCREATE_MULTITHREADED | D3DCREATE_FPU_PRESERVE,
                                 &d3dpp, &ctx->d3d9device);
    if (FAILED(hr)) {
@@ -31,6 +31,7 @@
 #include "libavutil/avstring.h"
 #include "libavutil/bprint.h"
 #include "libavutil/channel_layout.h"
+#include "libavutil/display.h"
 #include "libavutil/opt.h"
 #include "libavutil/pixdesc.h"
 #include "libavutil/pixfmt.h"
@@ -84,7 +85,7 @@ void choose_sample_fmt(AVStream *st, AVCodec *codec)
                break;
        }
        if (*p == -1) {
-            if((codec->capabilities & CODEC_CAP_LOSSLESS) && av_get_sample_fmt_name(st->codec->sample_fmt) > av_get_sample_fmt_name(codec->sample_fmts[0]))
+            if((codec->capabilities & AV_CODEC_CAP_LOSSLESS) && av_get_sample_fmt_name(st->codec->sample_fmt) > av_get_sample_fmt_name(codec->sample_fmts[0]))
                av_log(NULL, AV_LOG_ERROR, "Conversion will not be lossless.\n");
            if(av_get_sample_fmt_name(st->codec->sample_fmt))
            av_log(NULL, AV_LOG_WARNING,
@@ -288,6 +289,45 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
    ist->filters[ist->nb_filters - 1] = fg->inputs[fg->nb_inputs - 1];
 }

+int init_complex_filtergraph(FilterGraph *fg)
+{
+    AVFilterInOut *inputs, *outputs, *cur;
+    AVFilterGraph *graph;
+    int ret = 0;
+
+    /* this graph is only used for determining the kinds of inputs
+     * and outputs we have, and is discarded on exit from this function */
+    graph = avfilter_graph_alloc();
+    if (!graph)
+        return AVERROR(ENOMEM);
+
+    ret = avfilter_graph_parse2(graph, fg->graph_desc, &inputs, &outputs);
+    if (ret < 0)
+        goto fail;
+
+    for (cur = inputs; cur; cur = cur->next)
+        init_input_filter(fg, cur);
+
+    for (cur = outputs; cur;) {
+        GROW_ARRAY(fg->outputs, fg->nb_outputs);
+        fg->outputs[fg->nb_outputs - 1] = av_mallocz(sizeof(*fg->outputs[0]));
+        if (!fg->outputs[fg->nb_outputs - 1])
+            exit_program(1);
+
+        fg->outputs[fg->nb_outputs - 1]->graph   = fg;
+        fg->outputs[fg->nb_outputs - 1]->out_tmp = cur;
+        fg->outputs[fg->nb_outputs - 1]->type    = avfilter_pad_get_type(cur->filter_ctx->output_pads,
+                                                                         cur->pad_idx);
+        cur = cur->next;
+        fg->outputs[fg->nb_outputs - 1]->out_tmp->next = NULL;
+    }
+
+fail:
+    avfilter_inout_free(&inputs);
+    avfilter_graph_free(&graph);
+    return ret;
+}
+
 static int insert_trim(int64_t start_time, int64_t duration,
                       AVFilterContext **last_filter, int *pad_idx,
                       const char *filter_name)
@@ -339,6 +379,28 @@ static int insert_trim(int64_t start_time, int64_t duration,
    return 0;
 }

+static int insert_filter(AVFilterContext **last_filter, int *pad_idx,
+                         const char *filter_name, const char *args)
+{
+    AVFilterGraph *graph = (*last_filter)->graph;
+    AVFilterContext *ctx;
+    int ret;
+
+    ret = avfilter_graph_create_filter(&ctx,
+                                       avfilter_get_by_name(filter_name),
+                                       filter_name, args, NULL, graph);
+    if (ret < 0)
+        return ret;
+
+    ret = avfilter_link(*last_filter, *pad_idx, ctx, 0);
+    if (ret < 0)
+        return ret;
+
+    *last_filter = ctx;
+    *pad_idx     = 0;
+    return 0;
+}
+
 static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out)
 {
    char *pix_fmts;
@@ -361,11 +423,17 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter,
    if (codec->width || codec->height) {
        char args[255];
        AVFilterContext *filter;
+        AVDictionaryEntry *e = NULL;

-        snprintf(args, sizeof(args), "%d:%d:0x%X",
+        snprintf(args, sizeof(args), "%d:%d",
                 codec->width,
-                 codec->height,
-                 (unsigned)ost->sws_flags);
+                 codec->height);
+
+        while ((e = av_dict_get(ost->sws_dict, "", e,
+                                AV_DICT_IGNORE_SUFFIX))) {
+            av_strlcatf(args, sizeof(args), ":%s=%s", e->key, e->value);
+        }
+
        snprintf(name, sizeof(name), "scaler for output stream %d:%d",
                 ost->file_index, ost->index);
        if ((ret = avfilter_graph_create_filter(&filter, avfilter_get_by_name("scale"),
@@ -476,7 +544,7 @@ static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter,
                   av_get_default_channel_layout(ost->audio_channels_mapped));
        for (i = 0; i < ost->audio_channels_mapped; i++)
            if (ost->audio_channels_map[i] != -1)
-                av_bprintf(&pan_buf, ":c%d=c%d", i, ost->audio_channels_map[i]);
+                av_bprintf(&pan_buf, "|c%d=c%d", i, ost->audio_channels_map[i]);

        AUTO_INSERT_FILTER("-map_channel", "pan", pan_buf.str);
        av_bprint_finalize(&pan_buf, NULL);
@@ -579,6 +647,11 @@ int configure_output_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOu
    av_freep(&ofilter->name);
    DESCRIBE_FILTER_LINK(ofilter, out, 0);

+    if (!ofilter->ost) {
+        av_log(NULL, AV_LOG_FATAL, "Filter %s has a unconnected output\n", ofilter->name);
+        exit_program(1);
+    }
+
    switch (avfilter_pad_get_type(out->filter_ctx->output_pads, out->pad_idx)) {
    case AVMEDIA_TYPE_VIDEO: return configure_output_video_filter(fg, ofilter, out);
    case AVMEDIA_TYPE_AUDIO: return configure_output_audio_filter(fg, ofilter, out);
@@ -609,8 +682,8 @@ static int sub2video_prepare(InputStream *ist)
        }
        av_log(avf, AV_LOG_INFO, "sub2video: using %dx%d canvas\n", w, h);
    }
-    ist->sub2video.w = ist->dec_ctx->width  = ist->resample_width  = w;
-    ist->sub2video.h = ist->dec_ctx->height = ist->resample_height = h;
+    ist->sub2video.w = ist->resample_width  = w;
+    ist->sub2video.h = ist->resample_height = h;

    /* rectangles are AV_PIX_FMT_PAL8, but we have no guarantee that the
       palettes for all rectangles are identical or compatible */
@@ -665,7 +738,7 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
             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->dec_ctx->flags&CODEC_FLAG_BITEXACT) ? SWS_BITEXACT:0));
+             SWS_BILINEAR + ((ist->dec_ctx->flags&AV_CODEC_FLAG_BITEXACT) ? SWS_BITEXACT:0));
    if (fr.num && fr.den)
        av_bprintf(&args, ":frame_rate=%d/%d", fr.num, fr.den);
    snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index,
@@ -676,6 +749,27 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
        return ret;
    last_filter = ifilter->filter;

+    if (ist->autorotate) {
+        double theta = get_rotation(ist->st);
+
+        if (fabs(theta - 90) < 1.0) {
+            ret = insert_filter(&last_filter, &pad_idx, "transpose", "clock");
+        } else if (fabs(theta - 180) < 1.0) {
+            ret = insert_filter(&last_filter, &pad_idx, "hflip", NULL);
+            if (ret < 0)
+                return ret;
+            ret = insert_filter(&last_filter, &pad_idx, "vflip", NULL);
+        } else if (fabs(theta - 270) < 1.0) {
+            ret = insert_filter(&last_filter, &pad_idx, "transpose", "cclock");
+        } else if (fabs(theta) > 1.0) {
+            char rotate_buf[64];
+            snprintf(rotate_buf, sizeof(rotate_buf), "%f*PI/180", theta);
+            ret = insert_filter(&last_filter, &pad_idx, "rotate", rotate_buf);
+        }
+        if (ret < 0)
+            return ret;
+    }
+
    if (ist->framerate.num) {
        AVFilterContext *setpts;

@@ -860,7 +954,7 @@ static int configure_input_filter(FilterGraph *fg, InputFilter *ifilter,
 int configure_filtergraph(FilterGraph *fg)
 {
    AVFilterInOut *inputs, *outputs, *cur;
-    int ret, i, init = !fg->graph, simple = !fg->graph_desc;
+    int ret, i, simple = !fg->graph_desc;
    const char *graph_desc = simple ? fg->outputs[0]->ost->avfilter :
                                      fg->graph_desc;

@@ -873,7 +967,13 @@ int configure_filtergraph(FilterGraph *fg)
        char args[512];
        AVDictionaryEntry *e = NULL;

-        snprintf(args, sizeof(args), "flags=0x%X", (unsigned)ost->sws_flags);
+        args[0] = 0;
+        while ((e = av_dict_get(ost->sws_dict, "", e,
+                                AV_DICT_IGNORE_SUFFIX))) {
+            av_strlcatf(args, sizeof(args), "%s=%s:", e->key, e->value);
+        }
+        if (strlen(args))
+            args[strlen(args)-1] = 0;
        fg->graph->scale_sws_opts = av_strdup(args);

        args[0] = 0;
@@ -903,14 +1003,30 @@ int configure_filtergraph(FilterGraph *fg)
        return ret;

    if (simple && (!inputs || inputs->next || !outputs || outputs->next)) {
-        av_log(NULL, AV_LOG_ERROR, "Simple filtergraph '%s' does not have "
-               "exactly one input and output.\n", graph_desc);
+        const char *num_inputs;
+        const char *num_outputs;
+        if (!outputs) {
+            num_outputs = "0";
+        } else if (outputs->next) {
+            num_outputs = ">1";
+        } else {
+            num_outputs = "1";
+        }
+        if (!inputs) {
+            num_inputs = "0";
+        } else if (inputs->next) {
+            num_inputs = ">1";
+        } else {
+            num_inputs = "1";
+        }
+        av_log(NULL, AV_LOG_ERROR, "Simple filtergraph '%s' was expected "
+               "to have exactly 1 input and 1 output."
+               " However, it had %s input(s) and %s output(s)."
+               " Please adjust, or use a complex filtergraph (-filter_complex) instead.\n",
+               graph_desc, num_inputs, num_outputs);
        return AVERROR(EINVAL);
    }

-    for (cur = inputs; !simple && init && cur; cur = cur->next)
-        init_input_filter(fg, cur);
-
    for (cur = inputs, i = 0; cur; cur = cur->next, i++)
        if ((ret = configure_input_filter(fg, fg->inputs[i], cur)) < 0) {
            avfilter_inout_free(&inputs);
@@ -919,27 +1035,12 @@ int configure_filtergraph(FilterGraph *fg)
        }
    avfilter_inout_free(&inputs);

-    if (!init || simple) {
-        /* we already know the mappings between lavfi outputs and output streams,
-         * so we can finish the setup */
-        for (cur = outputs, i = 0; cur; cur = cur->next, i++)
-            configure_output_filter(fg, fg->outputs[i], cur);
-        avfilter_inout_free(&outputs);
+    for (cur = outputs, i = 0; cur; cur = cur->next, i++)
+        configure_output_filter(fg, fg->outputs[i], cur);
+    avfilter_inout_free(&outputs);

-        if ((ret = avfilter_graph_config(fg->graph, NULL)) < 0)
-            return ret;
-    } else {
-        /* wait until output mappings are processed */
-        for (cur = outputs; cur;) {
-            GROW_ARRAY(fg->outputs, fg->nb_outputs);
-            if (!(fg->outputs[fg->nb_outputs - 1] = av_mallocz(sizeof(*fg->outputs[0]))))
-                exit_program(1);
-            fg->outputs[fg->nb_outputs - 1]->graph   = fg;
-            fg->outputs[fg->nb_outputs - 1]->out_tmp = cur;
-            cur = cur->next;
-            fg->outputs[fg->nb_outputs - 1]->out_tmp->next = NULL;
-        }
-    }
+    if ((ret = avfilter_graph_config(fg->graph, NULL)) < 0)
+        return ret;

    fg->reconfiguration = 1;

@@ -947,7 +1048,7 @@ int configure_filtergraph(FilterGraph *fg)
        OutputStream *ost = fg->outputs[i]->ost;
        if (ost &&
            ost->enc->type == AVMEDIA_TYPE_AUDIO &&
-            !(ost->enc->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE))
+            !(ost->enc->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE))
            av_buffersink_set_frame_size(ost->filter->filter,
                                         ost->enc_ctx->frame_size);
    }
@@ -40,6 +40,9 @@
 #include "libavutil/parseutils.h"
 #include "libavutil/pixdesc.h"
 #include "libavutil/pixfmt.h"
+#include "libavutil/time_internal.h"
+
+#define DEFAULT_PASS_LOGFILENAME_PREFIX "ffmpeg2pass"

 #define MATCH_PER_STREAM_OPT(name, type, outvar, fmtctx, st)\
 {\
@@ -71,7 +74,10 @@ const HWAccel hwaccels[] = {
    { "dxva2", dxva2_init, HWACCEL_DXVA2, AV_PIX_FMT_DXVA2_VLD },
 #endif
 #if CONFIG_VDA
-    { "vda",   vda_init,   HWACCEL_VDA,   AV_PIX_FMT_VDA },
+    { "vda",   videotoolbox_init,   HWACCEL_VDA,   AV_PIX_FMT_VDA },
+#endif
+#if CONFIG_VIDEOTOOLBOX
+    { "videotoolbox",   videotoolbox_init,   HWACCEL_VIDEOTOOLBOX,   AV_PIX_FMT_VIDEOTOOLBOX },
 #endif
    { 0 },
 };
@@ -110,6 +116,9 @@ static int no_file_overwrite  = 0;
 static int do_psnr            = 0;
 static int input_sync;
 static int override_ffserver  = 0;
+static int input_stream_potentially_available = 0;
+static int ignore_unknown_streams = 0;
+static int copy_unknown_streams = 0;

 static void uninit_options(OptionsContext *o)
 {
@@ -150,12 +159,25 @@ static void init_options(OptionsContext *o)
    o->stop_time = INT64_MAX;
    o->mux_max_delay  = 0.7;
    o->start_time     = AV_NOPTS_VALUE;
+    o->start_time_eof = AV_NOPTS_VALUE;
    o->recording_time = INT64_MAX;
    o->limit_filesize = UINT64_MAX;
    o->chapters_input_file = INT_MAX;
    o->accurate_seek  = 1;
 }

+static int show_hwaccels(void *optctx, const char *opt, const char *arg)
+{
+    int i;
+
+    printf("Hardware acceleration methods:\n");
+    for (i = 0; i < FF_ARRAY_ELEMS(hwaccels) - 1; i++) {
+        printf("%s\n", hwaccels[i].name);
+    }
+    printf("\n");
+    return 0;
+}
+
 /* return a copy of the input with the stream specifiers removed from the keys */
 static AVDictionary *strip_specifiers(AVDictionary *dict)
 {
@@ -227,6 +249,7 @@ static int opt_map(void *optctx, const char *opt, const char *arg)
    int sync_file_idx = -1, sync_stream_idx = 0;
    char *p, *sync;
    char *map;
+    char *allow_unused;

    if (*arg == '-') {
        negative = 1;
@@ -271,6 +294,8 @@ static int opt_map(void *optctx, const char *opt, const char *arg)
            exit_program(1);
        }
    } else {
+        if (allow_unused = strchr(map, '?'))
+            *allow_unused = 0;
        file_idx = strtol(map, &p, 0);
        if (file_idx >= nb_input_files || file_idx < 0) {
            av_log(NULL, AV_LOG_FATAL, "Invalid input file index: %d.\n", file_idx);
@@ -308,8 +333,13 @@ static int opt_map(void *optctx, const char *opt, const char *arg)
    }

    if (!m) {
-        av_log(NULL, AV_LOG_FATAL, "Stream map '%s' matches no streams.\n", arg);
-        exit_program(1);
+        if (allow_unused) {
+            av_log(NULL, AV_LOG_VERBOSE, "Stream map '%s' matches no streams; ignoring.\n", arg);
+        } else {
+            av_log(NULL, AV_LOG_FATAL, "Stream map '%s' matches no streams.\n"
+                                       "To ignore this, add a trailing '?' to the map.\n", arg);
+            exit_program(1);
+        }
    }

    av_freep(&map);
@@ -601,6 +631,9 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
        ist->ts_scale = 1.0;
        MATCH_PER_STREAM_OPT(ts_scale, dbl, ist->ts_scale, ic, st);

+        ist->autorotate = 1;
+        MATCH_PER_STREAM_OPT(autorotate, i, ist->autorotate, ic, st);
+
        MATCH_PER_STREAM_OPT(codec_tags, str, codec_tag, ic, st);
        if (codec_tag) {
            uint32_t tag = strtol(codec_tag, &next, 0);
@@ -641,9 +674,11 @@ 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 FF_API_EMU_EDGE
            if (av_codec_get_lowres(dec)) {
                dec->flags |= CODEC_FLAG_EMU_EDGE;
            }
+#endif

            ist->resample_height  = ist->dec_ctx->height;
            ist->resample_width   = ist->dec_ctx->width;
@@ -916,14 +951,33 @@ static int open_input_file(OptionsContext *o, const char *filename)
        }
    }

+    if (o->start_time_eof != AV_NOPTS_VALUE) {
+        if (ic->duration>0) {
+            o->start_time = o->start_time_eof + ic->duration;
+        } else
+            av_log(NULL, AV_LOG_WARNING, "Cannot use -sseof, duration of %s not known\n", filename);
+    }
    timestamp = (o->start_time == AV_NOPTS_VALUE) ? 0 : o->start_time;
    /* add the stream start time */
-    if (ic->start_time != AV_NOPTS_VALUE)
+    if (!o->seek_timestamp && ic->start_time != AV_NOPTS_VALUE)
        timestamp += ic->start_time;

    /* if seeking requested, we execute it */
    if (o->start_time != AV_NOPTS_VALUE) {
-        ret = avformat_seek_file(ic, -1, INT64_MIN, timestamp, timestamp, 0);
+        int64_t seek_timestamp = timestamp;
+
+        if (!(ic->iformat->flags & AVFMT_SEEK_TO_PTS)) {
+            int dts_heuristic = 0;
+            for (i=0; i<ic->nb_streams; i++) {
+                AVCodecContext *avctx = ic->streams[i]->codec;
+                if (avctx->has_b_frames)
+                    dts_heuristic = 1;
+            }
+            if (dts_heuristic) {
+                seek_timestamp -= 3*AV_TIME_BASE / 23;
+            }
+        }
+        ret = avformat_seek_file(ic, -1, INT64_MIN, seek_timestamp, seek_timestamp, 0);
        if (ret < 0) {
            av_log(NULL, AV_LOG_WARNING, "%s: could not seek to position %0.3f\n",
                   filename, (double)timestamp / AV_TIME_BASE);
@@ -1008,6 +1062,8 @@ static int open_input_file(OptionsContext *o, const char *filename)
        av_dict_free(&opts[i]);
    av_freep(&opts);

+    input_stream_potentially_available = 1;
+
    return 0;
 }

@@ -1182,12 +1238,13 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e
        uint32_t tag = strtol(codec_tag, &next, 0);
        if (*next)
            tag = AV_RL32(codec_tag);
+        ost->st->codec->codec_tag =
        ost->enc_ctx->codec_tag = tag;
    }

    MATCH_PER_STREAM_OPT(qscale, dbl, qscale, oc, st);
    if (qscale >= 0) {
-        ost->enc_ctx->flags |= CODEC_FLAG_QSCALE;
+        ost->enc_ctx->flags |= AV_CODEC_FLAG_QSCALE;
        ost->enc_ctx->global_quality = FF_QP2LAMBDA * qscale;
    }

@@ -1195,9 +1252,9 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e
    ost->disposition = av_strdup(ost->disposition);

    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
-        ost->enc_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
+        ost->enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;

-    av_opt_get_int(o->g->sws_opts, "sws_flags", 0, &ost->sws_flags);
+    av_dict_copy(&ost->sws_dict, o->g->sws_dict, 0);

    av_dict_copy(&ost->swr_opts, o->g->swr_opts, 0);
    if (ost->enc && av_get_exact_bits_per_sample(ost->enc->id) == 24)
@@ -1417,17 +1474,17 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
        video_enc->rc_override_count = i;

        if (do_psnr)
-            video_enc->flags|= CODEC_FLAG_PSNR;
+            video_enc->flags|= AV_CODEC_FLAG_PSNR;

        /* two pass mode */
        MATCH_PER_STREAM_OPT(pass, i, do_pass, oc, st);
        if (do_pass) {
            if (do_pass & 1) {
-                video_enc->flags |= CODEC_FLAG_PASS1;
+                video_enc->flags |= AV_CODEC_FLAG_PASS1;
                av_dict_set(&ost->encoder_opts, "flags", "+pass1", AV_DICT_APPEND);
            }
            if (do_pass & 2) {
-                video_enc->flags |= CODEC_FLAG_PASS2;
+                video_enc->flags |= AV_CODEC_FLAG_PASS2;
                av_dict_set(&ost->encoder_opts, "flags", "+pass2", AV_DICT_APPEND);
            }
        }
@@ -1437,6 +1494,40 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
            !(ost->logfile_prefix = av_strdup(ost->logfile_prefix)))
            exit_program(1);

+        if (do_pass) {
+            char logfilename[1024];
+            FILE *f;
+
+            snprintf(logfilename, sizeof(logfilename), "%s-%d.log",
+                     ost->logfile_prefix ? ost->logfile_prefix :
+                                           DEFAULT_PASS_LOGFILENAME_PREFIX,
+                     i);
+            if (!strcmp(ost->enc->name, "libx264")) {
+                av_dict_set(&ost->encoder_opts, "stats", logfilename, AV_DICT_DONT_OVERWRITE);
+            } else {
+                if (video_enc->flags & AV_CODEC_FLAG_PASS2) {
+                    char  *logbuffer = read_file(logfilename);
+
+                    if (!logbuffer) {
+                        av_log(NULL, AV_LOG_FATAL, "Error reading log file '%s' for pass-2 encoding\n",
+                               logfilename);
+                        exit_program(1);
+                    }
+                    video_enc->stats_in = logbuffer;
+                }
+                if (video_enc->flags & AV_CODEC_FLAG_PASS1) {
+                    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));
+                        exit_program(1);
+                    }
+                    ost->logfile = f;
+                }
+            }
+        }
+
        MATCH_PER_STREAM_OPT(forced_key_frames, str, ost->forced_keyframes, oc, st);
        if (ost->forced_keyframes)
            ost->forced_keyframes = av_strdup(ost->forced_keyframes);
@@ -1546,6 +1637,19 @@ static OutputStream *new_data_stream(OptionsContext *o, AVFormatContext *oc, int
    return ost;
 }

+static OutputStream *new_unknown_stream(OptionsContext *o, AVFormatContext *oc, int source_index)
+{
+    OutputStream *ost;
+
+    ost = new_output_stream(o, oc, AVMEDIA_TYPE_UNKNOWN, source_index);
+    if (!ost->stream_copy) {
+        av_log(NULL, AV_LOG_FATAL, "Unknown stream encoding not supported yet (only streamcopy)\n");
+        exit_program(1);
+    }
+
+    return ost;
+}
+
 static OutputStream *new_attachment_stream(OptionsContext *o, AVFormatContext *oc, int source_index)
 {
    OutputStream *ost = new_output_stream(o, oc, AVMEDIA_TYPE_ATTACHMENT, source_index);
@@ -1702,8 +1806,7 @@ static void init_output_filter(OutputFilter *ofilter, OptionsContext *o,
 {
    OutputStream *ost;

-    switch (avfilter_pad_get_type(ofilter->out_tmp->filter_ctx->output_pads,
-                                  ofilter->out_tmp->pad_idx)) {
+    switch (ofilter->type) {
    case AVMEDIA_TYPE_VIDEO: ost = new_video_stream(o, oc, -1); break;
    case AVMEDIA_TYPE_AUDIO: ost = new_audio_stream(o, oc, -1); break;
    default:
@@ -1736,13 +1839,21 @@ static void init_output_filter(OutputFilter *ofilter, OptionsContext *o,
        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);
-    }
    avfilter_inout_free(&ofilter->out_tmp);
 }

+static int init_complex_filters(void)
+{
+    int i, ret = 0;
+
+    for (i = 0; i < nb_filtergraphs; i++) {
+        ret = init_complex_filtergraph(filtergraphs[i]);
+        if (ret < 0)
+            return ret;
+    }
+    return 0;
+}
+
 static int configure_complex_filters(void)
 {
    int i, ret = 0;
@@ -1765,10 +1876,6 @@ static int open_output_file(OptionsContext *o, const char *filename)
    AVDictionary *unused_opts = NULL;
    AVDictionaryEntry *e = NULL;

-    if (configure_complex_filters() < 0) {
-        av_log(NULL, AV_LOG_FATAL, "Error configuring filters.\n");
-        exit_program(1);
-    }

    if (o->stop_time != INT64_MAX && o->recording_time != INT64_MAX) {
        o->stop_time = INT64_MAX;
@@ -1823,8 +1930,7 @@ static int open_output_file(OptionsContext *o, const char *filename)
            if (!ofilter->out_tmp || ofilter->out_tmp->name)
                continue;

-            switch (avfilter_pad_get_type(ofilter->out_tmp->filter_ctx->output_pads,
-                                          ofilter->out_tmp->pad_idx)) {
+            switch (ofilter->type) {
            case AVMEDIA_TYPE_VIDEO:    o->video_disable    = 1; break;
            case AVMEDIA_TYPE_AUDIO:    o->audio_disable    = 1; break;
            case AVMEDIA_TYPE_SUBTITLE: o->subtitle_disable = 1; break;
@@ -1883,7 +1989,7 @@ static int open_output_file(OptionsContext *o, const char *filename)
            for (i = 0; i < nb_input_streams; i++) {
                int new_area;
                ist = input_streams[i];
-                new_area = ist->st->codec->width * ist->st->codec->height;
+                new_area = ist->st->codec->width * ist->st->codec->height + 100000000*!!ist->st->codec_info_nb_frames;
                if((qcr!=MKTAG('A', 'P', 'I', 'C')) && (ist->st->disposition & AV_DISPOSITION_ATTACHED_PIC))
                    new_area = 1;
                if (ist->st->codec->codec_type == AVMEDIA_TYPE_VIDEO &&
@@ -1900,12 +2006,14 @@ static int open_output_file(OptionsContext *o, const char *filename)

        /* audio: most channels */
        if (!o->audio_disable && av_guess_codec(oc->oformat, NULL, filename, NULL, AVMEDIA_TYPE_AUDIO) != AV_CODEC_ID_NONE) {
-            int channels = 0, idx = -1;
+            int best_score = 0, idx = -1;
            for (i = 0; i < nb_input_streams; i++) {
+                int score;
                ist = input_streams[i];
+                score = ist->st->codec->channels + 100000000*!!ist->st->codec_info_nb_frames;
                if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
-                    ist->st->codec->channels > channels) {
-                    channels = ist->st->codec->channels;
+                    score > best_score) {
+                    best_score = score;
                    idx = i;
                }
            }
@@ -1992,17 +2100,33 @@ loop_end:
                if(o->    data_disable && ist->st->codec->codec_type == AVMEDIA_TYPE_DATA)
                    continue;

+                ost = NULL;
                switch (ist->st->codec->codec_type) {
                case AVMEDIA_TYPE_VIDEO:      ost = new_video_stream     (o, oc, src_idx); break;
                case AVMEDIA_TYPE_AUDIO:      ost = new_audio_stream     (o, oc, src_idx); break;
                case AVMEDIA_TYPE_SUBTITLE:   ost = new_subtitle_stream  (o, oc, src_idx); break;
                case AVMEDIA_TYPE_DATA:       ost = new_data_stream      (o, oc, src_idx); break;
                case AVMEDIA_TYPE_ATTACHMENT: ost = new_attachment_stream(o, oc, src_idx); break;
+                case AVMEDIA_TYPE_UNKNOWN:
+                    if (copy_unknown_streams) {
+                        ost = new_unknown_stream   (o, oc, src_idx);
+                        break;
+                    }
                default:
-                    av_log(NULL, AV_LOG_FATAL, "Cannot map stream #%d:%d - unsupported type.\n",
+                    av_log(NULL, ignore_unknown_streams ? AV_LOG_WARNING : AV_LOG_FATAL,
+                           "Cannot map stream #%d:%d - unsupported type.\n",
                           map->file_index, map->stream_index);
-                    exit_program(1);
+                    if (!ignore_unknown_streams) {
+                        av_log(NULL, AV_LOG_FATAL,
+                               "If you want unsupported types ignored instead "
+                               "of failing, please use the -ignore_unknown option\n"
+                               "If you want them copied, please use -copy_unknown\n");
+                        exit_program(1);
+                    }
                }
+                if (ost)
+                    ost->sync_ist = input_streams[  input_files[map->sync_file_index]->ist_index
+                                                  + map->sync_stream_index];
            }
        }
    }
@@ -2032,11 +2156,11 @@ loop_end:
        avio_read(pb, attachment, len);

        ost = new_attachment_stream(o, oc, -1);
-        ost->stream_copy               = 0;
+        ost->stream_copy               = 1;
        ost->attachment_filename       = o->attachments[i];
        ost->finished                  = 1;
-        ost->enc_ctx->extradata      = attachment;
-        ost->enc_ctx->extradata_size = len;
+        ost->st->codec->extradata      = attachment;
+        ost->st->codec->extradata_size = len;

        p = strrchr(o->attachments[i], '/');
        av_dict_set(&ost->st->metadata, "filename", (p && *p) ? p + 1 : o->attachments[i], AV_DICT_DONT_OVERWRITE);
@@ -2054,6 +2178,12 @@ loop_end:
                exit_program(1);
    }

+    if (!oc->nb_streams && !(oc->oformat->flags & AVFMT_NOSTREAMS)) {
+        av_dump_format(oc, nb_output_files - 1, oc->filename, 1);
+        av_log(NULL, AV_LOG_ERROR, "Output file #%d does not contain any stream\n", nb_output_files - 1);
+        exit_program(1);
+    }
+
    /* check if all codec options have been used */
    unused_opts = strip_specifiers(o->g->codec_opts);
    for (i = of->ost_index; i < nb_output_streams; i++) {
@@ -2096,6 +2226,17 @@ loop_end:
    }
    av_dict_free(&unused_opts);

+    /* set the encoding/decoding_needed flags */
+    for (i = of->ost_index; i < nb_output_streams; i++) {
+        OutputStream *ost = output_streams[i];
+
+        ost->encoding_needed = !ost->stream_copy;
+        if (ost->encoding_needed && ost->source_index >= 0) {
+            InputStream *ist = input_streams[ost->source_index];
+            ist->decoding_needed |= DECODING_FOR_OST;
+        }
+    }
+
    /* check filename in case of an image number is expected */
    if (oc->oformat->flags & AVFMT_NEEDNUMBER) {
        if (!av_filename_number_test(oc->filename)) {
@@ -2104,6 +2245,12 @@ loop_end:
        }
    }

+    if (!(oc->oformat->flags & AVFMT_NOSTREAMS) && !input_stream_potentially_available) {
+        av_log(NULL, AV_LOG_ERROR,
+               "No input streams but output needs an input stream\n");
+        exit_program(1);
+    }
+
    if (!(oc->oformat->flags & AVFMT_NOFILE)) {
        /* test if it already exists to avoid losing precious files */
        assert_file_overwrite(filename);
@@ -2172,8 +2319,11 @@ loop_end:
                continue;
            ist = input_streams[output_streams[i]->source_index];
            av_dict_copy(&output_streams[i]->st->metadata, ist->st->metadata, AV_DICT_DONT_OVERWRITE);
-            if (!output_streams[i]->stream_copy)
+            if (!output_streams[i]->stream_copy) {
                av_dict_set(&output_streams[i]->st->metadata, "encoder", NULL, 0);
+                if (ist->autorotate)
+                    av_dict_set(&output_streams[i]->st->metadata, "rotate", NULL, 0);
+            }
        }

    /* process manually set metadata */
@@ -2182,6 +2332,7 @@ loop_end:
        char type, *val;
        const char *stream_spec;
        int index = 0, j, ret = 0;
+        char now_time[256];

        val = strchr(o->metadata[i].u.str, '=');
        if (!val) {
@@ -2191,11 +2342,26 @@ loop_end:
        }
        *val++ = 0;

+        if (!strcmp(o->metadata[i].u.str, "creation_time") &&
+            !strcmp(val, "now")) {
+            time_t now = time(0);
+            struct tm *ptm, tmbuf;
+            ptm = localtime_r(&now, &tmbuf);
+            if (ptm) {
+                if (strftime(now_time, sizeof(now_time), "%Y-%m-%d %H:%M:%S", ptm))
+                    val = now_time;
+            }
+        }
+
        parse_meta_type(o->metadata[i].specifier, &type, &index, &stream_spec);
        if (type == 's') {
            for (j = 0; j < oc->nb_streams; j++) {
+                ost = output_streams[nb_output_streams - oc->nb_streams + j];
                if ((ret = check_stream_specifier(oc, oc->streams[j], stream_spec)) > 0) {
                    av_dict_set(&oc->streams[j]->metadata, o->metadata[i].u.str, *val ? val : NULL, 0);
+                    if (!strcmp(o->metadata[i].u.str, "rotate")) {
+                        ost->rotate_overridden = 1;
+                    }
                } else if (ret < 0)
                    exit_program(1);
            }
@@ -2282,9 +2448,9 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
        opt_default(NULL, "g", norm == PAL ? "15" : "18");

        opt_default(NULL, "b:v", "1150000");
-        opt_default(NULL, "maxrate", "1150000");
-        opt_default(NULL, "minrate", "1150000");
-        opt_default(NULL, "bufsize", "327680"); // 40*1024*8;
+        opt_default(NULL, "maxrate:v", "1150000");
+        opt_default(NULL, "minrate:v", "1150000");
+        opt_default(NULL, "bufsize:v", "327680"); // 40*1024*8;

        opt_default(NULL, "b:a", "224000");
        parse_option(o, "ar", "44100", options);
@@ -2311,9 +2477,9 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
        opt_default(NULL, "g", norm == PAL ? "15" : "18");

        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, "maxrate:v", "2516000");
+        opt_default(NULL, "minrate:v", "0"); // 1145000;
+        opt_default(NULL, "bufsize:v", "1835008"); // 224*1024*8;
        opt_default(NULL, "scan_offset", "1");

        opt_default(NULL, "b:a", "224000");
@@ -2333,9 +2499,9 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
        opt_default(NULL, "g", norm == PAL ? "15" : "18");

        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;
+        opt_default(NULL, "maxrate:v", "9000000");
+        opt_default(NULL, "minrate:v", "0"); // 1500000;
+        opt_default(NULL, "bufsize:v", "1835008"); // 224*1024*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
@@ -2379,6 +2545,11 @@ static int opt_vstats(void *optctx, const char *opt, const char *arg)
    time_t today2 = time(NULL);
    struct tm *today = localtime(&today2);

+    if (!today) { // maybe tomorrow
+        av_log(NULL, AV_LOG_FATAL, "Unable to get current time: %s\n", strerror(errno));
+        exit_program(1);
+    }
+
    snprintf(filename, sizeof(filename), "vstats_%02d%02d%02d.log", today->tm_hour, today->tm_min,
             today->tm_sec);
    return opt_vstats_file(NULL, opt, filename);
@@ -2608,6 +2779,9 @@ static int opt_filter_complex(void *optctx, const char *opt, const char *arg)
    filtergraphs[nb_filtergraphs - 1]->graph_desc = av_strdup(arg);
    if (!filtergraphs[nb_filtergraphs - 1]->graph_desc)
        return AVERROR(ENOMEM);
+
+    input_stream_potentially_available = 1;
+
    return 0;
 }

@@ -2622,6 +2796,9 @@ static int opt_filter_complex_script(void *optctx, const char *opt, const char *
        return AVERROR(ENOMEM);
    filtergraphs[nb_filtergraphs - 1]->index      = nb_filtergraphs - 1;
    filtergraphs[nb_filtergraphs - 1]->graph_desc = graph_desc;
+
+    input_stream_potentially_available = 1;
+
    return 0;
 }

@@ -2706,8 +2883,8 @@ enum OptGroup {
 };

 static const OptionGroupDef groups[] = {
-    [GROUP_OUTFILE] = { "output file",  NULL, OPT_OUTPUT },
-    [GROUP_INFILE]  = { "input file",   "i",  OPT_INPUT },
+    [GROUP_OUTFILE] = { "output url",  NULL, OPT_OUTPUT },
+    [GROUP_INFILE]  = { "input url",   "i",  OPT_INPUT },
 };

 static int open_files(OptionGroupList *l, const char *inout,
@@ -2773,6 +2950,13 @@ int ffmpeg_parse_options(int argc, char **argv)
        goto fail;
    }

+    /* create the complex filtergraphs */
+    ret = init_complex_filters();
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_FATAL, "Error initializing complex filters.\n");
+        goto fail;
+    }
+
    /* open output files */
    ret = open_files(&octx.groups[GROUP_OUTFILE], "output", open_output_file);
    if (ret < 0) {
@@ -2780,6 +2964,13 @@ int ffmpeg_parse_options(int argc, char **argv)
        goto fail;
    }

+    /* configure the complex filtergraphs */
+    ret = configure_complex_filters();
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_FATAL, "Error configuring complex filters.\n");
+        goto fail;
+    }
+
 fail:
    uninit_parse_context(&octx);
    if (ret < 0) {
@@ -2817,6 +3008,10 @@ const OptionDef options[] = {
        "overwrite output files" },
    { "n",              OPT_BOOL,                                    {              &no_file_overwrite },
        "never overwrite output files" },
+    { "ignore_unknown", OPT_BOOL,                                    {              &ignore_unknown_streams },
+        "Ignore unknown stream types" },
+    { "copy_unknown",   OPT_BOOL | OPT_EXPERT,                       {              &copy_unknown_streams },
+        "Copy unknown stream types" },
    { "c",              HAS_ARG | OPT_STRING | OPT_SPEC |
                        OPT_INPUT | OPT_OUTPUT,                      { .off       = OFFSET(codec_names) },
        "codec name", "codec" },
@@ -2850,6 +3045,12 @@ 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" },
+    { "sseof",          HAS_ARG | OPT_TIME | OPT_OFFSET |
+                        OPT_INPUT | OPT_OUTPUT,                      { .off = OFFSET(start_time_eof) },
+        "set the start time offset relative to EOF", "time_off" },
+    { "seek_timestamp", HAS_ARG | OPT_INT | OPT_OFFSET |
+                        OPT_INPUT,                                   { .off = OFFSET(seek_timestamp) },
+        "enable/disable seeking by timestamp with -ss" },
    { "accurate_seek",  OPT_BOOL | OPT_OFFSET | OPT_EXPERT |
                        OPT_INPUT,                                   { .off = OFFSET(accurate_seek) },
        "enable/disable accurate seeking with -ss" },
@@ -2859,7 +3060,7 @@ const OptionDef options[] = {
    { "itsscale",       HAS_ARG | OPT_DOUBLE | OPT_SPEC |
                        OPT_EXPERT | OPT_INPUT,                      { .off = OFFSET(ts_scale) },
        "set the input ts scale", "scale" },
-    { "timestamp",      HAS_ARG | OPT_PERFILE,                       { .func_arg = opt_recording_timestamp },
+    { "timestamp",      HAS_ARG | OPT_PERFILE | OPT_OUTPUT,          { .func_arg = opt_recording_timestamp },
        "set the recording timestamp ('now' to set the current time)", "time" },
    { "metadata",       HAS_ARG | OPT_STRING | OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(metadata) },
        "add metadata", "string=string" },
@@ -2884,8 +3085,8 @@ const OptionDef options[] = {
                        OPT_INPUT,                                   { .off = OFFSET(rate_emu) },
        "read input at native frame rate", "" },
    { "target",         HAS_ARG | OPT_PERFILE | OPT_OUTPUT,          { .func_arg = opt_target },
-        "specify target file type (\"vcd\", \"svcd\", \"dvd\","
-        " \"dv\", \"dv50\", \"pal-vcd\", \"ntsc-svcd\", ...)", "type" },
+        "specify target file type (\"vcd\", \"svcd\", \"dvd\", \"dv\" or \"dv50\" "
+        "with optional prefixes \"pal-\", \"ntsc-\" or \"film-\")", "type" },
    { "vsync",          HAS_ARG | OPT_EXPERT,                        { opt_vsync },
        "video sync method", "" },
    { "frame_drop_threshold", HAS_ARG | OPT_FLOAT | OPT_EXPERT,      { &frame_drop_threshold },
@@ -3025,7 +3226,7 @@ const OptionDef options[] = {
                      OPT_INPUT | OPT_OUTPUT,                                    { .off = OFFSET(top_field_first) },
        "top=1/bottom=0/auto=-1 field first", "" },
    { "vtag",         OPT_VIDEO | HAS_ARG | OPT_EXPERT  | OPT_PERFILE |
-                      OPT_OUTPUT,                                                { .func_arg = opt_old2new },
+                      OPT_INPUT | OPT_OUTPUT,                                    { .func_arg = opt_old2new },
        "force video tag/fourcc", "fourcc/tag" },
    { "qphist",       OPT_VIDEO | OPT_BOOL | OPT_EXPERT ,                        { &qp_hist },
        "show QP histogram" },
@@ -3047,10 +3248,18 @@ const OptionDef options[] = {
        "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" },
+        "select a device for HW acceleration", "devicename" },
 #if HAVE_VDPAU_X11
    { "vdpau_api_ver", HAS_ARG | OPT_INT | OPT_EXPERT, { &vdpau_api_ver }, "" },
 #endif
+#if CONFIG_VDA || CONFIG_VIDEOTOOLBOX
+    { "videotoolbox_pixfmt", HAS_ARG | OPT_STRING | OPT_EXPERT, { &videotoolbox_pixfmt}, "" },
+#endif
+    { "hwaccels",         OPT_EXIT,                                              { .func_arg = show_hwaccels },
+        "show available HW acceleration methods" },
+    { "autorotate",       HAS_ARG | OPT_BOOL | OPT_SPEC |
+                          OPT_EXPERT | OPT_INPUT,                                { .off = OFFSET(autorotate) },
+        "automatically insert correct rotate filters" },

    /* audio options */
    { "aframes",        OPT_AUDIO | HAS_ARG  | OPT_PERFILE | OPT_OUTPUT,           { .func_arg = opt_audio_frames },
@@ -1,134 +0,0 @@
-/*
- * 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 "libavcodec/avcodec.h"
-#include "libavcodec/vda.h"
-#include "libavutil/imgutils.h"
-
-#include "ffmpeg.h"
-
-typedef struct VDAContext {
-    AVFrame *tmp_frame;
-} VDAContext;
-
-static int vda_retrieve_data(AVCodecContext *s, AVFrame *frame)
-{
-    InputStream *ist = s->opaque;
-    VDAContext  *vda = ist->hwaccel_ctx;
-    CVPixelBufferRef pixbuf = (CVPixelBufferRef)frame->data[3];
-    OSType pixel_format = CVPixelBufferGetPixelFormatType(pixbuf);
-    CVReturn err;
-    uint8_t *data[4] = { 0 };
-    int linesize[4] = { 0 };
-    int planes, ret, i;
-
-    av_frame_unref(vda->tmp_frame);
-
-    switch (pixel_format) {
-    case kCVPixelFormatType_420YpCbCr8Planar: vda->tmp_frame->format = AV_PIX_FMT_YUV420P; break;
-    case kCVPixelFormatType_422YpCbCr8:       vda->tmp_frame->format = AV_PIX_FMT_UYVY422; break;
-    default:
-        av_log(NULL, AV_LOG_ERROR,
-               "Unsupported pixel format: %u\n", pixel_format);
-        return AVERROR(ENOSYS);
-    }
-
-    vda->tmp_frame->width  = frame->width;
-    vda->tmp_frame->height = frame->height;
-    ret = av_frame_get_buffer(vda->tmp_frame, 32);
-    if (ret < 0)
-        return ret;
-
-    err = CVPixelBufferLockBaseAddress(pixbuf, kCVPixelBufferLock_ReadOnly);
-    if (err != kCVReturnSuccess) {
-        av_log(NULL, AV_LOG_ERROR, "Error locking the pixel buffer.\n");
-        return AVERROR_UNKNOWN;
-    }
-
-    if (CVPixelBufferIsPlanar(pixbuf)) {
-
-        planes = CVPixelBufferGetPlaneCount(pixbuf);
-        for (i = 0; i < planes; i++) {
-            data[i]     = CVPixelBufferGetBaseAddressOfPlane(pixbuf, i);
-            linesize[i] = CVPixelBufferGetBytesPerRowOfPlane(pixbuf, i);
-        }
-    } else {
-        data[0] = CVPixelBufferGetBaseAddress(pixbuf);
-        linesize[0] = CVPixelBufferGetBytesPerRow(pixbuf);
-    }
-
-    av_image_copy(vda->tmp_frame->data, vda->tmp_frame->linesize,
-                  data, linesize, vda->tmp_frame->format,
-                  frame->width, frame->height);
-
-    ret = av_frame_copy_props(vda->tmp_frame, frame);
-    if (ret < 0)
-        return ret;
-
-    av_frame_unref(frame);
-    av_frame_move_ref(frame, vda->tmp_frame);
-
-    return 0;
-}
-
-static void vda_uninit(AVCodecContext *s)
-{
-    InputStream *ist = s->opaque;
-    VDAContext  *vda = ist->hwaccel_ctx;
-
-    ist->hwaccel_uninit        = NULL;
-    ist->hwaccel_retrieve_data = NULL;
-
-    av_frame_free(&vda->tmp_frame);
-
-    av_vda_default_free(s);
-    av_freep(&ist->hwaccel_ctx);
-}
-
-int vda_init(AVCodecContext *s)
-{
-    InputStream *ist = s->opaque;
-    int loglevel = (ist->hwaccel_id == HWACCEL_AUTO) ? AV_LOG_VERBOSE : AV_LOG_ERROR;
-    VDAContext *vda;
-    int ret;
-
-    vda = av_mallocz(sizeof(*vda));
-    if (!vda)
-        return AVERROR(ENOMEM);
-
-    ist->hwaccel_ctx           = vda;
-    ist->hwaccel_uninit        = vda_uninit;
-    ist->hwaccel_retrieve_data = vda_retrieve_data;
-
-    vda->tmp_frame = av_frame_alloc();
-    if (!vda->tmp_frame) {
-        ret = AVERROR(ENOMEM);
-        goto fail;
-    }
-
-    ret = av_vda_default_init(s);
-    if (ret < 0) {
-        av_log(NULL, loglevel, "Error creating VDA decoder.\n");
-        goto fail;
-    }
-
-    return 0;
-fail:
-    vda_uninit(s);
-    return ret;
-}
@@ -289,7 +289,8 @@ do {

        s->hwaccel_context = vdpau_ctx;
    } else
-    if (av_vdpau_bind_context(s, ctx->device, ctx->get_proc_address, 0))
+    if (av_vdpau_bind_context(s, ctx->device, ctx->get_proc_address,
+                              AV_HWACCEL_FLAG_IGNORE_LEVEL))
        goto fail;

    ctx->get_information_string(&vendor);
@@ -0,0 +1,187 @@
+/*
+ * 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 <CoreServices/CoreServices.h>
+
+#include "config.h"
+#include "libavcodec/avcodec.h"
+#if CONFIG_VDA
+#  include "libavcodec/vda.h"
+#endif
+#if CONFIG_VIDEOTOOLBOX
+#  include "libavcodec/videotoolbox.h"
+#endif
+#include "libavutil/imgutils.h"
+#include "ffmpeg.h"
+
+typedef struct VTContext {
+    AVFrame *tmp_frame;
+} VTContext;
+
+char *videotoolbox_pixfmt;
+
+static int videotoolbox_retrieve_data(AVCodecContext *s, AVFrame *frame)
+{
+    InputStream *ist = s->opaque;
+    VTContext  *vt = ist->hwaccel_ctx;
+    CVPixelBufferRef pixbuf = (CVPixelBufferRef)frame->data[3];
+    OSType pixel_format = CVPixelBufferGetPixelFormatType(pixbuf);
+    CVReturn err;
+    uint8_t *data[4] = { 0 };
+    int linesize[4] = { 0 };
+    int planes, ret, i;
+    char codec_str[32];
+
+    av_frame_unref(vt->tmp_frame);
+
+    switch (pixel_format) {
+    case kCVPixelFormatType_420YpCbCr8Planar: vt->tmp_frame->format = AV_PIX_FMT_YUV420P; break;
+    case kCVPixelFormatType_422YpCbCr8:       vt->tmp_frame->format = AV_PIX_FMT_UYVY422; break;
+    case kCVPixelFormatType_32BGRA:           vt->tmp_frame->format = AV_PIX_FMT_BGRA; break;
+#ifdef kCFCoreFoundationVersionNumber10_7
+    case kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange: vt->tmp_frame->format = AV_PIX_FMT_NV12; break;
+#endif
+    default:
+        av_get_codec_tag_string(codec_str, sizeof(codec_str), s->codec_tag);
+        av_log(NULL, AV_LOG_ERROR,
+               "%s: Unsupported pixel format: %s\n", codec_str, videotoolbox_pixfmt);
+        return AVERROR(ENOSYS);
+    }
+
+    vt->tmp_frame->width  = frame->width;
+    vt->tmp_frame->height = frame->height;
+    ret = av_frame_get_buffer(vt->tmp_frame, 32);
+    if (ret < 0)
+        return ret;
+
+    err = CVPixelBufferLockBaseAddress(pixbuf, kCVPixelBufferLock_ReadOnly);
+    if (err != kCVReturnSuccess) {
+        av_log(NULL, AV_LOG_ERROR, "Error locking the pixel buffer.\n");
+        return AVERROR_UNKNOWN;
+    }
+
+    if (CVPixelBufferIsPlanar(pixbuf)) {
+
+        planes = CVPixelBufferGetPlaneCount(pixbuf);
+        for (i = 0; i < planes; i++) {
+            data[i]     = CVPixelBufferGetBaseAddressOfPlane(pixbuf, i);
+            linesize[i] = CVPixelBufferGetBytesPerRowOfPlane(pixbuf, i);
+        }
+    } else {
+        data[0] = CVPixelBufferGetBaseAddress(pixbuf);
+        linesize[0] = CVPixelBufferGetBytesPerRow(pixbuf);
+    }
+
+    av_image_copy(vt->tmp_frame->data, vt->tmp_frame->linesize,
+                  (const uint8_t **)data, linesize, vt->tmp_frame->format,
+                  frame->width, frame->height);
+
+    ret = av_frame_copy_props(vt->tmp_frame, frame);
+    CVPixelBufferUnlockBaseAddress(pixbuf, kCVPixelBufferLock_ReadOnly);
+    if (ret < 0)
+        return ret;
+
+    av_frame_unref(frame);
+    av_frame_move_ref(frame, vt->tmp_frame);
+
+    return 0;
+}
+
+static void videotoolbox_uninit(AVCodecContext *s)
+{
+    InputStream *ist = s->opaque;
+    VTContext  *vt = ist->hwaccel_ctx;
+
+    ist->hwaccel_uninit        = NULL;
+    ist->hwaccel_retrieve_data = NULL;
+
+    av_frame_free(&vt->tmp_frame);
+
+    if (ist->hwaccel_id == HWACCEL_VIDEOTOOLBOX) {
+#if CONFIG_VIDEOTOOLBOX
+        av_videotoolbox_default_free(s);
+#endif
+    } else {
+#if CONFIG_VDA
+        av_vda_default_free(s);
+#endif
+    }
+    av_freep(&ist->hwaccel_ctx);
+}
+
+int videotoolbox_init(AVCodecContext *s)
+{
+    InputStream *ist = s->opaque;
+    int loglevel = (ist->hwaccel_id == HWACCEL_AUTO) ? AV_LOG_VERBOSE : AV_LOG_ERROR;
+    int ret = 0;
+    VTContext *vt;
+
+    vt = av_mallocz(sizeof(*vt));
+    if (!vt)
+        return AVERROR(ENOMEM);
+
+    ist->hwaccel_ctx           = vt;
+    ist->hwaccel_uninit        = videotoolbox_uninit;
+    ist->hwaccel_retrieve_data = videotoolbox_retrieve_data;
+
+    vt->tmp_frame = av_frame_alloc();
+    if (!vt->tmp_frame) {
+        ret = AVERROR(ENOMEM);
+        goto fail;
+    }
+
+    if (ist->hwaccel_id == HWACCEL_VIDEOTOOLBOX) {
+#if CONFIG_VIDEOTOOLBOX
+        if (!videotoolbox_pixfmt) {
+            ret = av_videotoolbox_default_init(s);
+        } else {
+            AVVideotoolboxContext *vtctx = av_videotoolbox_alloc_context();
+            CFStringRef pixfmt_str = CFStringCreateWithCString(kCFAllocatorDefault,
+                                                               videotoolbox_pixfmt,
+                                                               kCFStringEncodingUTF8);
+            vtctx->cv_pix_fmt_type = UTGetOSTypeFromString(pixfmt_str);
+            ret = av_videotoolbox_default_init2(s, vtctx);
+            CFRelease(pixfmt_str);
+        }
+#endif
+    } else {
+#if CONFIG_VDA
+        if (!videotoolbox_pixfmt) {
+            ret = av_vda_default_init(s);
+        } else {
+            AVVDAContext *vdactx = av_vda_alloc_context();
+            CFStringRef pixfmt_str = CFStringCreateWithCString(kCFAllocatorDefault,
+                                                               videotoolbox_pixfmt,
+                                                               kCFStringEncodingUTF8);
+            vdactx->cv_pix_fmt_type = UTGetOSTypeFromString(pixfmt_str);
+            ret = av_vda_default_init2(s, vdactx);
+            CFRelease(pixfmt_str);
+        }
+#endif
+    }
+    if (ret < 0) {
+        av_log(NULL, loglevel,
+               "Error creating %s decoder.\n", ist->hwaccel_id == HWACCEL_VIDEOTOOLBOX ? "Videotoolbox" : "VDA");
+        goto fail;
+    }
+
+    return 0;
+fail:
+    videotoolbox_uninit(s);
+    return ret;
+}
@@ -31,7 +31,7 @@
 #include <stdint.h>

 #include "libavutil/avstring.h"
-#include "libavutil/colorspace.h"
+#include "libavutil/eval.h"
 #include "libavutil/mathematics.h"
 #include "libavutil/pixdesc.h"
 #include "libavutil/imgutils.h"
@@ -65,7 +65,9 @@ const char program_name[] = "ffplay";
 const int program_birth_year = 2003;

 #define MAX_QUEUE_SIZE (15 * 1024 * 1024)
-#define MIN_FRAMES 5
+#define MIN_FRAMES 25
+#define EXTERNAL_CLOCK_MIN_FRAMES 2
+#define EXTERNAL_CLOCK_MAX_FRAMES 10

 /* Minimum SDL audio buffer size, in samples. */
 #define SDL_AUDIO_MIN_BUFFER_SIZE 512
@@ -101,7 +103,7 @@ const int program_birth_year = 2003;

 #define CURSOR_HIDE_DELAY 1000000

-static int64_t sws_flags = SWS_BICUBIC;
+static unsigned sws_flags = SWS_BICUBIC;

 typedef struct MyAVPacketList {
    AVPacket pkt;
@@ -222,6 +224,9 @@ typedef struct VideoState {
    Decoder viddec;
    Decoder subdec;

+    int viddec_width;
+    int viddec_height;
+
    int audio_stream;

    int av_sync_type;
@@ -277,6 +282,7 @@ typedef struct VideoState {
 #if !CONFIG_AVFILTER
    struct SwsContext *img_convert_ctx;
 #endif
+    struct SwsContext *sub_convert_ctx;
    SDL_Rect last_display_rect;
    int eof;

@@ -828,229 +834,50 @@ static void fill_border(int xleft, int ytop, int width, int height, int x, int y
 #define ALPHA_BLEND(a, oldp, newp, s)\
 ((((oldp << s) * (255 - (a))) + (newp * (a))) / (255 << s))

-#define RGBA_IN(r, g, b, a, s)\
-{\
-    unsigned int v = ((const uint32_t *)(s))[0];\
-    a = (v >> 24) & 0xff;\
-    r = (v >> 16) & 0xff;\
-    g = (v >> 8) & 0xff;\
-    b = v & 0xff;\
-}
-
-#define YUVA_IN(y, u, v, a, s, pal)\
-{\
-    unsigned int val = ((const uint32_t *)(pal))[*(const uint8_t*)(s)];\
-    a = (val >> 24) & 0xff;\
-    y = (val >> 16) & 0xff;\
-    u = (val >> 8) & 0xff;\
-    v = val & 0xff;\
-}
-
-#define YUVA_OUT(d, y, u, v, a)\
-{\
-    ((uint32_t *)(d))[0] = (a << 24) | (y << 16) | (u << 8) | v;\
-}


 #define BPP 1

 static void blend_subrect(AVPicture *dst, const AVSubtitleRect *rect, int imgw, int imgh)
 {
-    int wrap, wrap3, width2, skip2;
-    int y, u, v, a, u1, v1, a1, w, h;
+    int x, y, Y, U, V, A;
    uint8_t *lum, *cb, *cr;
-    const uint8_t *p;
-    const uint32_t *pal;
    int dstx, dsty, dstw, dsth;
+    const AVPicture *src = &rect->pict;

    dstw = av_clip(rect->w, 0, imgw);
    dsth = av_clip(rect->h, 0, imgh);
    dstx = av_clip(rect->x, 0, imgw - dstw);
    dsty = av_clip(rect->y, 0, imgh - dsth);
-    lum = dst->data[0] + dsty * dst->linesize[0];
-    cb  = dst->data[1] + (dsty >> 1) * dst->linesize[1];
-    cr  = dst->data[2] + (dsty >> 1) * dst->linesize[2];
+    lum = dst->data[0] + dstx + dsty * dst->linesize[0];
+    cb  = dst->data[1] + dstx/2 + (dsty >> 1) * dst->linesize[1];
+    cr  = dst->data[2] + dstx/2 + (dsty >> 1) * dst->linesize[2];

-    width2 = ((dstw + 1) >> 1) + (dstx & ~dstw & 1);
-    skip2 = dstx >> 1;
-    wrap = dst->linesize[0];
-    wrap3 = rect->pict.linesize[0];
-    p = rect->pict.data[0];
-    pal = (const uint32_t *)rect->pict.data[1];  /* Now in YCrCb! */
-
-    if (dsty & 1) {
-        lum += dstx;
-        cb += skip2;
-        cr += skip2;
-
-        if (dstx & 1) {
-            YUVA_IN(y, u, v, a, p, pal);
-            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
-            cb[0] = ALPHA_BLEND(a >> 2, cb[0], u, 0);
-            cr[0] = ALPHA_BLEND(a >> 2, cr[0], v, 0);
-            cb++;
-            cr++;
-            lum++;
-            p += BPP;
-        }
-        for (w = dstw - (dstx & 1); w >= 2; w -= 2) {
-            YUVA_IN(y, u, v, a, p, pal);
-            u1 = u;
-            v1 = v;
-            a1 = a;
-            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
-
-            YUVA_IN(y, u, v, a, p + BPP, pal);
-            u1 += u;
-            v1 += v;
-            a1 += a;
-            lum[1] = ALPHA_BLEND(a, lum[1], y, 0);
-            cb[0] = ALPHA_BLEND(a1 >> 2, cb[0], u1, 1);
-            cr[0] = ALPHA_BLEND(a1 >> 2, cr[0], v1, 1);
-            cb++;
-            cr++;
-            p += 2 * BPP;
-            lum += 2;
-        }
-        if (w) {
-            YUVA_IN(y, u, v, a, p, pal);
-            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
-            cb[0] = ALPHA_BLEND(a >> 2, cb[0], u, 0);
-            cr[0] = ALPHA_BLEND(a >> 2, cr[0], v, 0);
-            p++;
+    for (y = 0; y<dsth; y++) {
+        for (x = 0; x<dstw; x++) {
+            Y = src->data[0][x + y*src->linesize[0]];
+            A = src->data[3][x + y*src->linesize[3]];
+            lum[0] = ALPHA_BLEND(A, lum[0], Y, 0);
            lum++;
        }
-        p += wrap3 - dstw * BPP;
-        lum += wrap - dstw - dstx;
-        cb += dst->linesize[1] - width2 - skip2;
-        cr += dst->linesize[2] - width2 - skip2;
+        lum += dst->linesize[0] - dstw;
    }
-    for (h = dsth - (dsty & 1); h >= 2; h -= 2) {
-        lum += dstx;
-        cb += skip2;
-        cr += skip2;

-        if (dstx & 1) {
-            YUVA_IN(y, u, v, a, p, pal);
-            u1 = u;
-            v1 = v;
-            a1 = a;
-            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
-            p += wrap3;
-            lum += wrap;
-            YUVA_IN(y, u, v, a, p, pal);
-            u1 += u;
-            v1 += v;
-            a1 += a;
-            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
-            cb[0] = ALPHA_BLEND(a1 >> 2, cb[0], u1, 1);
-            cr[0] = ALPHA_BLEND(a1 >> 2, cr[0], v1, 1);
+    for (y = 0; y<dsth/2; y++) {
+        for (x = 0; x<dstw/2; x++) {
+            U = src->data[1][x + y*src->linesize[1]];
+            V = src->data[2][x + y*src->linesize[2]];
+            A = src->data[3][2*x     +  2*y   *src->linesize[3]]
+              + src->data[3][2*x + 1 +  2*y   *src->linesize[3]]
+              + src->data[3][2*x + 1 + (2*y+1)*src->linesize[3]]
+              + src->data[3][2*x     + (2*y+1)*src->linesize[3]];
+            cb[0] = ALPHA_BLEND(A>>2, cb[0], U, 0);
+            cr[0] = ALPHA_BLEND(A>>2, cr[0], V, 0);
            cb++;
            cr++;
-            p += -wrap3 + BPP;
-            lum += -wrap + 1;
-        }
-        for (w = dstw - (dstx & 1); w >= 2; w -= 2) {
-            YUVA_IN(y, u, v, a, p, pal);
-            u1 = u;
-            v1 = v;
-            a1 = a;
-            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
-
-            YUVA_IN(y, u, v, a, p + BPP, pal);
-            u1 += u;
-            v1 += v;
-            a1 += a;
-            lum[1] = ALPHA_BLEND(a, lum[1], y, 0);
-            p += wrap3;
-            lum += wrap;
-
-            YUVA_IN(y, u, v, a, p, pal);
-            u1 += u;
-            v1 += v;
-            a1 += a;
-            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
-
-            YUVA_IN(y, u, v, a, p + BPP, pal);
-            u1 += u;
-            v1 += v;
-            a1 += a;
-            lum[1] = ALPHA_BLEND(a, lum[1], y, 0);
-
-            cb[0] = ALPHA_BLEND(a1 >> 2, cb[0], u1, 2);
-            cr[0] = ALPHA_BLEND(a1 >> 2, cr[0], v1, 2);
-
-            cb++;
-            cr++;
-            p += -wrap3 + 2 * BPP;
-            lum += -wrap + 2;
-        }
-        if (w) {
-            YUVA_IN(y, u, v, a, p, pal);
-            u1 = u;
-            v1 = v;
-            a1 = a;
-            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
-            p += wrap3;
-            lum += wrap;
-            YUVA_IN(y, u, v, a, p, pal);
-            u1 += u;
-            v1 += v;
-            a1 += a;
-            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
-            cb[0] = ALPHA_BLEND(a1 >> 2, cb[0], u1, 1);
-            cr[0] = ALPHA_BLEND(a1 >> 2, cr[0], v1, 1);
-            cb++;
-            cr++;
-            p += -wrap3 + BPP;
-            lum += -wrap + 1;
-        }
-        p += wrap3 + (wrap3 - dstw * BPP);
-        lum += wrap + (wrap - dstw - dstx);
-        cb += dst->linesize[1] - width2 - skip2;
-        cr += dst->linesize[2] - width2 - skip2;
-    }
-    /* handle odd height */
-    if (h) {
-        lum += dstx;
-        cb += skip2;
-        cr += skip2;
-
-        if (dstx & 1) {
-            YUVA_IN(y, u, v, a, p, pal);
-            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
-            cb[0] = ALPHA_BLEND(a >> 2, cb[0], u, 0);
-            cr[0] = ALPHA_BLEND(a >> 2, cr[0], v, 0);
-            cb++;
-            cr++;
-            lum++;
-            p += BPP;
-        }
-        for (w = dstw - (dstx & 1); w >= 2; w -= 2) {
-            YUVA_IN(y, u, v, a, p, pal);
-            u1 = u;
-            v1 = v;
-            a1 = a;
-            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
-
-            YUVA_IN(y, u, v, a, p + BPP, pal);
-            u1 += u;
-            v1 += v;
-            a1 += a;
-            lum[1] = ALPHA_BLEND(a, lum[1], y, 0);
-            cb[0] = ALPHA_BLEND(a1 >> 2, cb[0], u, 1);
-            cr[0] = ALPHA_BLEND(a1 >> 2, cr[0], v, 1);
-            cb++;
-            cr++;
-            p += 2 * BPP;
-            lum += 2;
-        }
-        if (w) {
-            YUVA_IN(y, u, v, a, p, pal);
-            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
-            cb[0] = ALPHA_BLEND(a >> 2, cb[0], u, 0);
-            cr[0] = ALPHA_BLEND(a >> 2, cr[0], v, 0);
        }
+        cb += dst->linesize[1] - dstw/2;
+        cr += dst->linesize[2] - dstw/2;
    }
 }

@@ -1305,6 +1132,7 @@ static void stream_close(VideoState *is)
 #if !CONFIG_AVFILTER
    sws_freeContext(is->img_convert_ctx);
 #endif
+    sws_freeContext(is->sub_convert_ctx);
    av_free(is);
 }

@@ -1474,11 +1302,11 @@ static double get_master_clock(VideoState *is)
 }

 static void check_external_clock_speed(VideoState *is) {
-   if (is->video_stream >= 0 && is->videoq.nb_packets <= MIN_FRAMES / 2 ||
-       is->audio_stream >= 0 && is->audioq.nb_packets <= MIN_FRAMES / 2) {
+   if (is->video_stream >= 0 && is->videoq.nb_packets <= EXTERNAL_CLOCK_MIN_FRAMES ||
+       is->audio_stream >= 0 && is->audioq.nb_packets <= EXTERNAL_CLOCK_MIN_FRAMES) {
       set_clock_speed(&is->extclk, FFMAX(EXTERNAL_CLOCK_SPEED_MIN, is->extclk.speed - EXTERNAL_CLOCK_SPEED_STEP));
-   } else if ((is->video_stream < 0 || is->videoq.nb_packets > MIN_FRAMES * 2) &&
-              (is->audio_stream < 0 || is->audioq.nb_packets > MIN_FRAMES * 2)) {
+   } else if ((is->video_stream < 0 || is->videoq.nb_packets > EXTERNAL_CLOCK_MAX_FRAMES) &&
+              (is->audio_stream < 0 || is->audioq.nb_packets > EXTERNAL_CLOCK_MAX_FRAMES)) {
       set_clock_speed(&is->extclk, FFMIN(EXTERNAL_CLOCK_SPEED_MAX, is->extclk.speed + EXTERNAL_CLOCK_SPEED_STEP));
   } else {
       double speed = is->extclk.speed;
@@ -1531,7 +1359,7 @@ static void step_to_next_frame(VideoState *is)

 static double compute_target_delay(double delay, VideoState *is)
 {
-    double sync_threshold, diff;
+    double sync_threshold, diff = 0;

    /* update delay to follow master synchronisation source */
    if (get_master_sync_type(is) != AV_SYNC_VIDEO_MASTER) {
@@ -1553,7 +1381,7 @@ static double compute_target_delay(double delay, VideoState *is)
        }
    }

-    av_dlog(NULL, "video: delay=%0.3f A-V=%f\n",
+    av_log(NULL, AV_LOG_TRACE, "video: delay=%0.3f A-V=%f\n",
            delay, -diff);

    return delay;
@@ -1850,7 +1678,18 @@ static int queue_picture(VideoState *is, AVFrame *src_frame, double pts, double
        av_picture_copy(&pict, (AVPicture *)src_frame,
                        src_frame->format, vp->width, vp->height);
 #else
-        av_opt_get_int(sws_opts, "sws_flags", 0, &sws_flags);
+        {
+            AVDictionaryEntry *e = av_dict_get(sws_dict, "sws_flags", NULL, 0);
+            if (e) {
+                const AVClass *class = sws_get_class();
+                const AVOption    *o = av_opt_find(&class, "sws_flags", NULL, 0,
+                                                   AV_OPT_SEARCH_FAKE_OBJ);
+                int ret = av_opt_eval_flags(&class, o, e->value, &sws_flags);
+                if (ret < 0)
+                    exit(1);
+            }
+        }
+
        is->img_convert_ctx = sws_getCachedContext(is->img_convert_ctx,
            vp->width, vp->height, src_frame->format, vp->width, vp->height,
            AV_PIX_FMT_YUV420P, sws_flags, NULL, NULL, NULL);
@@ -1892,6 +1731,9 @@ static int get_video_frame(VideoState *is, AVFrame *frame)

        frame->sample_aspect_ratio = av_guess_sample_aspect_ratio(is->ic, is->video_st, frame);

+        is->viddec_width  = frame->width;
+        is->viddec_height = frame->height;
+
        if (framedrop>0 || (framedrop && get_master_sync_type(is) != AV_SYNC_VIDEO_MASTER)) {
            if (frame->pts != AV_NOPTS_VALUE) {
                double diff = dpts - get_master_clock(is);
@@ -1957,15 +1799,23 @@ fail:
 static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const char *vfilters, AVFrame *frame)
 {
    static const enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_YUV420P, AV_PIX_FMT_NONE };
-    char sws_flags_str[128];
+    char sws_flags_str[512] = "";
    char buffersrc_args[256];
    int ret;
    AVFilterContext *filt_src = NULL, *filt_out = NULL, *last_filter = NULL;
    AVCodecContext *codec = is->video_st->codec;
    AVRational fr = av_guess_frame_rate(is->ic, is->video_st, NULL);
+    AVDictionaryEntry *e = NULL;
+
+    while ((e = av_dict_get(sws_dict, "", e, AV_DICT_IGNORE_SUFFIX))) {
+        if (!strcmp(e->key, "sws_flags")) {
+            av_strlcatf(sws_flags_str, sizeof(sws_flags_str), "%s=%s:", "flags", e->value);
+        } else
+            av_strlcatf(sws_flags_str, sizeof(sws_flags_str), "%s=%s:", e->key, e->value);
+    }
+    if (strlen(sws_flags_str))
+        sws_flags_str[strlen(sws_flags_str)-1] = '\0';

-    av_opt_get_int(sws_opts, "sws_flags", 0, &sws_flags);
-    snprintf(sws_flags_str, sizeof(sws_flags_str), "flags=%"PRId64, sws_flags);
    graph->scale_sws_opts = av_strdup(sws_flags_str);

    snprintf(buffersrc_args, sizeof(buffersrc_args),
@@ -1995,20 +1845,20 @@ static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const c

 /* Note: this macro adds a filter before the lastly added filter, so the
 * processing order of the filters is in reverse */
-#define INSERT_FILT(name, arg) do {                                         \
-    AVFilterContext *filt_ctx;                                              \
-                                                                            \
-    ret = avfilter_graph_create_filter(&filt_ctx,                           \
-                                       avfilter_get_by_name(name),          \
-                                       "ffplay_" name, arg, NULL, graph);   \
-    if (ret < 0)                                                            \
-        goto fail;                                                          \
-                                                                            \
-    ret = avfilter_link(filt_ctx, 0, last_filter, 0);                       \
-    if (ret < 0)                                                            \
-        goto fail;                                                          \
-                                                                            \
-    last_filter = filt_ctx;                                                 \
+#define INSERT_FILT(name, arg) do {                                          \
+    AVFilterContext *filt_ctx;                                               \
+                                                                             \
+    ret = avfilter_graph_create_filter(&filt_ctx,                            \
+                                       avfilter_get_by_name(name),           \
+                                       "ffplay_" name, arg, NULL, graph);    \
+    if (ret < 0)                                                             \
+        goto fail;                                                           \
+                                                                             \
+    ret = avfilter_link(filt_ctx, 0, last_filter, 0);                        \
+    if (ret < 0)                                                             \
+        goto fail;                                                           \
+                                                                             \
+    last_filter = filt_ctx;                                                  \
 } while (0)

    /* SDL YUV code is not handling odd width/height for some driver
@@ -2016,20 +1866,19 @@ static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const c
    INSERT_FILT("crop", "floor(in_w/2)*2:floor(in_h/2)*2");

    if (autorotate) {
-        AVDictionaryEntry *rotate_tag = av_dict_get(is->video_st->metadata, "rotate", NULL, 0);
-        if (rotate_tag && *rotate_tag->value && strcmp(rotate_tag->value, "0")) {
-            if (!strcmp(rotate_tag->value, "90")) {
-                INSERT_FILT("transpose", "clock");
-            } else if (!strcmp(rotate_tag->value, "180")) {
-                INSERT_FILT("hflip", NULL);
-                INSERT_FILT("vflip", NULL);
-            } else if (!strcmp(rotate_tag->value, "270")) {
-                INSERT_FILT("transpose", "cclock");
-            } else {
-                char rotate_buf[64];
-                snprintf(rotate_buf, sizeof(rotate_buf), "%s*PI/180", rotate_tag->value);
-                INSERT_FILT("rotate", rotate_buf);
-            }
+        double theta  = get_rotation(is->video_st);
+
+        if (fabs(theta - 90) < 1.0) {
+            INSERT_FILT("transpose", "clock");
+        } else if (fabs(theta - 180) < 1.0) {
+            INSERT_FILT("hflip", NULL);
+            INSERT_FILT("vflip", NULL);
+        } else if (fabs(theta - 270) < 1.0) {
+            INSERT_FILT("transpose", "cclock");
+        } else if (fabs(theta) > 1.0) {
+            char rotate_buf[64];
+            snprintf(rotate_buf, sizeof(rotate_buf), "%f*PI/180", theta);
+            INSERT_FILT("rotate", rotate_buf);
        }
    }

@@ -2231,10 +2080,19 @@ static int video_thread(void *arg)
    enum AVPixelFormat last_format = -2;
    int last_serial = -1;
    int last_vfilter_idx = 0;
+    if (!graph) {
+        av_frame_free(&frame);
+        return AVERROR(ENOMEM);
+    }
+
 #endif

-    if (!frame)
+    if (!frame) {
+#if CONFIG_AVFILTER
+        avfilter_graph_free(&graph);
+#endif
        return AVERROR(ENOMEM);
+    }

    for (;;) {
        ret = get_video_frame(is, frame);
@@ -2319,8 +2177,7 @@ static int subtitle_thread(void *arg)
    Frame *sp;
    int got_subtitle;
    double pts;
-    int i, j;
-    int r, g, b, y, u, v, a;
+    int i;

    for (;;) {
        if (!(sp = frame_queue_peek_writable(&is->subpq)))
@@ -2339,14 +2196,41 @@ static int subtitle_thread(void *arg)

            for (i = 0; i < sp->sub.num_rects; i++)
            {
-                for (j = 0; j < sp->sub.rects[i]->nb_colors; j++)
-                {
-                    RGBA_IN(r, g, b, a, (uint32_t*)sp->sub.rects[i]->pict.data[1] + j);
-                    y = RGB_TO_Y_CCIR(r, g, b);
-                    u = RGB_TO_U_CCIR(r, g, b, 0);
-                    v = RGB_TO_V_CCIR(r, g, b, 0);
-                    YUVA_OUT((uint32_t*)sp->sub.rects[i]->pict.data[1] + j, y, u, v, a);
+                int in_w = sp->sub.rects[i]->w;
+                int in_h = sp->sub.rects[i]->h;
+                int subw = is->subdec.avctx->width  ? is->subdec.avctx->width  : is->viddec_width;
+                int subh = is->subdec.avctx->height ? is->subdec.avctx->height : is->viddec_height;
+                int out_w = is->viddec_width  ? in_w * is->viddec_width  / subw : in_w;
+                int out_h = is->viddec_height ? in_h * is->viddec_height / subh : in_h;
+                AVPicture newpic;
+
+                //can not use avpicture_alloc as it is not compatible with avsubtitle_free()
+                av_image_fill_linesizes(newpic.linesize, AV_PIX_FMT_YUVA420P, out_w);
+                newpic.data[0] = av_malloc(newpic.linesize[0] * out_h);
+                newpic.data[3] = av_malloc(newpic.linesize[3] * out_h);
+                newpic.data[1] = av_malloc(newpic.linesize[1] * ((out_h+1)/2));
+                newpic.data[2] = av_malloc(newpic.linesize[2] * ((out_h+1)/2));
+
+                is->sub_convert_ctx = sws_getCachedContext(is->sub_convert_ctx,
+                    in_w, in_h, AV_PIX_FMT_PAL8, out_w, out_h,
+                    AV_PIX_FMT_YUVA420P, sws_flags, NULL, NULL, NULL);
+                if (!is->sub_convert_ctx || !newpic.data[0] || !newpic.data[3] ||
+                    !newpic.data[1] || !newpic.data[2]
+                ) {
+                    av_log(NULL, AV_LOG_FATAL, "Cannot initialize the sub conversion context\n");
+                    exit(1);
                }
+                sws_scale(is->sub_convert_ctx,
+                          (void*)sp->sub.rects[i]->pict.data, sp->sub.rects[i]->pict.linesize,
+                          0, in_h, newpic.data, newpic.linesize);
+
+                av_free(sp->sub.rects[i]->pict.data[0]);
+                av_free(sp->sub.rects[i]->pict.data[1]);
+                sp->sub.rects[i]->pict = newpic;
+                sp->sub.rects[i]->w = out_w;
+                sp->sub.rects[i]->h = out_h;
+                sp->sub.rects[i]->x = sp->sub.rects[i]->x * out_w / in_w;
+                sp->sub.rects[i]->y = sp->sub.rects[i]->y * out_h / in_h;
            }

            /* now we can update the picture count */
@@ -2403,9 +2287,9 @@ static int synchronize_audio(VideoState *is, int nb_samples)
                    wanted_nb_samples = nb_samples + (int)(diff * is->audio_src.freq);
                    min_nb_samples = ((nb_samples * (100 - SAMPLE_CORRECTION_PERCENT_MAX) / 100));
                    max_nb_samples = ((nb_samples * (100 + SAMPLE_CORRECTION_PERCENT_MAX) / 100));
-                    wanted_nb_samples = FFMIN(FFMAX(wanted_nb_samples, min_nb_samples), max_nb_samples);
+                    wanted_nb_samples = av_clip(wanted_nb_samples, min_nb_samples, max_nb_samples);
                }
-                av_dlog(NULL, "diff=%f adiff=%f sample_diff=%d apts=%0.3f %f\n",
+                av_log(NULL, AV_LOG_TRACE, "diff=%f adiff=%f sample_diff=%d apts=%0.3f %f\n",
                        diff, avg_diff, wanted_nb_samples - nb_samples,
                        is->audio_clock, is->audio_diff_threshold);
            }
@@ -2439,6 +2323,13 @@ static int audio_decode_frame(VideoState *is)
        return -1;

    do {
+#if defined(_WIN32)
+        while (frame_queue_nb_remaining(&is->sampq) == 0) {
+            if ((av_gettime_relative() - audio_callback_time) > 1000000LL * is->audio_hw_buf_size / is->audio_tgt.bytes_per_sec / 2)
+                return -1;
+            av_usleep (1000);
+        }
+#endif
        if (!(af = frame_queue_peek_readable(&is->sampq)))
            return -1;
        frame_queue_next(&is->sampq);
@@ -2686,10 +2577,15 @@ static int stream_component_open(VideoState *is, int stream_index)
    }
    av_codec_set_lowres(avctx, stream_lowres);

+#if FF_API_EMU_EDGE
    if(stream_lowres) avctx->flags |= CODEC_FLAG_EMU_EDGE;
-    if (fast)   avctx->flags2 |= CODEC_FLAG2_FAST;
-    if(codec->capabilities & CODEC_CAP_DR1)
+#endif
+    if (fast)
+        avctx->flags2 |= AV_CODEC_FLAG2_FAST;
+#if FF_API_EMU_EDGE
+    if(codec->capabilities & AV_CODEC_CAP_DR1)
        avctx->flags |= CODEC_FLAG_EMU_EDGE;
+#endif

    opts = filter_codec_opts(codec_opts, avctx->codec_id, ic, ic->streams[stream_index], codec);
    if (!av_dict_get(opts, "threads", NULL, 0))
@@ -2762,6 +2658,9 @@ static int stream_component_open(VideoState *is, int stream_index)
        is->video_stream = stream_index;
        is->video_st = ic->streams[stream_index];

+        is->viddec_width  = avctx->width;
+        is->viddec_height = avctx->height;
+
        decoder_init(&is->viddec, avctx, &is->videoq, is->continue_read_thread);
        decoder_start(&is->viddec, video_thread, is);
        is->queue_attachments_req = 1;
@@ -2970,7 +2869,7 @@ static int read_thread(void *arg)
        AVStream *st = ic->streams[i];
        enum AVMediaType type = st->codec->codec_type;
        st->discard = AVDISCARD_ALL;
-        if (wanted_stream_spec[type] && st_index[type] == -1)
+        if (type >= 0 && wanted_stream_spec[type] && st_index[type] == -1)
            if (avformat_match_stream_specifier(ic, st, wanted_stream_spec[type]) > 0)
                st_index[type] = i;
    }
@@ -3785,6 +3684,8 @@ int main(int argc, char **argv)
    VideoState *is;
    char dummy_videodriver[] = "SDL_VIDEODRIVER=dummy";

+    init_dynload();
+
    av_log_set_flags(AV_LOG_SKIP_REPEATED);
    parse_loglevel(argc, argv, options);

@@ -33,10 +33,12 @@
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
 #include "libavutil/bprint.h"
+#include "libavutil/display.h"
 #include "libavutil/hash.h"
 #include "libavutil/opt.h"
 #include "libavutil/pixdesc.h"
 #include "libavutil/dict.h"
+#include "libavutil/intreadwrite.h"
 #include "libavutil/libm.h"
 #include "libavutil/parseutils.h"
 #include "libavutil/timecode.h"
@@ -135,6 +137,8 @@ typedef enum {
    SECTION_ID_PACKET,
    SECTION_ID_PACKETS,
    SECTION_ID_PACKETS_AND_FRAMES,
+    SECTION_ID_PACKET_SIDE_DATA_LIST,
+    SECTION_ID_PACKET_SIDE_DATA,
    SECTION_ID_PIXEL_FORMAT,
    SECTION_ID_PIXEL_FORMAT_FLAGS,
    SECTION_ID_PIXEL_FORMAT_COMPONENT,
@@ -153,6 +157,8 @@ typedef enum {
    SECTION_ID_STREAM_DISPOSITION,
    SECTION_ID_STREAMS,
    SECTION_ID_STREAM_TAGS,
+    SECTION_ID_STREAM_SIDE_DATA_LIST,
+    SECTION_ID_STREAM_SIDE_DATA,
    SECTION_ID_SUBTITLE,
 } SectionID;

@@ -172,7 +178,9 @@ static struct section sections[] = {
    [SECTION_ID_LIBRARY_VERSION] =    { SECTION_ID_LIBRARY_VERSION, "library_version", 0, { -1 } },
    [SECTION_ID_PACKETS] =            { SECTION_ID_PACKETS, "packets", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PACKET, -1} },
    [SECTION_ID_PACKETS_AND_FRAMES] = { SECTION_ID_PACKETS_AND_FRAMES, "packets_and_frames", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PACKET, -1} },
-    [SECTION_ID_PACKET] =             { SECTION_ID_PACKET, "packet", 0, { -1 } },
+    [SECTION_ID_PACKET] =             { SECTION_ID_PACKET, "packet", 0, { SECTION_ID_PACKET_SIDE_DATA_LIST, -1 } },
+    [SECTION_ID_PACKET_SIDE_DATA_LIST] ={ SECTION_ID_PACKET_SIDE_DATA_LIST, "side_data_list", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PACKET_SIDE_DATA, -1 } },
+    [SECTION_ID_PACKET_SIDE_DATA] =     { SECTION_ID_PACKET_SIDE_DATA, "side_data", 0, { -1 } },
    [SECTION_ID_PIXEL_FORMATS] =      { SECTION_ID_PIXEL_FORMATS, "pixel_formats", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PIXEL_FORMAT, -1 } },
    [SECTION_ID_PIXEL_FORMAT] =       { SECTION_ID_PIXEL_FORMAT, "pixel_format", 0, { SECTION_ID_PIXEL_FORMAT_FLAGS, SECTION_ID_PIXEL_FORMAT_COMPONENTS, -1 } },
    [SECTION_ID_PIXEL_FORMAT_FLAGS] = { SECTION_ID_PIXEL_FORMAT_FLAGS, "flags", 0, { -1 }, .unique_name = "pixel_format_flags" },
@@ -191,9 +199,11 @@ static struct section sections[] = {
                                          SECTION_ID_PACKETS, SECTION_ID_ERROR, SECTION_ID_PROGRAM_VERSION, SECTION_ID_LIBRARY_VERSIONS,
                                          SECTION_ID_PIXEL_FORMATS, -1} },
    [SECTION_ID_STREAMS] =            { SECTION_ID_STREAMS, "streams", SECTION_FLAG_IS_ARRAY, { SECTION_ID_STREAM, -1 } },
-    [SECTION_ID_STREAM] =             { SECTION_ID_STREAM, "stream", 0, { SECTION_ID_STREAM_DISPOSITION, SECTION_ID_STREAM_TAGS, -1 } },
+    [SECTION_ID_STREAM] =             { SECTION_ID_STREAM, "stream", 0, { SECTION_ID_STREAM_DISPOSITION, SECTION_ID_STREAM_TAGS, SECTION_ID_STREAM_SIDE_DATA_LIST, -1 } },
    [SECTION_ID_STREAM_DISPOSITION] = { SECTION_ID_STREAM_DISPOSITION, "disposition", 0, { -1 }, .unique_name = "stream_disposition" },
    [SECTION_ID_STREAM_TAGS] =        { SECTION_ID_STREAM_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "stream_tags" },
+    [SECTION_ID_STREAM_SIDE_DATA_LIST] ={ SECTION_ID_STREAM_SIDE_DATA_LIST, "side_data_list", SECTION_FLAG_IS_ARRAY, { SECTION_ID_STREAM_SIDE_DATA, -1 } },
+    [SECTION_ID_STREAM_SIDE_DATA] =     { SECTION_ID_STREAM_SIDE_DATA, "side_data", 0, { -1 } },
    [SECTION_ID_SUBTITLE] =           { SECTION_ID_SUBTITLE, "subtitle", 0, { -1 } },
 };

@@ -718,6 +728,32 @@ static void writer_print_data_hash(WriterContext *wctx, const char *name,
    writer_print_string(wctx, name, buf, 0);
 }

+static void writer_print_integers(WriterContext *wctx, const char *name,
+                                  uint8_t *data, int size, const char *format,
+                                  int columns, int bytes, int offset_add)
+{
+    AVBPrint bp;
+    int offset = 0, l, i;
+
+    av_bprint_init(&bp, 0, AV_BPRINT_SIZE_UNLIMITED);
+    av_bprintf(&bp, "\n");
+    while (size) {
+        av_bprintf(&bp, "%08x: ", offset);
+        l = FFMIN(size, columns);
+        for (i = 0; i < l; i++) {
+            if      (bytes == 1) av_bprintf(&bp, format, *data);
+            else if (bytes == 2) av_bprintf(&bp, format, AV_RN16(data));
+            else if (bytes == 4) av_bprintf(&bp, format, AV_RN32(data));
+            data += bytes;
+            size --;
+        }
+        av_bprintf(&bp, "\n");
+        offset += offset_add;
+    }
+    writer_print_string(wctx, name, bp.str, 0);
+    av_bprint_finalize(&bp, NULL);
+}
+
 #define MAX_REGISTERED_WRITERS_NB 64

 static const Writer *registered_writers[MAX_REGISTERED_WRITERS_NB + 1];
@@ -1696,6 +1732,57 @@ static inline int show_tags(WriterContext *w, AVDictionary *tags, int section_id
    return ret;
 }

+
+static void print_color_range(WriterContext *w, enum AVColorRange color_range, const char *fallback)
+{
+    const char *val = av_color_range_name(color_range);
+    if (!val || color_range == AVCOL_RANGE_UNSPECIFIED) {
+        print_str_opt("color_range", fallback);
+    } else {
+        print_str("color_range", val);
+    }
+}
+
+static void print_color_space(WriterContext *w, enum AVColorSpace color_space)
+{
+    const char *val = av_color_space_name(color_space);
+    if (!val || color_space == AVCOL_SPC_UNSPECIFIED) {
+        print_str_opt("color_space", "unknown");
+    } else {
+        print_str("color_space", val);
+    }
+}
+
+static void print_primaries(WriterContext *w, enum AVColorPrimaries color_primaries)
+{
+    const char *val = av_color_primaries_name(color_primaries);
+    if (!val || color_primaries == AVCOL_PRI_UNSPECIFIED) {
+        print_str_opt("color_primaries", "unknown");
+    } else {
+        print_str("color_primaries", val);
+    }
+}
+
+static void print_color_trc(WriterContext *w, enum AVColorTransferCharacteristic color_trc)
+{
+    const char *val = av_color_transfer_name(color_trc);
+    if (!val || color_trc == AVCOL_TRC_UNSPECIFIED) {
+        print_str_opt("color_transfer", "unknown");
+    } else {
+        print_str("color_transfer", val);
+    }
+}
+
+static void print_chroma_location(WriterContext *w, enum AVChromaLocation chroma_location)
+{
+    const char *val = av_chroma_location_name(chroma_location);
+    if (!val || chroma_location == AVCHROMA_LOC_UNSPECIFIED) {
+        print_str_opt("chroma_location", "unspecified");
+    } else {
+        print_str("chroma_location", val);
+    }
+}
+
 static void show_packet(WriterContext *w, AVFormatContext *fmt_ctx, AVPacket *pkt, int packet_idx)
 {
    char val_str[128];
@@ -1723,6 +1810,25 @@ static void show_packet(WriterContext *w, AVFormatContext *fmt_ctx, AVPacket *pk
    if (pkt->pos != -1) print_fmt    ("pos", "%"PRId64, pkt->pos);
    else                print_str_opt("pos", "N/A");
    print_fmt("flags", "%c",      pkt->flags & AV_PKT_FLAG_KEY ? 'K' : '_');
+
+    if (pkt->side_data_elems) {
+        int i;
+        writer_print_section_header(w, SECTION_ID_PACKET_SIDE_DATA_LIST);
+        for (i = 0; i < pkt->side_data_elems; i++) {
+            AVPacketSideData *sd = &pkt->side_data[i];
+            const char *name = av_packet_side_data_name(sd->type);
+            writer_print_section_header(w, SECTION_ID_PACKET_SIDE_DATA);
+            print_str("side_data_type", name ? name : "unknown");
+            print_int("side_data_size", sd->size);
+            if (sd->type == AV_PKT_DATA_DISPLAYMATRIX && sd->size >= 9*4) {
+                writer_print_integers(w, "displaymatrix", sd->data, 9, " %11d", 3, 4, 1);
+                print_int("rotation", av_display_rotation_get((int32_t *)sd->data));
+            }
+            writer_print_section_footer(w);
+        }
+        writer_print_section_footer(w);
+    }
+
    if (do_show_data)
        writer_print_data(w, "data", pkt->data, pkt->size);
    writer_print_data_hash(w, "data_hash", pkt->data, pkt->size);
@@ -1769,6 +1875,7 @@ static void show_frame(WriterContext *w, AVFrame *frame, AVStream *stream,
    s = av_get_media_type_string(stream->codec->codec_type);
    if (s) print_str    ("media_type", s);
    else   print_str_opt("media_type", "unknown");
+    print_int("stream_index",           stream->index);
    print_int("key_frame",              frame->key_frame);
    print_ts  ("pkt_pts",               frame->pkt_pts);
    print_time("pkt_pts_time",          frame->pkt_pts, &stream->time_base);
@@ -1833,6 +1940,11 @@ static void show_frame(WriterContext *w, AVFrame *frame, AVStream *stream,
            name = av_frame_side_data_name(sd->type);
            print_str("side_data_type", name ? name : "unknown");
            print_int("side_data_size", sd->size);
+            if (sd->type == AV_FRAME_DATA_DISPLAYMATRIX && sd->size >= 9*4) {
+                abort();
+                writer_print_integers(w, "displaymatrix", sd->data, 9, " %11d", 3, 4, 1);
+                print_int("rotation", av_display_rotation_get((int32_t *)sd->data));
+            }
            writer_print_section_footer(w);
        }
        writer_print_section_footer(w);
@@ -2094,6 +2206,8 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
        case AVMEDIA_TYPE_VIDEO:
            print_int("width",        dec_ctx->width);
            print_int("height",       dec_ctx->height);
+            print_int("coded_width",  dec_ctx->coded_width);
+            print_int("coded_height", dec_ctx->coded_height);
            print_int("has_b_frames", dec_ctx->has_b_frames);
            sar = av_guess_sample_aspect_ratio(fmt_ctx, stream, NULL);
            if (sar.den) {
@@ -2111,28 +2225,12 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
            if (s) print_str    ("pix_fmt", s);
            else   print_str_opt("pix_fmt", "unknown");
            print_int("level",   dec_ctx->level);
-            if (dec_ctx->color_range != AVCOL_RANGE_UNSPECIFIED)
-                print_str    ("color_range", av_color_range_name(dec_ctx->color_range));
-            else
-                print_str_opt("color_range", "N/A");
-            s = av_get_colorspace_name(dec_ctx->colorspace);
-            if (s) print_str    ("color_space", s);
-            else   print_str_opt("color_space", "unknown");

-            if (dec_ctx->color_trc != AVCOL_TRC_UNSPECIFIED)
-                print_str("color_transfer", av_color_transfer_name(dec_ctx->color_trc));
-            else
-                print_str_opt("color_transfer", av_color_transfer_name(dec_ctx->color_trc));
-
-            if (dec_ctx->color_primaries != AVCOL_PRI_UNSPECIFIED)
-                print_str("color_primaries", av_color_primaries_name(dec_ctx->color_primaries));
-            else
-                print_str_opt("color_primaries", av_color_primaries_name(dec_ctx->color_primaries));
-
-            if (dec_ctx->chroma_sample_location != AVCHROMA_LOC_UNSPECIFIED)
-                print_str("chroma_location", av_chroma_location_name(dec_ctx->chroma_sample_location));
-            else
-                print_str_opt("chroma_location", av_chroma_location_name(dec_ctx->chroma_sample_location));
+            print_color_range(w, dec_ctx->color_range, "N/A");
+            print_color_space(w, dec_ctx->colorspace);
+            print_color_trc(w, dec_ctx->color_trc);
+            print_primaries(w, dec_ctx->color_primaries);
+            print_chroma_location(w, dec_ctx->chroma_sample_location);

            if (dec_ctx->timecode_frame_start >= 0) {
                char tcbuf[AV_TIMECODE_STR_SIZE];
@@ -2239,6 +2337,25 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
    if (do_show_stream_tags)
        ret = show_tags(w, stream->metadata, in_program ? SECTION_ID_PROGRAM_STREAM_TAGS : SECTION_ID_STREAM_TAGS);

+    if (stream->nb_side_data) {
+        int i;
+        writer_print_section_header(w, SECTION_ID_STREAM_SIDE_DATA_LIST);
+        for (i = 0; i < stream->nb_side_data; i++) {
+            AVPacketSideData *sd = &stream->side_data[i];
+            const char *name = av_packet_side_data_name(sd->type);
+
+            writer_print_section_header(w, SECTION_ID_STREAM_SIDE_DATA);
+            print_str("side_data_type", name ? name : "unknown");
+            print_int("side_data_size", sd->size);
+            if (sd->type == AV_PKT_DATA_DISPLAYMATRIX && sd->size >= 9*4) {
+                writer_print_integers(w, "displaymatrix", sd->data, 9, " %11d", 3, 4, 1);
+                print_int("rotation", av_display_rotation_get((int32_t *)sd->data));
+            }
+            writer_print_section_footer(w);
+        }
+        writer_print_section_footer(w);
+    }
+
    writer_print_section_footer(w);
    av_bprint_finalize(&pbuf, NULL);
    fflush(stdout);
@@ -2749,6 +2866,9 @@ static int opt_show_format_entry(void *optctx, const char *opt, const char *arg)
    char *buf = av_asprintf("format=%s", arg);
    int ret;

+    if (!buf)
+        return AVERROR(ENOMEM);
+
    av_log(NULL, AV_LOG_WARNING,
           "Option '%s' is deprecated, use '-show_entries format=%s' instead\n",
           opt, arg);
@@ -3057,6 +3177,8 @@ int main(int argc, char **argv)
    char *w_name = NULL, *w_args = NULL;
    int ret, i;

+    init_dynload();
+
    av_log_set_flags(AV_LOG_SKIP_REPEATED);
    register_exit(ffprobe_cleanup);

@@ -31,7 +31,7 @@
 #include <stdlib.h>
 #include <stdio.h>
 #include "libavformat/avformat.h"
-// FIXME those are internal headers, ffserver _really_ shouldn't use them
+/* FIXME: those are internal headers, ffserver _really_ shouldn't use them */
 #include "libavformat/ffm.h"
 #include "libavformat/network.h"
 #include "libavformat/os_support.h"
@@ -209,6 +209,7 @@ static void close_connection(HTTPContext *c);

 /* HTTP handling */
 static int handle_connection(HTTPContext *c);
+static inline void print_stream_params(AVIOContext *pb, FFServerStream *stream);
 static void compute_status(HTTPContext *c);
 static int open_input_stream(HTTPContext *c, const char *info);
 static int http_parse_request(HTTPContext *c);
@@ -250,7 +251,8 @@ static unsigned int nb_connections;

 static uint64_t current_bandwidth;

-static int64_t cur_time;           // Making this global saves on passing it around everywhere
+/* Making this global saves on passing it around everywhere */
+static int64_t cur_time;

 static AVLFG random_state;

@@ -314,12 +316,12 @@ static char *ctime1(char *buf2, int buf_size)
 static void http_vlog(const char *fmt, va_list vargs)
 {
    static int print_prefix = 1;
+    char buf[32];

    if (!logfile)
        return;

    if (print_prefix) {
-        char buf[32];
        ctime1(buf, sizeof(buf));
        fprintf(logfile, "%s ", buf);
    }
@@ -504,8 +506,7 @@ static void start_multicast(void)
        random1 = av_lfg_get(&random_state);

        /* open the RTP connection */
-        snprintf(session_id, sizeof(session_id), "%08x%08x",
-                 random0, random1);
+        snprintf(session_id, sizeof(session_id), "%08x%08x", random0, random1);

        /* choose a port if none given */
        if (stream->multicast_port == 0) {
@@ -630,9 +631,8 @@ static int http_server(void)
                    poll_entry++;
                } else {
                    /* when ffserver is doing the timing, we work by
-                       looking at which packet needs to be sent every
-                       10 ms */
-                    /* one tick wait XXX: 10 ms assumed */
+                     * looking at which packet needs to be sent every
+                     * 10 ms (one tick wait XXX: 10 ms assumed) */
                    if (delay > 10)
                        delay = 10;
                }
@@ -655,7 +655,7 @@ static int http_server(void)
        }

        /* wait for an event on one connection. We poll at least every
-           second to handle timeouts */
+         * second to handle timeouts */
        do {
            ret = poll(poll_table, poll_entry - poll_table, delay);
            if (ret < 0 && ff_neterrno() != AVERROR(EAGAIN) &&
@@ -703,13 +703,9 @@ static void start_wait_request(HTTPContext *c, int is_rtsp)
    c->buffer_ptr = c->buffer;
    c->buffer_end = c->buffer + c->buffer_size - 1; /* leave room for '\0' */

-    if (is_rtsp) {
-        c->timeout = cur_time + RTSP_REQUEST_TIMEOUT;
-        c->state = RTSPSTATE_WAIT_REQUEST;
-    } else {
-        c->timeout = cur_time + HTTP_REQUEST_TIMEOUT;
-        c->state = HTTPSTATE_WAIT_REQUEST;
-    }
+    c->state = is_rtsp ? RTSPSTATE_WAIT_REQUEST : HTTPSTATE_WAIT_REQUEST;
+    c->timeout = cur_time +
+                 (is_rtsp ? RTSP_REQUEST_TIMEOUT : HTTP_REQUEST_TIMEOUT);
 }

 static void http_send_too_busy_reply(int fd)
@@ -787,7 +783,6 @@ static void close_connection(HTTPContext *c)
    HTTPContext **cp, *c1;
    int i, nb_streams;
    AVFormatContext *ctx;
-    URLContext *h;
    AVStream *st;

    /* remove connection from list */
@@ -832,9 +827,7 @@ static void close_connection(HTTPContext *c)
            av_freep(&ctx->streams[0]);
            av_freep(&ctx);
        }
-        h = c->rtp_handles[i];
-        if (h)
-            ffurl_close(h);
+        ffurl_close(c->rtp_handles[i]);
    }

    ctx = &c->fmt_ctx;
@@ -903,11 +896,11 @@ static int handle_connection(HTTPContext *c)
        if ((ptr >= c->buffer + 2 && !memcmp(ptr-2, "\n\n", 2)) ||
            (ptr >= c->buffer + 4 && !memcmp(ptr-4, "\r\n\r\n", 4))) {
            /* request found : parse it and reply */
-            if (c->state == HTTPSTATE_WAIT_REQUEST) {
+            if (c->state == HTTPSTATE_WAIT_REQUEST)
                ret = http_parse_request(c);
-            } else {
+            else
                ret = rtsp_parse_request(c);
-            }
+
            if (ret < 0)
                return -1;
        } else if (ptr >= c->buffer_end) {
@@ -952,8 +945,8 @@ 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 set the speed). It may be better to verify
-           that we do not rely too much on the kernel queues */
+         * 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))
                return -1;
@@ -967,6 +960,10 @@ static int handle_connection(HTTPContext *c)
        /* close connection if trailer sent */
        if (c->state == HTTPSTATE_SEND_DATA_TRAILER)
            return -1;
+        /* Check if it is a single jpeg frame 123 */
+        if (c->stream->single_frame && c->data_count > c->cur_frame_bytes && c->cur_frame_bytes > 0) {
+            close_connection(c);
+        }
        break;
    case HTTPSTATE_RECEIVE_DATA:
        /* no need to read if no events */
@@ -1162,8 +1159,10 @@ static int modify_current_stream(HTTPContext *c, char *rates)
                break;
        }

-        if (c->switch_feed_streams[i] >= 0 && c->switch_feed_streams[i] != c->feed_streams[i])
+        if (c->switch_feed_streams[i] >= 0 &&
+            c->switch_feed_streams[i] != c->feed_streams[i]) {
            action_required = 1;
+        }
    }

    return action_required;
@@ -1204,6 +1203,10 @@ static FFServerIPAddressACL* parse_dynamic_acl(FFServerStream *stream,
    }

    acl = av_mallocz(sizeof(FFServerIPAddressACL));
+    if (!acl) {
+        fclose(f);
+        return NULL;
+    }

    /* Build ACL */
    while (fgets(line, sizeof(line), f)) {
@@ -1263,17 +1266,17 @@ static int validate_acl(FFServerStream *stream, HTTPContext *c)

    if (stream->dynamic_acl[0]) {
        acl = parse_dynamic_acl(stream, c);
-
        ret = validate_acl_list(acl, c);
-
        free_acl_list(acl);
    }

    return ret;
 }

-/* compute the real filename of a file by matching it without its
-   extensions to all the stream's filenames */
+/**
+ * compute the real filename of a file by matching it without its
+ * extensions to all the stream's filenames
+ */
 static void compute_real_filename(char *filename, int max_size)
 {
    char file1[1024];
@@ -1391,7 +1394,7 @@ static int http_parse_request(HTTPContext *c)
        compute_real_filename(filename, sizeof(filename) - 1);
    }

-    // "redirect" / request to index.html
+    /* "redirect" request to index.html */
    if (!strlen(filename))
        av_strlcpy(filename, "index.html", sizeof(filename) - 1);

@@ -1730,8 +1733,9 @@ static int http_parse_request(HTTPContext *c)
    return 0;
 send_status:
    compute_status(c);
-    c->http_error = 200; /* horrible : we use this value to avoid
-                            going to the send data state */
+    /* horrible: we use this value to avoid
+     * going to the send data state */
+    c->http_error = 200;
    c->state = HTTPSTATE_SEND_HEADER;
    return 0;
 }
@@ -1746,6 +1750,52 @@ static void fmt_bytecount(AVIOContext *pb, int64_t count)
    avio_printf(pb, "%"PRId64"%c", count, *s);
 }

+static inline void print_stream_params(AVIOContext *pb, FFServerStream *stream)
+{
+    int i, stream_no;
+    const char *type = "unknown";
+    char parameters[64];
+    AVStream *st;
+    AVCodec *codec;
+
+    stream_no = stream->nb_streams;
+
+    avio_printf(pb, "<table cellspacing=0 cellpadding=4><tr><th>Stream<th>"
+                    "type<th>kbits/s<th align=left>codec<th align=left>"
+                    "Parameters\n");
+
+    for (i = 0; i < stream_no; i++) {
+        st = stream->streams[i];
+        codec = avcodec_find_encoder(st->codec->codec_id);
+
+        parameters[0] = 0;
+
+        switch(st->codec->codec_type) {
+        case AVMEDIA_TYPE_AUDIO:
+            type = "audio";
+            snprintf(parameters, sizeof(parameters), "%d channel(s), %d Hz",
+                     st->codec->channels, st->codec->sample_rate);
+            break;
+        case AVMEDIA_TYPE_VIDEO:
+            type = "video";
+            snprintf(parameters, sizeof(parameters),
+                     "%dx%d, q=%d-%d, fps=%d", st->codec->width,
+                     st->codec->height, st->codec->qmin, st->codec->qmax,
+                     st->codec->time_base.den / st->codec->time_base.num);
+            break;
+        default:
+            abort();
+        }
+
+        avio_printf(pb, "<tr><td align=right>%d<td>%s<td align=right>%d"
+                        "<td>%s<td>%s\n",
+                    i, type, st->codec->bit_rate/1000,
+                    codec ? codec->name : "", parameters);
+     }
+
+     avio_printf(pb, "</table>\n");
+}
+
 static void compute_status(HTTPContext *c)
 {
    HTTPContext *c1;
@@ -1796,8 +1846,8 @@ static void compute_status(HTTPContext *c)
                strcpy(eosf - 3, ".ram");
            else if (stream->fmt && !strcmp(stream->fmt->name, "rtp")) {
                /* generate a sample RTSP director if
-                   unicast. Generate an SDP redirector if
-                   multicast */
+                 * unicast. Generate an SDP redirector if
+                 * multicast */
                eosf = strrchr(sfilename, '.');
                if (!eosf)
                    eosf = sfilename + strlen(sfilename);
@@ -1886,7 +1936,7 @@ static void compute_status(HTTPContext *c)

        avio_printf(pb, "<h2>Feed %s</h2>", stream->filename);
        if (stream->pid) {
-            avio_printf(pb, "Running as pid %d.\n", stream->pid);
+            avio_printf(pb, "Running as pid %"PRId64".\n", (int64_t) stream->pid);

 #if defined(linux)
            {
@@ -1895,8 +1945,8 @@ static void compute_status(HTTPContext *c)

                /* This is somewhat linux specific I guess */
                snprintf(ps_cmd, sizeof(ps_cmd),
-                         "ps -o \"%%cpu,cputime\" --no-headers %d",
-                         stream->pid);
+                         "ps -o \"%%cpu,cputime\" --no-headers %"PRId64"",
+                         (int64_t) stream->pid);

                 pid_stat = popen(ps_cmd, "r");
                 if (pid_stat) {
@@ -1916,42 +1966,7 @@ static void compute_status(HTTPContext *c)
            avio_printf(pb, "<p>");
        }

-        avio_printf(pb, "<table cellspacing=0 cellpadding=4><tr><th>Stream<th>"
-                        "type<th>kbits/s<th align=left>codec<th align=left>"
-                        "Parameters\n");
-
-        for (i = 0; i < stream->nb_streams; i++) {
-            AVStream *st = stream->streams[i];
-            AVCodec *codec = avcodec_find_encoder(st->codec->codec_id);
-            const char *type = "unknown";
-            char parameters[64];
-
-            parameters[0] = 0;
-
-            switch(st->codec->codec_type) {
-            case AVMEDIA_TYPE_AUDIO:
-                type = "audio";
-                snprintf(parameters, sizeof(parameters), "%d channel(s), %d Hz",
-                         st->codec->channels, st->codec->sample_rate);
-                break;
-            case AVMEDIA_TYPE_VIDEO:
-                type = "video";
-                snprintf(parameters, sizeof(parameters),
-                         "%dx%d, q=%d-%d, fps=%d", st->codec->width,
-                         st->codec->height, st->codec->qmin, st->codec->qmax,
-                         st->codec->time_base.den / st->codec->time_base.num);
-                break;
-            default:
-                abort();
-            }
-
-            avio_printf(pb, "<tr><td align=right>%d<td>%s<td align=right>%d"
-                            "<td>%s<td>%s\n",
-                        i, type, st->codec->bit_rate/1000,
-                        codec ? codec->name : "", parameters);
-        }
-
-        avio_printf(pb, "</table>\n");
+        print_stream_params(pb, stream);
        stream = stream->next;
    }

@@ -2055,12 +2070,19 @@ static int open_input_stream(HTTPContext *c, const char *info)
    ret = avformat_open_input(&s, input_filename, c->stream->ifmt,
                              &c->stream->in_opts);
    if (ret < 0) {
-        http_log("Could not open input '%s': %s\n", input_filename, av_err2str(ret));
+        http_log("Could not open input '%s': %s\n",
+                 input_filename, av_err2str(ret));
        return ret;
    }

    /* set buffer size */
-    if (buf_size > 0) ffio_set_buf_size(s->pb, buf_size);
+    if (buf_size > 0) {
+        ret = ffio_set_buf_size(s->pb, buf_size);
+        if (ret < 0) {
+            http_log("Failed to set buffer size\n");
+            return ret;
+        }
+    }

    s->flags |= AVFMT_FLAG_GENPTS;
    c->fmt_in = s;
@@ -2096,8 +2118,7 @@ static int64_t get_server_clock(HTTPContext *c)
    return (cur_time - c->start_time) * 1000;
 }

-/* return the estimated time at which the current packet must be sent
-   (in us) */
+/* return the estimated time (in us) at which the current packet must be sent */
 static int64_t get_packet_send_clock(HTTPContext *c)
 {
    int bytes_left, bytes_sent, frame_bytes;
@@ -2105,11 +2126,10 @@ static int64_t get_packet_send_clock(HTTPContext *c)
    frame_bytes = c->cur_frame_bytes;
    if (frame_bytes <= 0)
        return c->cur_pts;
-    else {
-        bytes_left = c->buffer_end - c->buffer_ptr;
-        bytes_sent = frame_bytes - bytes_left;
-        return c->cur_pts + (c->cur_frame_duration * bytes_sent) / frame_bytes;
-    }
+
+    bytes_left = c->buffer_end - c->buffer_ptr;
+    bytes_sent = frame_bytes - bytes_left;
+    return c->cur_pts + (c->cur_frame_duration * bytes_sent) / frame_bytes;
 }


@@ -2122,15 +2142,22 @@ static int http_prepare_data(HTTPContext *c)
    switch(c->state) {
    case HTTPSTATE_SEND_DATA_HEADER:
        ctx = avformat_alloc_context();
+        if (!ctx)
+            return AVERROR(ENOMEM);
        c->fmt_ctx = *ctx;
        av_freep(&ctx);
        av_dict_copy(&(c->fmt_ctx.metadata), c->stream->metadata, 0);
-        c->fmt_ctx.streams = av_mallocz_array(c->stream->nb_streams, sizeof(AVStream *));
+        c->fmt_ctx.streams = av_mallocz_array(c->stream->nb_streams,
+                                              sizeof(AVStream *));
+        if (!c->fmt_ctx.streams)
+            return AVERROR(ENOMEM);

        for(i=0;i<c->stream->nb_streams;i++) {
            AVStream *src;
            c->fmt_ctx.streams[i] = av_mallocz(sizeof(AVStream));
-            /* if file or feed, then just take streams from FFServerStream struct */
+
+            /* if file or feed, then just take streams from FFServerStream
+             * struct */
            if (!c->stream->feed ||
                c->stream->feed == c->stream)
                src = c->stream->streams[i];
@@ -2195,23 +2222,23 @@ static int http_prepare_data(HTTPContext *c)
            if (ret < 0) {
                if (c->stream->feed) {
                    /* if coming from feed, it means we reached the end of the
-                       ffm file, so must wait for more data */
+                     * ffm file, so must wait for more data */
                    c->state = HTTPSTATE_WAIT_FEED;
                    return 1; /* state changed */
-                } else if (ret == AVERROR(EAGAIN)) {
+                }
+                if (ret == AVERROR(EAGAIN)) {
                    /* input not ready, come back later */
                    return 0;
+                }
+                if (c->stream->loop) {
+                    avformat_close_input(&c->fmt_in);
+                    if (open_input_stream(c, "") < 0)
+                        goto no_loop;
+                    goto redo;
                } else {
-                    if (c->stream->loop) {
-                        avformat_close_input(&c->fmt_in);
-                        if (open_input_stream(c, "") < 0)
-                            goto no_loop;
-                        goto redo;
-                    } else {
                    no_loop:
                        /* must send trailer now because EOF or error */
                        c->state = HTTPSTATE_SEND_DATA_TRAILER;
-                    }
                }
            } else {
                int source_index = pkt.stream_index;
@@ -2280,10 +2307,11 @@ static int http_prepare_data(HTTPContext *c)
                            max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
                        else
                            max_packet_size = c->rtp_handles[c->packet_stream_index]->max_packet_size;
-                        ret = ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size);
-                    } else {
+                        ret = ffio_open_dyn_packet_buf(&ctx->pb,
+                                                       max_packet_size);
+                    } else
                        ret = avio_open_dyn_buf(&ctx->pb);
-                    }
+
                    if (ret < 0) {
                        /* XXX: potential leak */
                        return -1;
@@ -2292,10 +2320,13 @@ static int http_prepare_data(HTTPContext *c)

                    ctx->pb->seekable = 0;
                    if (pkt.dts != AV_NOPTS_VALUE)
-                        pkt.dts = av_rescale_q(pkt.dts, ist->time_base, ost->time_base);
+                        pkt.dts = av_rescale_q(pkt.dts, ist->time_base,
+                                               ost->time_base);
                    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);
+                        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 ((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));
@@ -2343,7 +2374,8 @@ 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) */
+ * (either UDP or TCP)
+ */
 static int http_send_data(HTTPContext *c)
 {
    int len, ret;
@@ -2424,8 +2456,8 @@ static int http_send_data(HTTPContext *c)
                        rtsp_c->packet_buffer_ptr += len;
                    if (rtsp_c->packet_buffer_ptr < rtsp_c->packet_buffer_end) {
                        /* if we could not send all the data, we will
-                           send it later, so a new state is needed to
-                           "lock" the RTSP TCP connection */
+                         * send it later, so a new state is needed to
+                         * "lock" the RTSP TCP connection */
                        rtsp_c->state = RTSPSTATE_SEND_PACKET;
                        break;
                    } else
@@ -2437,7 +2469,8 @@ static int http_send_data(HTTPContext *c)
                    ffurl_write(c->rtp_handles[c->packet_stream_index],
                                c->buffer_ptr, len);
                    c->buffer_ptr += len;
-                    /* here we continue as we can send several packets per 10 ms slot */
+                    /* here we continue as we can send several packets
+                     * per 10 ms slot */
                }
            } else {
                /* TCP data output */
@@ -2508,9 +2541,8 @@ static int http_start_receive_data(HTTPContext *c)
            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;
        }
+        c->stream->feed_write_index = ret;
    }

    c->stream->feed_write_index = FFMAX(ffm_read_write_index(fd),
@@ -2548,16 +2580,17 @@ static int http_receive_data(HTTPContext *c)
        } else if (c->buffer_ptr - c->buffer >= 2 &&
                   !memcmp(c->buffer_ptr - 1, "\r\n", 2)) {
            c->chunk_size = strtol(c->buffer, 0, 16);
-            if (c->chunk_size == 0) // end of stream
+            if (c->chunk_size <= 0) { // end of stream or invalid chunk size
+                c->chunk_size = 0;
                goto fail;
+            }
            c->buffer_ptr = c->buffer;
            break;
-        } else if (++loop_run > 10) {
+        } else if (++loop_run > 10)
            /* no chunk header, abort */
            goto fail;
-        } else {
+        else
            c->buffer_ptr++;
-        }
    }

    if (c->buffer_end > c->buffer_ptr) {
@@ -2572,6 +2605,7 @@ static int http_receive_data(HTTPContext *c)
            /* end of connection : close it */
            goto fail;
        else {
+            av_assert0(len <= c->chunk_size);
            c->chunk_size -= len;
            c->buffer_ptr += len;
            c->data_count += len;
@@ -2590,7 +2624,7 @@ static int http_receive_data(HTTPContext *c)
    if (c->buffer_ptr >= c->buffer_end) {
        FFServerStream *feed = c->stream;
        /* a packet has been received : write it in the store, except
-           if header */
+         * if header */
        if (c->data_count > FFM_PACKET_SIZE) {
            /* XXX: use llseek or url_seek
             * XXX: Should probably fail? */
@@ -2642,6 +2676,9 @@ static int http_receive_data(HTTPContext *c)

            pb = avio_alloc_context(c->buffer, c->buffer_end - c->buffer,
                                    0, NULL, NULL, NULL, NULL);
+            if (!pb)
+                goto fail;
+
            pb->seekable = 0;

            s->pb = pb;
@@ -2793,10 +2830,10 @@ static int rtsp_parse_request(HTTPContext *c)
 the_end:
    len = avio_close_dyn_buf(c->pb, &c->pb_buffer);
    c->pb = NULL; /* safety */
-    if (len < 0) {
+    if (len < 0)
        /* XXX: cannot do more */
        return -1;
-    }
+
    c->buffer_ptr = c->pb_buffer;
    c->buffer_end = c->pb_buffer + len;
    c->state = RTSPSTATE_SEND_REPLY;
@@ -2815,9 +2852,9 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,
    *pbuffer = NULL;

    avc =  avformat_alloc_context();
-    if (!avc || !rtp_format) {
+    if (!avc || !rtp_format)
        return -1;
-    }
+
    avc->oformat = rtp_format;
    av_dict_set(&avc->metadata, "title",
                entry ? entry->value : "No Title", 0);
@@ -2826,13 +2863,15 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,
        snprintf(avc->filename, 1024, "rtp://%s:%d?multicast=1?ttl=%d",
                 inet_ntoa(stream->multicast_ip),
                 stream->multicast_port, stream->multicast_ttl);
-    } else {
+    } else
        snprintf(avc->filename, 1024, "rtp://0.0.0.0");
-    }

-    if (!(avc->streams = av_malloc_array(avc->nb_streams, sizeof(*avc->streams))))
+    avc->streams = av_malloc_array(avc->nb_streams, sizeof(*avc->streams));
+    if (!avc->streams)
        goto sdp_done;
-    if (!(avs = av_malloc_array(avc->nb_streams, sizeof(*avs))))
+
+    avs = av_malloc_array(avc->nb_streams, sizeof(*avs));
+    if (!avs)
        goto sdp_done;

    for(i = 0; i < stream->nb_streams; i++) {
@@ -2840,6 +2879,8 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,
        avc->streams[i]->codec = stream->streams[i]->codec;
    }
    *pbuffer = av_mallocz(2048);
+    if (!*pbuffer)
+        goto sdp_done;
    av_sdp_create(&avc, 1, *pbuffer, 2048);

 sdp_done:
@@ -2853,7 +2894,7 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,

 static void rtsp_cmd_options(HTTPContext *c, const char *url)
 {
-//    rtsp_reply_header(c, RTSP_STATUS_OK);
+    /* rtsp_reply_header(c, RTSP_STATUS_OK); */
    avio_printf(c->pb, "RTSP/1.0 %d %s\r\n", RTSP_STATUS_OK, "OK");
    avio_printf(c->pb, "CSeq: %d\r\n", c->seq);
    avio_printf(c->pb, "Public: %s\r\n",
@@ -2894,7 +2935,8 @@ static void rtsp_cmd_describe(HTTPContext *c, const char *url)
    /* get the host IP */
    len = sizeof(my_addr);
    getsockname(c->fd, (struct sockaddr *)&my_addr, &len);
-    content_length = prepare_sdp_description(stream, &content, my_addr.sin_addr);
+    content_length = prepare_sdp_description(stream, &content,
+                                             my_addr.sin_addr);
    if (content_length < 0) {
        rtsp_reply_error(c, RTSP_STATUS_INTERNAL);
        return;
@@ -3019,7 +3061,7 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url,
    }

    /* test if stream is OK (test needed because several SETUP needs
-       to be done for a given file) */
+     * to be done for a given file) */
    if (rtp_c->stream != stream) {
        rtsp_reply_error(c, RTSP_STATUS_SERVICE);
        return;
@@ -3080,8 +3122,10 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url,
 }


-/* find an RTP connection by using the session ID. Check consistency
-   with filename */
+/**
+ * find an RTP connection by using the session ID. Check consistency
+ * with filename
+ */
 static HTTPContext *find_rtp_session_with_url(const char *url,
                                              const char *session_id)
 {
@@ -3104,10 +3148,10 @@ static HTTPContext *find_rtp_session_with_url(const char *url,
    for(s=0; s<rtp_c->stream->nb_streams; ++s) {
      snprintf(buf, sizeof(buf), "%s/streamid=%d",
        rtp_c->stream->filename, s);
-      if(!strncmp(path, buf, sizeof(buf))) {
-    // XXX: Should we reply with RTSP_STATUS_ONLY_AGGREGATE if nb_streams>1?
+      if(!strncmp(path, buf, sizeof(buf)))
+        /* XXX: Should we reply with RTSP_STATUS_ONLY_AGGREGATE
+         * if nb_streams>1? */
        return rtp_c;
-      }
    }
    len = strlen(path);
    if (len > 0 && path[len - 1] == '/' &&
@@ -3185,7 +3229,7 @@ static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
    const char *proto_str;

    /* XXX: should output a warning page when coming
-       close to the connection limit */
+     * close to the connection limit */
    if (nb_connections >= config.nb_max_connections)
        goto fail;

@@ -3240,9 +3284,11 @@ static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
    return NULL;
 }

-/* add a new RTP stream in an RTP connection (used in RTSP SETUP
-   command). If RTP/TCP protocol is used, TCP connection 'rtsp_c' is
-   used. */
+/**
+ * add a new RTP stream in an RTP connection (used in RTSP SETUP
+ * command). If RTP/TCP protocol is used, TCP connection 'rtsp_c' is
+ * used.
+ */
 static int rtp_new_av_stream(HTTPContext *c,
                             int stream_index, struct sockaddr_in *dest_addr,
                             HTTPContext *rtsp_c)
@@ -3320,10 +3366,10 @@ static int rtp_new_av_stream(HTTPContext *c,

    /* 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) {
+    if (ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size) < 0)
        /* XXX: close stream */
        goto fail;
-    }
+
    if (avformat_write_header(ctx, NULL) < 0) {
    fail:
        if (h)
@@ -3355,13 +3401,17 @@ static AVStream *add_av_stream1(FFServerStream *stream,
        return NULL;
    if (copy) {
        fst->codec = avcodec_alloc_context3(codec->codec);
+        if (!fst->codec) {
+            av_free(fst);
+            return NULL;
+        }
        avcodec_copy_context(fst->codec, codec);
-    } else {
+    } else
        /* live streams must use the actual feed's codec since it may be
         * updated later to carry extradata needed by them.
         */
        fst->codec = codec;
-    }
+
    fst->priv_data = av_mallocz(sizeof(FeedData));
    fst->index = stream->nb_streams;
    avpriv_set_pts_info(fst, 33, 1, 90000);
@@ -3463,8 +3513,7 @@ static void extract_mpeg4_header(AVFormatContext *infile)
                if (p[0] == 0x00 && p[1] == 0x00 &&
                    p[2] == 0x01 && p[3] == 0xb6) {
                    size = p - pkt.data;
-                    //                    av_hex_dump_log(infile, AV_LOG_DEBUG, pkt.data, size);
-                    st->codec->extradata = av_mallocz(size + FF_INPUT_BUFFER_PADDING_SIZE);
+                    st->codec->extradata = av_mallocz(size + AV_INPUT_BUFFER_PADDING_SIZE);
                    st->codec->extradata_size = size;
                    memcpy(st->codec->extradata, pkt.data, size);
                    break;
@@ -3494,7 +3543,7 @@ static void build_file_streams(void)
            /* open stream */
            if (stream->fmt && !strcmp(stream->fmt->name, "rtp")) {
                /* specific case : if transport stream output to RTP,
-                   we use a raw transport stream reader */
+                 * we use a raw transport stream reader */
                av_dict_set(&stream->in_opts, "mpeg2ts_compute_pcr", "1", 0);
            }

@@ -3516,7 +3565,7 @@ static void build_file_streams(void)
                remove_stream(stream);
            } else {
                /* find all the AVStreams inside and reference them in
-                   'stream' */
+                 * 'stream' */
                if (avformat_find_stream_info(infile, NULL) < 0) {
                    http_log("Could not find codec parameters from '%s'\n",
                             stream->feed_filename);
@@ -3543,15 +3592,17 @@ static void build_feed_streams(void)
    /* gather all streams */
    for(stream = config.first_stream; stream; stream = stream->next) {
        feed = stream->feed;
-        if (feed) {
-            if (stream->is_feed) {
-                for(i=0;i<stream->nb_streams;i++)
-                    stream->feed_streams[i] = i;
-            } else {
-                /* we handle a stream coming from a feed */
-                for(i=0;i<stream->nb_streams;i++)
-                    stream->feed_streams[i] = add_av_stream(feed, stream->streams[i]);
-            }
+        if (!feed)
+            continue;
+
+        if (stream->is_feed) {
+            for(i=0;i<stream->nb_streams;i++)
+                stream->feed_streams[i] = i;
+        } else {
+            /* we handle a stream coming from a feed */
+            for(i=0;i<stream->nb_streams;i++)
+                stream->feed_streams[i] = add_av_stream(feed,
+                                                        stream->streams[i]);
        }
    }

@@ -3566,7 +3617,12 @@ static void build_feed_streams(void)

            if (avformat_open_input(&s, feed->feed_filename, NULL, NULL) >= 0) {
                /* set buffer size */
-                ffio_set_buf_size(s->pb, FFM_PACKET_SIZE);
+                int ret = ffio_set_buf_size(s->pb, FFM_PACKET_SIZE);
+                if (ret < 0) {
+                    http_log("Failed to set buffer size\n");
+                    exit(1);
+                }
+
                /* Now see if it matches */
                if (s->nb_streams == feed->nb_streams) {
                    matches = 1;
@@ -3637,6 +3693,11 @@ static void build_feed_streams(void)
        if (avio_check(feed->feed_filename, AVIO_FLAG_WRITE) <= 0) {
            AVFormatContext *s = avformat_alloc_context();

+            if (!s) {
+                http_log("Failed to allocate context\n");
+                exit(1);
+            }
+
            if (feed->readonly) {
                http_log("Unable to create feed file '%s' as it is marked readonly\n",
                    feed->feed_filename);
@@ -3708,24 +3769,24 @@ static void compute_bandwidth(void)
 static void handle_child_exit(int sig)
 {
    pid_t pid;
-    int status;
+    int status, uptime;

    while ((pid = waitpid(-1, &status, WNOHANG)) > 0) {
        FFServerStream *feed;

        for (feed = config.first_feed; feed; feed = feed->next) {
-            if (feed->pid == pid) {
-                int uptime = time(0) - feed->pid_start;
+            if (feed->pid != pid)
+                continue;

-                feed->pid = 0;
-                fprintf(stderr,
-                        "%s: Pid %d exited with status %d after %d seconds\n",
-                        feed->filename, pid, status, uptime);
+            uptime = time(0) - feed->pid_start;
+            feed->pid = 0;
+            fprintf(stderr,
+                    "%s: Pid %"PRId64" exited with status %d after %d seconds\n",
+                    feed->filename, (int64_t) pid, status, uptime);

-                if (uptime < 30)
-                    /* Turn off any more restarts */
-                    ffserver_free_child_args(&feed->child_argv);
-            }
+            if (uptime < 30)
+                /* Turn off any more restarts */
+                ffserver_free_child_args(&feed->child_argv);
        }
    }

@@ -3759,6 +3820,8 @@ int main(int argc, char **argv)
    struct sigaction sigact = { { 0 } };
    int ret = 0;

+    init_dynload();
+
    config.filename = av_strdup("/etc/ffserver.conf");

    parse_loglevel(argc, argv, options);
@@ -230,9 +230,9 @@ static void add_codec(FFServerStream *stream, AVCodecContext *av,
    /* compute default parameters */
    switch(av->codec_type) {
    case AVMEDIA_TYPE_AUDIO:
-        if (!av_dict_get(recommended, "ab", NULL, 0)) {
+        if (!av_dict_get(recommended, "b", NULL, 0)) {
            av->bit_rate = 64000;
-            av_dict_set_int(&recommended, "ab", av->bit_rate, 0);
+            av_dict_set_int(&recommended, "b", av->bit_rate, 0);
            WARNING("Setting default value for audio bit rate = %d. "
                    "Use NoDefaults to disable it.\n",
                    av->bit_rate);
@@ -862,8 +862,10 @@ static int ffserver_parse_config_stream(FFServerConfig *config, const char *cmd,
        } else {
            stream->stream_type = STREAM_TYPE_LIVE;
            /* JPEG cannot be used here, so use single frame MJPEG */
-            if (!strcmp(arg, "jpeg"))
-                strcpy(arg, "mjpeg");
+            if (!strcmp(arg, "jpeg")) {
+                strcpy(arg, "singlejpeg");
+                stream->single_frame=1;
+            }
            stream->fmt = ffserver_guess_format(arg, NULL, NULL);
            if (!stream->fmt)
                ERROR("Unknown Format: '%s'\n", arg);
@@ -921,7 +923,7 @@ static int ffserver_parse_config_stream(FFServerConfig *config, const char *cmd,
        ffserver_get_arg(arg, sizeof(arg), p);
        ffserver_set_float_param(&f, arg, 1000, -FLT_MAX, FLT_MAX, config,
                "Invalid %s: '%s'\n", cmd, arg);
-        if (ffserver_save_avoption_int("ab", (int64_t)lrintf(f),
+        if (ffserver_save_avoption_int("b", (int64_t)lrintf(f),
                                       AV_OPT_FLAG_AUDIO_PARAM, config) < 0)
            goto nomem;
    } else if (!av_strcasecmp(cmd, "AudioChannels")) {
@@ -79,6 +79,7 @@ typedef struct FFServerStream {
    int multicast_port;           /* first port used for multicast */
    int multicast_ttl;
    int loop;                     /* if true, send the stream in loops (only meaningful if file) */
+    char single_frame;            /* only single frame */

    /* feed specific */
    int feed_opened;              /* true if someone is writing to the feed */
@@ -38,15 +38,15 @@ static av_cold int zero12v_decode_init(AVCodecContext *avctx)
 static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
                                int *got_frame, AVPacket *avpkt)
 {
-    int line = 0, ret;
+    int line, ret;
    const int width = avctx->width;
    AVFrame *pic = data;
    uint16_t *y, *u, *v;
    const uint8_t *line_end, *src = avpkt->data;
    int stride = avctx->width * 8 / 3;

-    if (width == 1) {
-        av_log(avctx, AV_LOG_ERROR, "Width 1 not supported.\n");
+    if (width <= 1 || avctx->height <= 0) {
+        av_log(avctx, AV_LOG_ERROR, "Dimensions %dx%d not supported.\n", width, avctx->height);
        return AVERROR_INVALIDDATA;
    }

@@ -67,45 +67,45 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
    pic->pict_type = AV_PICTURE_TYPE_I;
    pic->key_frame = 1;

-    y = (uint16_t *)pic->data[0];
-    u = (uint16_t *)pic->data[1];
-    v = (uint16_t *)pic->data[2];
    line_end = avpkt->data + stride;
+    for (line = 0; line < avctx->height; line++) {
+        uint16_t y_temp[6] = {0x8000, 0x8000, 0x8000, 0x8000, 0x8000, 0x8000};
+        uint16_t u_temp[3] = {0x8000, 0x8000, 0x8000};
+        uint16_t v_temp[3] = {0x8000, 0x8000, 0x8000};
+        int x;
+        y = (uint16_t *)(pic->data[0] + line * pic->linesize[0]);
+        u = (uint16_t *)(pic->data[1] + line * pic->linesize[1]);
+        v = (uint16_t *)(pic->data[2] + line * pic->linesize[2]);

-    while (line++ < avctx->height) {
-        while (1) {
-            uint32_t t = AV_RL32(src);
+        for (x = 0; x < width; x += 6) {
+            uint32_t t;
+
+            if (width - x < 6 || line_end - src < 16) {
+                y = y_temp;
+                u = u_temp;
+                v = v_temp;
+            }
+
+            if (line_end - src < 4)
+                break;
+
+            t = AV_RL32(src);
            src += 4;
            *u++ = t <<  6 & 0xFFC0;
            *y++ = t >>  4 & 0xFFC0;
            *v++ = t >> 14 & 0xFFC0;

-            if (src >= line_end - 1) {
-                *y = 0x80;
-                src++;
-                line_end += stride;
-                y = (uint16_t *)(pic->data[0] + line * pic->linesize[0]);
-                u = (uint16_t *)(pic->data[1] + line * pic->linesize[1]);
-                v = (uint16_t *)(pic->data[2] + line * pic->linesize[2]);
+            if (line_end - src < 4)
                break;
-            }

            t = AV_RL32(src);
            src += 4;
            *y++ = t <<  6 & 0xFFC0;
            *u++ = t >>  4 & 0xFFC0;
            *y++ = t >> 14 & 0xFFC0;
-            if (src >= line_end - 2) {
-                if (!(width & 1)) {
-                    *y = 0x80;
-                    src += 2;
-                }
-                line_end += stride;
-                y = (uint16_t *)(pic->data[0] + line * pic->linesize[0]);
-                u = (uint16_t *)(pic->data[1] + line * pic->linesize[1]);
-                v = (uint16_t *)(pic->data[2] + line * pic->linesize[2]);
+
+            if (line_end - src < 4)
                break;
-            }

            t = AV_RL32(src);
            src += 4;
@@ -113,15 +113,8 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
            *y++ = t >>  4 & 0xFFC0;
            *u++ = t >> 14 & 0xFFC0;

-            if (src >= line_end - 1) {
-                *y = 0x80;
-                src++;
-                line_end += stride;
-                y = (uint16_t *)(pic->data[0] + line * pic->linesize[0]);
-                u = (uint16_t *)(pic->data[1] + line * pic->linesize[1]);
-                v = (uint16_t *)(pic->data[2] + line * pic->linesize[2]);
+            if (line_end - src < 4)
                break;
-            }

            t = AV_RL32(src);
            src += 4;
@@ -129,18 +122,21 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
            *v++ = t >>  4 & 0xFFC0;
            *y++ = t >> 14 & 0xFFC0;

-            if (src >= line_end - 2) {
-                if (width & 1) {
-                    *y = 0x80;
-                    src += 2;
-                }
-                line_end += stride;
-                y = (uint16_t *)(pic->data[0] + line * pic->linesize[0]);
-                u = (uint16_t *)(pic->data[1] + line * pic->linesize[1]);
-                v = (uint16_t *)(pic->data[2] + line * pic->linesize[2]);
+            if (width - x < 6)
                break;
-            }
        }
+
+        if (x < width) {
+            y = x   + (uint16_t *)(pic->data[0] + line * pic->linesize[0]);
+            u = x/2 + (uint16_t *)(pic->data[1] + line * pic->linesize[1]);
+            v = x/2 + (uint16_t *)(pic->data[2] + line * pic->linesize[2]);
+            memcpy(y, y_temp, sizeof(*y) * (width - x));
+            memcpy(u, u_temp, sizeof(*u) * (width - x + 1) / 2);
+            memcpy(v, v_temp, sizeof(*v) * (width - x + 1) / 2);
+        }
+
+        line_end += stride;
+        src = line_end - stride;
    }

    *got_frame = 1;
@@ -155,5 +151,5 @@ AVCodec ff_zero12v_decoder = {
    .id             = AV_CODEC_ID_012V,
    .init           = zero12v_decode_init,
    .decode         = zero12v_decode_frame,
-    .capabilities   = CODEC_CAP_DR1,
+    .capabilities   = AV_CODEC_CAP_DR1,
 };
@@ -559,7 +559,7 @@ static inline void idct_put(FourXContext *f, int x, int y)
        idct(block[i]);
    }

-    if (!(f->avctx->flags & CODEC_FLAG_GRAY)) {
+    if (!(f->avctx->flags & AV_CODEC_FLAG_GRAY)) {
        for (i = 4; i < 6; i++)
            idct(block[i]);
    }
@@ -883,11 +883,11 @@ static int decode_frame(AVCodecContext *avctx, void *data,
        }
        cfrm = &f->cfrm[i];

-        if (data_size > UINT_MAX -  cfrm->size - FF_INPUT_BUFFER_PADDING_SIZE)
+        if (data_size > UINT_MAX -  cfrm->size - AV_INPUT_BUFFER_PADDING_SIZE)
            return AVERROR_INVALIDDATA;

        cfrm->data = av_fast_realloc(cfrm->data, &cfrm->allocated_size,
-                                     cfrm->size + data_size + FF_INPUT_BUFFER_PADDING_SIZE);
+                                     cfrm->size + data_size + AV_INPUT_BUFFER_PADDING_SIZE);
        // explicit check needed as memcpy below might not catch a NULL
        if (!cfrm->data) {
            av_log(f->avctx, AV_LOG_ERROR, "realloc failure\n");
@@ -1026,5 +1026,5 @@ AVCodec ff_fourxm_decoder = {
    .init           = decode_init,
    .close          = decode_end,
    .decode         = decode_frame,
-    .capabilities   = CODEC_CAP_DR1,
+    .capabilities   = AV_CODEC_CAP_DR1,
 };
@@ -120,12 +120,15 @@ static int decode_frame(AVCodecContext *avctx, void *data,
    }

    if (avctx->bits_per_coded_sample <= 8) {
+        int size;
        const uint8_t *pal = av_packet_get_side_data(avpkt,
                                                     AV_PKT_DATA_PALETTE,
-                                                     NULL);
-        if (pal) {
+                                                     &size);
+        if (pal && size == AVPALETTE_SIZE) {
            frame->palette_has_changed = 1;
            memcpy(c->pal, pal, AVPALETTE_SIZE);
+        } else if (pal) {
+            av_log(avctx, AV_LOG_ERROR, "Palette size %d is wrong\n", size);
        }

        memcpy (frame->data[1], c->pal, AVPALETTE_SIZE);
@@ -184,5 +187,5 @@ AVCodec ff_eightbps_decoder = {
    .priv_data_size = sizeof(EightBpsContext),
    .init           = decode_init,
    .decode         = decode_frame,
-    .capabilities   = CODEC_CAP_DR1,
+    .capabilities   = AV_CODEC_CAP_DR1,
 };
@@ -194,7 +194,7 @@ AVCodec ff_eightsvx_fib_decoder = {
  .init           = eightsvx_decode_init,
  .decode         = eightsvx_decode_frame,
  .close          = eightsvx_decode_close,
-  .capabilities   = CODEC_CAP_DR1,
+  .capabilities   = AV_CODEC_CAP_DR1,
  .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
                                                    AV_SAMPLE_FMT_NONE },
 };
@@ -209,7 +209,7 @@ AVCodec ff_eightsvx_exp_decoder = {
  .init           = eightsvx_decode_init,
  .decode         = eightsvx_decode_frame,
  .close          = eightsvx_decode_close,
-  .capabilities   = CODEC_CAP_DR1,
+  .capabilities   = AV_CODEC_CAP_DR1,
  .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
                                                    AV_SAMPLE_FMT_NONE },
 };
@@ -5,6 +5,7 @@ NAME = avcodec
 HEADERS = avcodec.h                                                     \
          avfft.h                                                       \
          dv_profile.h                                                  \
+          d3d11va.h                                                     \
          dxva2.h                                                       \
          old_codec_ids.h                                               \
          qsv.h                                                         \
@@ -12,6 +13,7 @@ HEADERS = avcodec.h                                                     \
          vda.h                                                         \
          vdpau.h                                                       \
          version.h                                                     \
+          videotoolbox.h                                                \
          vorbis_parser.h                                               \
          xvmc.h                                                        \

@@ -55,6 +57,7 @@ FFT-OBJS-$(CONFIG_HARDCODED_TABLES)    += cos_tables.o cos_fixed_tables.o
 OBJS-$(CONFIG_FFT)                     += avfft.o fft_fixed.o fft_float.o \
                                          fft_fixed_32.o fft_init_table.o \
                                          $(FFT-OBJS-yes)
+OBJS-$(CONFIG_FLACDSP)                 += flacdsp.o
 OBJS-$(CONFIG_FMTCONVERT)              += fmtconvert.o
 OBJS-$(CONFIG_GOLOMB)                  += golomb.o
 OBJS-$(CONFIG_H263DSP)                 += h263dsp.o
@@ -70,6 +73,8 @@ OBJS-$(CONFIG_IDCTDSP)                 += idctdsp.o simple_idct.o jrevdct.o
 OBJS-$(CONFIG_IIRFILTER)               += iirfilter.o
 OBJS-$(CONFIG_IMDCT15)                 += imdct15.o
 OBJS-$(CONFIG_INTRAX8)                 += intrax8.o intrax8dsp.o
+OBJS-$(CONFIG_IVIDSP)                  += ivi_dsp.o
+OBJS-$(CONFIG_JPEGTABLES)              += jpegtables.o
 OBJS-$(CONFIG_LIBXVID)                 += libxvid_rc.o
 OBJS-$(CONFIG_LLAUDDSP)                += lossless_audiodsp.o
 OBJS-$(CONFIG_LLVIDDSP)                += lossless_videodsp.o
@@ -84,34 +89,52 @@ OBJS-$(CONFIG_MPEGAUDIODSP)            += mpegaudiodsp.o                \
                                          mpegaudiodsp_data.o           \
                                          mpegaudiodsp_fixed.o          \
                                          mpegaudiodsp_float.o
-OBJS-$(CONFIG_MPEGVIDEO)               += mpegvideo.o mpegvideodsp.o    \
-                                          mpegvideo_motion.o mpegutils.o
+OBJS-$(CONFIG_MPEGVIDEO)               += mpegvideo.o mpegvideodsp.o rl.o \
+                                          mpegvideo_motion.o mpegutils.o \
+                                          mpegvideodata.o mpegpicture.o
 OBJS-$(CONFIG_MPEGVIDEOENC)            += mpegvideo_enc.o mpeg12data.o  \
                                          motion_est.o ratecontrol.o    \
                                          mpegvideoencdsp.o
+OBJS-$(CONFIG_MSS34DSP)                += mss34dsp.o
+OBJS-$(CONFIG_NVENC)                   += nvenc.o
 OBJS-$(CONFIG_PIXBLOCKDSP)             += pixblockdsp.o
 OBJS-$(CONFIG_QPELDSP)                 += qpeldsp.o
 OBJS-$(CONFIG_QSV)                     += qsv.o
+OBJS-$(CONFIG_QSVDEC)                  += qsvdec.o
+OBJS-$(CONFIG_QSVENC)                  += qsvenc.o
 OBJS-$(CONFIG_RANGECODER)              += rangecoder.o
 RDFT-OBJS-$(CONFIG_HARDCODED_TABLES)   += sin_tables.o
 OBJS-$(CONFIG_RDFT)                    += rdft.o $(RDFT-OBJS-yes)
-OBJS-$(CONFIG_SHARED)                  += log2_tab.o
-OBJS-$(CONFIG_SINEWIN)                 += sinewin.o
+OBJS-$(CONFIG_RV34DSP)                 += rv34dsp.o
+OBJS-$(CONFIG_SHARED)                  += log2_tab.o reverse.o
+OBJS-$(CONFIG_SINEWIN)                 += sinewin.o sinewin_fixed.o
+OBJS-$(CONFIG_SNAPPY)                  += snappy.o
 OBJS-$(CONFIG_STARTCODE)               += startcode.o
+OBJS-$(CONFIG_TEXTUREDSP)              += texturedsp.o
+OBJS-$(CONFIG_TEXTUREDSPENC)           += texturedspenc.o
 OBJS-$(CONFIG_TPELDSP)                 += tpeldsp.o
 OBJS-$(CONFIG_VIDEODSP)                += videodsp.o
 OBJS-$(CONFIG_VP3DSP)                  += vp3dsp.o
+OBJS-$(CONFIG_VP56DSP)                 += vp56dsp.o
+OBJS-$(CONFIG_VP8DSP)                  += vp8dsp.o
 OBJS-$(CONFIG_WMA_FREQS)               += wma_freqs.o
+OBJS-$(CONFIG_WMV2DSP)                 += wmv2dsp.o

 # decoders/encoders
 OBJS-$(CONFIG_ZERO12V_DECODER)         += 012v.o
 OBJS-$(CONFIG_A64MULTI_ENCODER)        += a64multienc.o elbg.o
 OBJS-$(CONFIG_A64MULTI5_ENCODER)       += a64multienc.o elbg.o
-OBJS-$(CONFIG_AAC_DECODER)             += aacdec.o aactab.o aacsbr.o aacps.o \
+OBJS-$(CONFIG_AAC_DECODER)             += aacdec.o aactab.o aacsbr.o aacps_float.o \
                                          aacadtsdec.o mpeg4audio.o kbdwin.o \
-                                          sbrdsp.o aacpsdsp.o
-OBJS-$(CONFIG_AAC_ENCODER)             += aacenc.o aaccoder.o    \
+                                          sbrdsp.o aacpsdsp_float.o
+OBJS-$(CONFIG_AAC_FIXED_DECODER)       += aacdec_fixed.o aactab.o aacsbr_fixed.o aacps_fixed.o \
+                                          aacadtsdec.o mpeg4audio.o kbdwin.o \
+                                          sbrdsp_fixed.o aacpsdsp_fixed.o
+OBJS-$(CONFIG_AAC_ENCODER)             += aacenc.o aaccoder.o aacenctab.o    \
                                          aacpsy.o aactab.o      \
+                                          aacenc_is.o \
+                                          aacenc_tns.o \
+                                          aacenc_pred.o \
                                          psymodel.o mpeg4audio.o kbdwin.o
 OBJS-$(CONFIG_AASC_DECODER)            += aasc.o msrledec.o
 OBJS-$(CONFIG_AC3_DECODER)             += ac3dec_float.o ac3dec_data.o ac3.o kbdwin.o
@@ -133,7 +156,7 @@ OBJS-$(CONFIG_AMRWB_DECODER)           += amrwbdec.o celp_filters.o   \
                                          celp_math.o acelp_filters.o \
                                          acelp_vectors.o             \
                                          acelp_pitch_delay.o
-OBJS-$(CONFIG_AMV_ENCODER)             += mjpegenc.o mjpeg.o mjpegenc_common.o \
+OBJS-$(CONFIG_AMV_ENCODER)             += mjpegenc.o mjpegenc_common.o \
                                          mpegvideo_enc.o motion_est.o \
                                          ratecontrol.o mpeg12data.o   \
                                          mpegvideo.o
@@ -141,6 +164,7 @@ OBJS-$(CONFIG_ANM_DECODER)             += anm.o
 OBJS-$(CONFIG_ANSI_DECODER)            += ansi.o cga_data.o
 OBJS-$(CONFIG_APE_DECODER)             += apedec.o
 OBJS-$(CONFIG_APNG_DECODER)            += png.o pngdec.o pngdsp.o
+OBJS-$(CONFIG_APNG_ENCODER)            += png.o pngenc.o
 OBJS-$(CONFIG_SSA_DECODER)             += assdec.o ass.o ass_split.o
 OBJS-$(CONFIG_SSA_ENCODER)             += assenc.o ass.o
 OBJS-$(CONFIG_ASS_DECODER)             += assdec.o ass.o ass_split.o
@@ -155,7 +179,7 @@ OBJS-$(CONFIG_ATRAC3P_DECODER)         += atrac3plusdec.o atrac3plus.o \
                                          atrac3plusdsp.o atrac.o
 OBJS-$(CONFIG_AURA_DECODER)            += cyuv.o
 OBJS-$(CONFIG_AURA2_DECODER)           += aura.o
-OBJS-$(CONFIG_AVRN_DECODER)            += avrndec.o mjpegdec.o mjpeg.o
+OBJS-$(CONFIG_AVRN_DECODER)            += avrndec.o mjpegdec.o
 OBJS-$(CONFIG_AVRP_DECODER)            += r210dec.o
 OBJS-$(CONFIG_AVRP_ENCODER)            += r210enc.o
 OBJS-$(CONFIG_AVS_DECODER)             += avs.o
@@ -184,7 +208,7 @@ OBJS-$(CONFIG_CINEPAK_DECODER)         += cinepak.o
 OBJS-$(CONFIG_CINEPAK_ENCODER)         += cinepakenc.o elbg.o
 OBJS-$(CONFIG_CLJR_DECODER)            += cljrdec.o
 OBJS-$(CONFIG_CLJR_ENCODER)            += cljrenc.o
-OBJS-$(CONFIG_CLLC_DECODER)            += cllc.o
+OBJS-$(CONFIG_CLLC_DECODER)            += cllc.o canopus.o
 OBJS-$(CONFIG_COOK_DECODER)            += cook.o
 OBJS-$(CONFIG_COMFORTNOISE_DECODER)    += cngdec.o celp_filters.o
 OBJS-$(CONFIG_COMFORTNOISE_ENCODER)    += cngenc.o
@@ -193,8 +217,9 @@ OBJS-$(CONFIG_CSCD_DECODER)            += cscd.o
 OBJS-$(CONFIG_CYUV_DECODER)            += cyuv.o
 OBJS-$(CONFIG_DCA_DECODER)             += dcadec.o dca.o dcadsp.o      \
                                          dcadata.o dca_exss.o         \
-                                          synth_filter.o
+                                          dca_xll.o synth_filter.o
 OBJS-$(CONFIG_DCA_ENCODER)             += dcaenc.o dca.o dcadata.o
+OBJS-$(CONFIG_DDS_DECODER)             += dds.o
 OBJS-$(CONFIG_DIRAC_DECODER)           += diracdec.o dirac.o diracdsp.o \
                                          dirac_arith.o mpeg12data.o dirac_dwt.o
 OBJS-$(CONFIG_DFA_DECODER)             += dfa.o
@@ -217,7 +242,7 @@ OBJS-$(CONFIG_DVVIDEO_DECODER)         += dvdec.o dv.o dvdata.o
 OBJS-$(CONFIG_DVVIDEO_ENCODER)         += dvenc.o dv.o dvdata.o
 OBJS-$(CONFIG_DXA_DECODER)             += dxa.o
 OBJS-$(CONFIG_DXTORY_DECODER)          += dxtory.o
-OBJS-$(CONFIG_EAC3_DECODER)            += eac3dec.o eac3_data.o
+OBJS-$(CONFIG_EAC3_DECODER)            += eac3_data.o
 OBJS-$(CONFIG_EAC3_ENCODER)            += eac3enc.o eac3_data.o
 OBJS-$(CONFIG_EACMV_DECODER)           += eacmv.o
 OBJS-$(CONFIG_EAMAD_DECODER)           += eamad.o eaidct.o mpeg12.o \
@@ -236,8 +261,8 @@ OBJS-$(CONFIG_FFV1_DECODER)            += ffv1dec.o ffv1.o
 OBJS-$(CONFIG_FFV1_ENCODER)            += ffv1enc.o ffv1.o
 OBJS-$(CONFIG_FFWAVESYNTH_DECODER)     += ffwavesynth.o
 OBJS-$(CONFIG_FIC_DECODER)             += fic.o
-OBJS-$(CONFIG_FLAC_DECODER)            += flacdec.o flacdata.o flac.o flacdsp.o
-OBJS-$(CONFIG_FLAC_ENCODER)            += flacenc.o flacdata.o flac.o flacdsp.o vorbis_data.o
+OBJS-$(CONFIG_FLAC_DECODER)            += flacdec.o flacdata.o flac.o
+OBJS-$(CONFIG_FLAC_ENCODER)            += flacenc.o flacdata.o flac.o vorbis_data.o
 OBJS-$(CONFIG_FLASHSV_DECODER)         += flashsv.o
 OBJS-$(CONFIG_FLASHSV_ENCODER)         += flashsvenc.o
 OBJS-$(CONFIG_FLASHSV2_ENCODER)        += flashsv2enc.o
@@ -246,7 +271,7 @@ OBJS-$(CONFIG_FLIC_DECODER)            += flicvideo.o
 OBJS-$(CONFIG_FOURXM_DECODER)          += 4xm.o
 OBJS-$(CONFIG_FRAPS_DECODER)           += fraps.o
 OBJS-$(CONFIG_FRWU_DECODER)            += frwu.o
-OBJS-$(CONFIG_G2M_DECODER)             += g2meet.o mjpeg.o
+OBJS-$(CONFIG_G2M_DECODER)             += g2meet.o elsdec.o
 OBJS-$(CONFIG_G723_1_DECODER)          += g723_1.o acelp_vectors.o \
                                          celp_filters.o celp_math.o
 OBJS-$(CONFIG_G723_1_ENCODER)          += g723_1.o acelp_vectors.o celp_math.o
@@ -259,20 +284,28 @@ OBJS-$(CONFIG_H261_DECODER)            += h261dec.o h261data.o h261.o
 OBJS-$(CONFIG_H261_ENCODER)            += h261enc.o h261data.o h261.o
 OBJS-$(CONFIG_H263_DECODER)            += h263dec.o h263.o ituh263dec.o        \
                                          mpeg4video.o mpeg4videodec.o flvdec.o\
-                                          intelh263dec.o
+                                          intelh263dec.o h263data.o
 OBJS-$(CONFIG_H263_ENCODER)            += mpeg4videoenc.o mpeg4video.o  \
-                                          h263.o ituh263enc.o flvenc.o
+                                          h263.o h263data.o ituh263enc.o flvenc.o
 OBJS-$(CONFIG_H264_DECODER)            += h264.o h264_cabac.o h264_cavlc.o \
                                          h264_direct.o h264_loopfilter.o  \
                                          h264_mb.o h264_picture.o h264_ps.o \
                                          h264_refs.o h264_sei.o h264_slice.o
+OBJS-$(CONFIG_H264_MMAL_DECODER)       += mmaldec.o
 OBJS-$(CONFIG_H264_VDA_DECODER)        += vda_h264_dec.o
-OBJS-$(CONFIG_H264_QSV_DECODER)        += qsv_h264.o
+OBJS-$(CONFIG_H264_QSV_DECODER)        += qsvdec_h2645.o
+OBJS-$(CONFIG_H264_QSV_ENCODER)        += qsvenc_h264.o
+OBJS-$(CONFIG_HAP_DECODER)             += hapdec.o hap.o
+OBJS-$(CONFIG_HAP_ENCODER)             += hapenc.o hap.o
 OBJS-$(CONFIG_HEVC_DECODER)            += hevc.o hevc_mvs.o hevc_ps.o hevc_sei.o \
                                          hevc_cabac.o hevc_refs.o hevcpred.o    \
-                                          hevcdsp.o hevc_filter.o
+                                          hevcdsp.o hevc_filter.o hevc_parse.o hevc_data.o
+OBJS-$(CONFIG_HEVC_QSV_DECODER)        += qsvdec_h2645.o
+OBJS-$(CONFIG_HEVC_QSV_ENCODER)        += qsvenc_hevc.o hevc_ps_enc.o hevc_parse.o
 OBJS-$(CONFIG_HNM4_VIDEO_DECODER)      += hnm4video.o
-OBJS-$(CONFIG_HQX_DECODER)             += hqx.o hqxvlc.o
+OBJS-$(CONFIG_HQ_HQA_DECODER)          += hq_hqa.o hq_hqadata.o hq_hqadsp.o \
+                                          canopus.o
+OBJS-$(CONFIG_HQX_DECODER)             += hqx.o hqxvlc.o hqxdsp.o canopus.o
 OBJS-$(CONFIG_HUFFYUV_DECODER)         += huffyuv.o huffyuvdec.o
 OBJS-$(CONFIG_HUFFYUV_ENCODER)         += huffyuv.o huffyuvenc.o
 OBJS-$(CONFIG_IDCIN_DECODER)           += idcinvideo.o
@@ -282,8 +315,8 @@ OBJS-$(CONFIG_IFF_ILBM_DECODER)        += iff.o
 OBJS-$(CONFIG_IMC_DECODER)             += imc.o
 OBJS-$(CONFIG_INDEO2_DECODER)          += indeo2.o
 OBJS-$(CONFIG_INDEO3_DECODER)          += indeo3.o
-OBJS-$(CONFIG_INDEO4_DECODER)          += indeo4.o ivi_common.o ivi_dsp.o
-OBJS-$(CONFIG_INDEO5_DECODER)          += indeo5.o ivi_common.o ivi_dsp.o
+OBJS-$(CONFIG_INDEO4_DECODER)          += indeo4.o ivi.o
+OBJS-$(CONFIG_INDEO5_DECODER)          += indeo5.o ivi.o
 OBJS-$(CONFIG_INTERPLAY_DPCM_DECODER)  += dpcm.o
 OBJS-$(CONFIG_INTERPLAY_VIDEO_DECODER) += interplayvideo.o
 OBJS-$(CONFIG_JACOSUB_DECODER)         += jacosubdec.o ass.o
@@ -297,7 +330,7 @@ OBJS-$(CONFIG_JV_DECODER)              += jvdec.o
 OBJS-$(CONFIG_KGV1_DECODER)            += kgv1dec.o
 OBJS-$(CONFIG_KMVC_DECODER)            += kmvc.o
 OBJS-$(CONFIG_LAGARITH_DECODER)        += lagarith.o lagarithrac.o
-OBJS-$(CONFIG_LJPEG_ENCODER)           += ljpegenc.o mjpeg.o mjpegenc_common.o
+OBJS-$(CONFIG_LJPEG_ENCODER)           += ljpegenc.o mjpegenc_common.o
 OBJS-$(CONFIG_LOCO_DECODER)            += loco.o
 OBJS-$(CONFIG_MACE3_DECODER)           += mace.o
 OBJS-$(CONFIG_MACE6_DECODER)           += mace.o
@@ -306,8 +339,8 @@ OBJS-$(CONFIG_METASOUND_DECODER)       += metasound.o metasound_data.o \
                                          twinvq.o
 OBJS-$(CONFIG_MICRODVD_DECODER)        += microdvddec.o ass.o
 OBJS-$(CONFIG_MIMIC_DECODER)           += mimic.o
-OBJS-$(CONFIG_MJPEG_DECODER)           += mjpegdec.o mjpeg.o
-OBJS-$(CONFIG_MJPEG_ENCODER)           += mjpegenc.o mjpeg.o mjpegenc_common.o
+OBJS-$(CONFIG_MJPEG_DECODER)           += mjpegdec.o
+OBJS-$(CONFIG_MJPEG_ENCODER)           += mjpegenc.o mjpegenc_common.o
 OBJS-$(CONFIG_MJPEGB_DECODER)          += mjpegbdec.o
 OBJS-$(CONFIG_MLP_DECODER)             += mlpdec.o mlpdsp.o
 OBJS-$(CONFIG_MMVIDEO_DECODER)         += mmvideo.o
@@ -335,6 +368,8 @@ OBJS-$(CONFIG_MPEG1VIDEO_DECODER)      += mpeg12dec.o mpeg12.o mpeg12data.o
 OBJS-$(CONFIG_MPEG1VIDEO_ENCODER)      += mpeg12enc.o mpeg12.o
 OBJS-$(CONFIG_MPEG2VIDEO_DECODER)      += mpeg12dec.o mpeg12.o mpeg12data.o
 OBJS-$(CONFIG_MPEG2VIDEO_ENCODER)      += mpeg12enc.o mpeg12.o
+OBJS-$(CONFIG_MPEG2_QSV_DECODER)       += qsvdec_mpeg2.o
+OBJS-$(CONFIG_MPEG2_QSV_ENCODER)       += qsvenc_mpeg2.o
 OBJS-$(CONFIG_MPEG4_DECODER)           += xvididct.o
 OBJS-$(CONFIG_MPL2_DECODER)            += mpl2dec.o ass.o
 OBJS-$(CONFIG_MSMPEG4V1_DECODER)       += msmpeg4dec.o msmpeg4.o msmpeg4data.o
@@ -343,20 +378,19 @@ OBJS-$(CONFIG_MSMPEG4V2_ENCODER)       += msmpeg4enc.o msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_MSMPEG4V3_DECODER)       += msmpeg4dec.o msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_MSMPEG4V3_ENCODER)       += msmpeg4enc.o msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_MSRLE_DECODER)           += msrle.o msrledec.o
-OBJS-$(CONFIG_MSA1_DECODER)            += mss3.o mss34dsp.o
+OBJS-$(CONFIG_MSA1_DECODER)            += mss3.o
 OBJS-$(CONFIG_MSS1_DECODER)            += mss1.o mss12.o
 OBJS-$(CONFIG_MSS2_DECODER)            += mss2.o mss12.o mss2dsp.o
 OBJS-$(CONFIG_MSVIDEO1_DECODER)        += msvideo1.o
 OBJS-$(CONFIG_MSVIDEO1_ENCODER)        += msvideo1enc.o elbg.o
 OBJS-$(CONFIG_MSZH_DECODER)            += lcldec.o
-OBJS-$(CONFIG_MTS2_DECODER)            += mss4.o mss34dsp.o
+OBJS-$(CONFIG_MTS2_DECODER)            += mss4.o
 OBJS-$(CONFIG_MVC1_DECODER)            += mvcdec.o
 OBJS-$(CONFIG_MVC2_DECODER)            += mvcdec.o
 OBJS-$(CONFIG_MXPEG_DECODER)           += mxpegdec.o
 OBJS-$(CONFIG_NELLYMOSER_DECODER)      += nellymoserdec.o nellymoser.o
 OBJS-$(CONFIG_NELLYMOSER_ENCODER)      += nellymoserenc.o nellymoser.o
 OBJS-$(CONFIG_NUV_DECODER)             += nuv.o rtjpeg.o
-OBJS-$(CONFIG_NVENC_ENCODER)           += nvenc.o
 OBJS-$(CONFIG_ON2AVC_DECODER)          += on2avc.o on2avcdata.o
 OBJS-$(CONFIG_OPUS_DECODER)            += opusdec.o opus.o opus_celt.o \
                                          opus_silk.o vorbis_data.o
@@ -414,8 +448,8 @@ OBJS-$(CONFIG_RV10_DECODER)            += rv10.o
 OBJS-$(CONFIG_RV10_ENCODER)            += rv10enc.o
 OBJS-$(CONFIG_RV20_DECODER)            += rv10.o
 OBJS-$(CONFIG_RV20_ENCODER)            += rv20enc.o
-OBJS-$(CONFIG_RV30_DECODER)            += rv30.o rv34.o rv30dsp.o rv34dsp.o
-OBJS-$(CONFIG_RV40_DECODER)            += rv40.o rv34.o rv34dsp.o rv40dsp.o
+OBJS-$(CONFIG_RV30_DECODER)            += rv30.o rv34.o rv30dsp.o
+OBJS-$(CONFIG_RV40_DECODER)            += rv40.o rv34.o rv40dsp.o
 OBJS-$(CONFIG_SAMI_DECODER)            += samidec.o ass.o
 OBJS-$(CONFIG_S302M_DECODER)           += s302m.o
 OBJS-$(CONFIG_S302M_ENCODER)           += s302menc.o
@@ -458,6 +492,7 @@ OBJS-$(CONFIG_TAK_DECODER)             += takdec.o tak.o
 OBJS-$(CONFIG_TARGA_DECODER)           += targa.o
 OBJS-$(CONFIG_TARGA_ENCODER)           += targaenc.o rle.o
 OBJS-$(CONFIG_TARGA_Y216_DECODER)      += targa_y216dec.o
+OBJS-$(CONFIG_TDSC_DECODER)            += tdsc.o
 OBJS-$(CONFIG_TIERTEXSEQVIDEO_DECODER) += tiertexseqv.o
 OBJS-$(CONFIG_TIFF_DECODER)            += tiff.o lzw.o faxcompr.o tiff_data.o tiff_common.o
 OBJS-$(CONFIG_TIFF_ENCODER)            += tiffenc.o rle.o lzwenc.o tiff_data.o
@@ -471,7 +506,7 @@ OBJS-$(CONFIG_TSCC2_DECODER)           += tscc2.o
 OBJS-$(CONFIG_TTA_DECODER)             += tta.o ttadata.o ttadsp.o
 OBJS-$(CONFIG_TTA_ENCODER)             += ttaenc.o ttadata.o
 OBJS-$(CONFIG_TWINVQ_DECODER)          += twinvqdec.o twinvq.o
-OBJS-$(CONFIG_TXD_DECODER)             += txd.o s3tc.o
+OBJS-$(CONFIG_TXD_DECODER)             += txd.o
 OBJS-$(CONFIG_ULTI_DECODER)            += ulti.o
 OBJS-$(CONFIG_UTVIDEO_DECODER)         += utvideodec.o utvideo.o
 OBJS-$(CONFIG_UTVIDEO_ENCODER)         += utvideoenc.o utvideo.o
@@ -491,6 +526,7 @@ OBJS-$(CONFIG_VC1_DECODER)             += vc1dec.o vc1_block.o vc1_loopfilter.o
                                          vc1dsp.o \
                                          msmpeg4dec.o msmpeg4.o msmpeg4data.o \
                                          wmv2dsp.o
+OBJS-$(CONFIG_VC1_QSV_DECODER)         += qsvdec_vc1.o
 OBJS-$(CONFIG_VCR1_DECODER)            += vcr1.o
 OBJS-$(CONFIG_VMDAUDIO_DECODER)        += vmdaudio.o
 OBJS-$(CONFIG_VMDVIDEO_DECODER)        += vmdvideo.o
@@ -500,13 +536,13 @@ OBJS-$(CONFIG_VORBIS_DECODER)          += vorbisdec.o vorbisdsp.o vorbis.o \
 OBJS-$(CONFIG_VORBIS_ENCODER)          += vorbisenc.o vorbis.o \
                                          vorbis_data.o
 OBJS-$(CONFIG_VP3_DECODER)             += vp3.o
-OBJS-$(CONFIG_VP5_DECODER)             += vp5.o vp56.o vp56data.o vp56dsp.o \
-                                          vp56rac.o
-OBJS-$(CONFIG_VP6_DECODER)             += vp6.o vp56.o vp56data.o vp56dsp.o \
+OBJS-$(CONFIG_VP5_DECODER)             += vp5.o vp56.o vp56data.o vp56rac.o
+OBJS-$(CONFIG_VP6_DECODER)             += vp6.o vp56.o vp56data.o \
                                          vp6dsp.o vp56rac.o
-OBJS-$(CONFIG_VP7_DECODER)             += vp8.o vp8dsp.o vp56rac.o
-OBJS-$(CONFIG_VP8_DECODER)             += vp8.o vp8dsp.o vp56rac.o
-OBJS-$(CONFIG_VP9_DECODER)             += vp9.o vp9dsp.o vp56rac.o
+OBJS-$(CONFIG_VP7_DECODER)             += vp8.o vp56rac.o
+OBJS-$(CONFIG_VP8_DECODER)             += vp8.o vp56rac.o
+OBJS-$(CONFIG_VP9_DECODER)             += vp9.o vp9dsp.o vp56rac.o vp9dsp_8bpp.o \
+                                          vp9dsp_10bpp.o vp9dsp_12bpp.o
 OBJS-$(CONFIG_VPLAYER_DECODER)         += textdec.o ass.o
 OBJS-$(CONFIG_VQA_DECODER)             += vqavideo.o
 OBJS-$(CONFIG_WAVPACK_DECODER)         += wavpack.o
@@ -526,9 +562,9 @@ OBJS-$(CONFIG_WMAVOICE_DECODER)        += wmavoice.o \
                                          acelp_vectors.o acelp_filters.o
 OBJS-$(CONFIG_WMV1_DECODER)            += msmpeg4dec.o msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_WMV1_ENCODER)            += msmpeg4enc.o
-OBJS-$(CONFIG_WMV2_DECODER)            += wmv2dec.o wmv2.o wmv2dsp.o \
+OBJS-$(CONFIG_WMV2_DECODER)            += wmv2dec.o wmv2.o \
                                          msmpeg4dec.o msmpeg4.o msmpeg4data.o
-OBJS-$(CONFIG_WMV2_ENCODER)            += wmv2enc.o wmv2.o wmv2dsp.o \
+OBJS-$(CONFIG_WMV2_ENCODER)            += wmv2enc.o wmv2.o \
                                          msmpeg4.o msmpeg4enc.o msmpeg4data.o
 OBJS-$(CONFIG_WNV1_DECODER)            += wnv1.o
 OBJS-$(CONFIG_WS_SND1_DECODER)         += ws-snd1.o
@@ -661,26 +697,39 @@ OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER)       += adpcmenc.o adpcm_data.o
 OBJS-$(CONFIG_VIMA_DECODER)               += vima.o adpcm_data.o

 # hardware accelerators
+OBJS-$(CONFIG_D3D11VA)                    += dxva2.o
 OBJS-$(CONFIG_DXVA2)                      += dxva2.o
 OBJS-$(CONFIG_VAAPI)                      += vaapi.o
-OBJS-$(CONFIG_VDA)                        += vda.o
+OBJS-$(CONFIG_VDA)                        += vda.o videotoolbox.o
+OBJS-$(CONFIG_VIDEOTOOLBOX)               += videotoolbox.o
 OBJS-$(CONFIG_VDPAU)                      += vdpau.o

 OBJS-$(CONFIG_H263_VAAPI_HWACCEL)         += vaapi_mpeg4.o
 OBJS-$(CONFIG_H263_VDPAU_HWACCEL)         += vdpau_mpeg4.o
+OBJS-$(CONFIG_H263_VIDEOTOOLBOX_HWACCEL)  += videotoolbox.o
+OBJS-$(CONFIG_H264_D3D11VA_HWACCEL)       += dxva2_h264.o
 OBJS-$(CONFIG_H264_DXVA2_HWACCEL)         += dxva2_h264.o
 OBJS-$(CONFIG_H264_VAAPI_HWACCEL)         += vaapi_h264.o
 OBJS-$(CONFIG_H264_VDA_HWACCEL)           += vda_h264.o
 OBJS-$(CONFIG_H264_VDPAU_HWACCEL)         += vdpau_h264.o
+OBJS-$(CONFIG_H264_VIDEOTOOLBOX_HWACCEL)  += videotoolbox.o
+OBJS-$(CONFIG_HEVC_D3D11VA_HWACCEL)       += dxva2_hevc.o
 OBJS-$(CONFIG_HEVC_DXVA2_HWACCEL)         += dxva2_hevc.o
+OBJS-$(CONFIG_HEVC_VAAPI_HWACCEL)         += vaapi_hevc.o
+OBJS-$(CONFIG_HEVC_VDPAU_HWACCEL)         += vdpau_hevc.o
 OBJS-$(CONFIG_MPEG1_VDPAU_HWACCEL)        += vdpau_mpeg12.o
+OBJS-$(CONFIG_MPEG1_VIDEOTOOLBOX_HWACCEL) += videotoolbox.o
 OBJS-$(CONFIG_MPEG1_XVMC_HWACCEL)         += mpegvideo_xvmc.o
+OBJS-$(CONFIG_MPEG2_D3D11VA_HWACCEL)      += dxva2_mpeg2.o
 OBJS-$(CONFIG_MPEG2_DXVA2_HWACCEL)        += dxva2_mpeg2.o
 OBJS-$(CONFIG_MPEG2_VAAPI_HWACCEL)        += vaapi_mpeg2.o
 OBJS-$(CONFIG_MPEG2_VDPAU_HWACCEL)        += vdpau_mpeg12.o
+OBJS-$(CONFIG_MPEG2_VIDEOTOOLBOX_HWACCEL) += videotoolbox.o
 OBJS-$(CONFIG_MPEG2_XVMC_HWACCEL)         += mpegvideo_xvmc.o
 OBJS-$(CONFIG_MPEG4_VAAPI_HWACCEL)        += vaapi_mpeg4.o
 OBJS-$(CONFIG_MPEG4_VDPAU_HWACCEL)        += vdpau_mpeg4.o
+OBJS-$(CONFIG_MPEG4_VIDEOTOOLBOX_HWACCEL) += videotoolbox.o
+OBJS-$(CONFIG_VC1_D3D11VA_HWACCEL)        += dxva2_vc1.o
 OBJS-$(CONFIG_VC1_DXVA2_HWACCEL)          += dxva2_vc1.o
 OBJS-$(CONFIG_VC1_VAAPI_HWACCEL)          += vaapi_vc1.o
 OBJS-$(CONFIG_VC1_VDPAU_HWACCEL)          += vdpau_vc1.o
@@ -716,7 +765,6 @@ OBJS-$(CONFIG_OGG_DEMUXER)             += mpeg12data.o \
 OBJS-$(CONFIG_OGG_MUXER)               += flac.o flacdata.o \
                                          vorbis_data.o
 OBJS-$(CONFIG_RTP_MUXER)               += mpeg4audio.o
-OBJS-$(CONFIG_RTPDEC)                  += mjpeg.o
 OBJS-$(CONFIG_SPDIF_DEMUXER)           += aacadtsdec.o mpeg4audio.o
 OBJS-$(CONFIG_SPDIF_MUXER)             += dca.o
 OBJS-$(CONFIG_TAK_DEMUXER)             += tak.o
@@ -731,6 +779,7 @@ OBJS-$(CONFIG_ELBG_FILTER)             += elbg.o
 # external codec libraries
 OBJS-$(CONFIG_LIBAACPLUS_ENCODER)         += libaacplus.o
 OBJS-$(CONFIG_LIBCELT_DECODER)            += libcelt_dec.o
+OBJS-$(CONFIG_LIBDCADEC_DECODER)          += libdcadec.o dca.o
 OBJS-$(CONFIG_LIBFAAC_ENCODER)            += libfaac.o
 OBJS-$(CONFIG_LIBFDK_AAC_DECODER)         += libfdk-aacdec.o
 OBJS-$(CONFIG_LIBFDK_AAC_ENCODER)         += libfdk-aacenc.o
@@ -740,6 +789,7 @@ OBJS-$(CONFIG_LIBGSM_MS_DECODER)          += libgsmdec.o
 OBJS-$(CONFIG_LIBGSM_MS_ENCODER)          += libgsmenc.o
 OBJS-$(CONFIG_LIBILBC_DECODER)            += libilbc.o
 OBJS-$(CONFIG_LIBILBC_ENCODER)            += libilbc.o
+OBJS-$(CONFIG_LIBKVAZAAR_ENCODER)         += libkvazaar.o
 OBJS-$(CONFIG_LIBMP3LAME_ENCODER)         += libmp3lame.o mpegaudiodecheader.o
 OBJS-$(CONFIG_LIBOPENCORE_AMRNB_DECODER)  += libopencore-amr.o
 OBJS-$(CONFIG_LIBOPENCORE_AMRNB_ENCODER)  += libopencore-amr.o
@@ -773,7 +823,9 @@ OBJS-$(CONFIG_LIBVPX_VP8_ENCODER)         += libvpxenc.o
 OBJS-$(CONFIG_LIBVPX_VP9_DECODER)         += libvpxdec.o libvpx.o
 OBJS-$(CONFIG_LIBVPX_VP9_ENCODER)         += libvpxenc.o libvpx.o
 OBJS-$(CONFIG_LIBWAVPACK_ENCODER)         += libwavpackenc.o
-OBJS-$(CONFIG_LIBWEBP_ENCODER)            += libwebpenc.o
+OBJS-$(CONFIG_LIBWEBP_ENCODER)            += libwebpenc_common.o libwebpenc.o
+OBJS-$(CONFIG_LIBWEBP_ANIM_ENCODER)       += libwebpenc_common.o libwebpenc_animencoder.o
+OBJS-$(CONFIG_LIBX262_ENCODER)            += libx264.o
 OBJS-$(CONFIG_LIBX264_ENCODER)            += libx264.o
 OBJS-$(CONFIG_LIBX265_ENCODER)            += libx265.o
 OBJS-$(CONFIG_LIBXAVS_ENCODER)            += libxavs.o
@@ -799,16 +851,17 @@ OBJS-$(CONFIG_DVD_NAV_PARSER)          += dvd_nav_parser.o
 OBJS-$(CONFIG_DVDSUB_PARSER)           += dvdsub_parser.o
 OBJS-$(CONFIG_FLAC_PARSER)             += flac_parser.o flacdata.o flac.o \
                                          vorbis_data.o
+OBJS-$(CONFIG_G729_PARSER)             += g729_parser.o
 OBJS-$(CONFIG_GSM_PARSER)              += gsm_parser.o
 OBJS-$(CONFIG_H261_PARSER)             += h261_parser.o
 OBJS-$(CONFIG_H263_PARSER)             += h263_parser.o
 OBJS-$(CONFIG_H264_PARSER)             += h264_parser.o
-OBJS-$(CONFIG_HEVC_PARSER)             += hevc_parser.o
+OBJS-$(CONFIG_HEVC_PARSER)             += hevc_parser.o hevc_parse.o hevc_ps.o hevc_data.o
 OBJS-$(CONFIG_MJPEG_PARSER)            += mjpeg_parser.o
 OBJS-$(CONFIG_MLP_PARSER)              += mlp_parser.o mlp.o
 OBJS-$(CONFIG_MPEG4VIDEO_PARSER)       += mpeg4video_parser.o h263.o \
                                          mpeg4videodec.o mpeg4video.o \
-                                          ituh263dec.o h263dec.o
+                                          ituh263dec.o h263dec.o h263data.o
 OBJS-$(CONFIG_PNG_PARSER)              += png_parser.o
 OBJS-$(CONFIG_MPEGAUDIO_PARSER)        += mpegaudio_parser.o \
                                          mpegaudiodecheader.o mpegaudiodata.o
@@ -831,9 +884,11 @@ OBJS-$(CONFIG_AAC_ADTSTOASC_BSF)          += aac_adtstoasc_bsf.o aacadtsdec.o \
 OBJS-$(CONFIG_CHOMP_BSF)                  += chomp_bsf.o
 OBJS-$(CONFIG_DUMP_EXTRADATA_BSF)         += dump_extradata_bsf.o
 OBJS-$(CONFIG_H264_MP4TOANNEXB_BSF)       += h264_mp4toannexb_bsf.o
+OBJS-$(CONFIG_HEVC_MP4TOANNEXB_BSF)       += hevc_mp4toannexb_bsf.o
 OBJS-$(CONFIG_IMX_DUMP_HEADER_BSF)        += imx_dump_header_bsf.o
-OBJS-$(CONFIG_MJPEG2JPEG_BSF)             += mjpeg2jpeg_bsf.o mjpeg.o
+OBJS-$(CONFIG_MJPEG2JPEG_BSF)             += mjpeg2jpeg_bsf.o
 OBJS-$(CONFIG_MJPEGA_DUMP_HEADER_BSF)     += mjpega_dump_header_bsf.o
+OBJS-$(CONFIG_MPEG4_UNPACK_BFRAMES_BSF)   += mpeg4_unpack_bframes_bsf.o
 OBJS-$(CONFIG_MOV2TEXTSUB_BSF)            += movsub_bsf.o
 OBJS-$(CONFIG_MP3_HEADER_DECOMPRESS_BSF)  += mp3_header_decompress_bsf.o \
                                             mpegaudiodata.o
@@ -854,26 +909,31 @@ SKIPHEADERS                            += %_tablegen.h                  \
                                          %_tables.h                    \
                                          aac_tablegen_decl.h           \
                                          fft-internal.h                \
-                                          libutvideo.h                  \
                                          old_codec_ids.h               \
                                          tableprint.h                  \
                                          tableprint_vlc.h              \
                                          $(ARCH)/vp56_arith.h          \

+SKIPHEADERS-$(CONFIG_D3D11VA)          += d3d11va.h dxva2_internal.h
 SKIPHEADERS-$(CONFIG_DXVA2)            += dxva2.h dxva2_internal.h
 SKIPHEADERS-$(CONFIG_LIBSCHROEDINGER)  += libschroedinger.h
 SKIPHEADERS-$(CONFIG_LIBUTVIDEO)       += libutvideo.h
+SKIPHEADERS-$(CONFIG_LIBWEBP_ENCODER)  += libwebpenc_common.h
 SKIPHEADERS-$(CONFIG_QSV)              += qsv.h qsv_internal.h
+SKIPHEADERS-$(CONFIG_QSVDEC)           += qsvdec.h
+SKIPHEADERS-$(CONFIG_QSVENC)           += qsvenc.h
 SKIPHEADERS-$(CONFIG_XVMC)             += xvmc.h
 SKIPHEADERS-$(CONFIG_VAAPI)            += vaapi_internal.h
-SKIPHEADERS-$(CONFIG_VDA)              += vda.h vda_internal.h
+SKIPHEADERS-$(CONFIG_VDA)              += vda.h vda_vt_internal.h
 SKIPHEADERS-$(CONFIG_VDPAU)            += vdpau.h vdpau_internal.h
+SKIPHEADERS-$(CONFIG_VIDEOTOOLBOX)     += videotoolbox.h vda_vt_internal.h

 TESTPROGS = imgconvert                                                  \
+            jpeg2000dwt                                                 \
+            mathops                                                    \
            options                                                     \
            avfft                                                       \

-
 TESTPROGS-$(CONFIG_CABAC)                 += cabac
 TESTPROGS-$(CONFIG_FFT)                   += fft fft-fixed fft-fixed32
 TESTPROGS-$(CONFIG_IDCTDSP)               += dct
@@ -889,9 +949,12 @@ TOOLS = fourcc2pixfmt

 HOSTPROGS = aac_tablegen                                                \
            aacps_tablegen                                              \
+            aacps_fixed_tablegen                                        \
            aacsbr_tablegen                                             \
+            aacsbr_fixed_tablegen                                       \
            cabac_tablegen                                              \
            cbrt_tablegen                                               \
+            cbrt_fixed_tablegen                                         \
            cos_tablegen                                                \
            dsd_tablegen                                                \
            dv_tablegen                                                 \
@@ -900,6 +963,7 @@ HOSTPROGS = aac_tablegen                                                \
            pcm_tablegen                                                \
            qdm2_tablegen                                               \
            sinewin_tablegen                                            \
+            sinewin_fixed_tablegen                                      \

 CLEANFILES = *_tables.c *_tables.h *_tablegen$(HOSTEXESUF)

@@ -918,8 +982,9 @@ else
 $(SUBDIR)%_tablegen$(HOSTEXESUF): HOSTCFLAGS += -DCONFIG_SMALL=0
 endif

-GEN_HEADERS = cabac_tables.h cbrt_tables.h aacps_tables.h aacsbr_tables.h aac_tables.h dsd_tables.h dv_tables.h     \
-              sinewin_tables.h mpegaudio_tables.h motionpixels_tables.h \
+GEN_HEADERS = cabac_tables.h cbrt_tables.h cbrt_fixed_tables.h aacps_tables.h aacps_fixed_tables.h aacsbr_tables.h \
+              aacsbr_fixed_tables.h aac_tables.h dsd_tables.h dv_tables.h     \
+              sinewin_tables.h sinewin_fixed_tables.h mpegaudio_tables.h motionpixels_tables.h \
              pcm_tables.h qdm2_tables.h
 GEN_HEADERS := $(addprefix $(SUBDIR), $(GEN_HEADERS))

@@ -928,13 +993,18 @@ $(GEN_HEADERS): $(SUBDIR)%_tables.h: $(SUBDIR)%_tablegen$(HOSTEXESUF)

 ifdef CONFIG_HARDCODED_TABLES
 $(SUBDIR)aacdec.o: $(SUBDIR)cbrt_tables.h
-$(SUBDIR)aacps.o: $(SUBDIR)aacps_tables.h
+$(SUBDIR)aacdec_fixed.o: $(SUBDIR)cbrt_fixed_tables.h
+$(SUBDIR)aacps_float.o: $(SUBDIR)aacps_tables.h
+$(SUBDIR)aacps_fixed.o: $(SUBDIR)aacps_fixed_tables.h
 $(SUBDIR)aacsbr.o: $(SUBDIR)aacsbr_tables.h
+$(SUBDIR)aacsbr_fixed.o: $(SUBDIR)aacsbr_fixed_tables.h
 $(SUBDIR)aactab.o: $(SUBDIR)aac_tables.h
+$(SUBDIR)aactab_fixed.o: $(SUBDIR)aac_fixed_tables.h
 $(SUBDIR)cabac.o: $(SUBDIR)cabac_tables.h
 $(SUBDIR)dsddec.o: $(SUBDIR)dsd_tables.h
 $(SUBDIR)dvenc.o: $(SUBDIR)dv_tables.h
 $(SUBDIR)sinewin.o: $(SUBDIR)sinewin_tables.h
+$(SUBDIR)sinewin_fixed.o: $(SUBDIR)sinewin_fixed_tables.h
 $(SUBDIR)mpegaudiodec_fixed.o: $(SUBDIR)mpegaudio_tables.h
 $(SUBDIR)mpegaudiodec_float.o: $(SUBDIR)mpegaudio_tables.h
 $(SUBDIR)motionpixels.o: $(SUBDIR)motionpixels_tables.h
@@ -66,7 +66,8 @@ static const int mc_colors[5]={0x0,0xb,0xc,0xf,0x1};
 //static const int mc_colors[5]={0x0,0x8,0xa,0xf,0x7};
 //static const int mc_colors[5]={0x0,0x9,0x8,0xa,0x3};

-static void to_meta_with_crop(AVCodecContext *avctx, const AVFrame *p, int *dest)
+static void to_meta_with_crop(AVCodecContext *avctx,
+                              const AVFrame *p, int *dest)
 {
    int blockx, blocky, x, y;
    int luma = 0;
@@ -234,7 +235,7 @@ static av_cold int a64multi_encode_init(AVCodecContext *avctx)
    }

    /* set up extradata */
-    if (!(avctx->extradata = av_mallocz(8 * 4 + FF_INPUT_BUFFER_PADDING_SIZE))) {
+    if (!(avctx->extradata = av_mallocz(8 * 4 + AV_INPUT_BUFFER_PADDING_SIZE))) {
        av_log(avctx, AV_LOG_ERROR, "Failed to allocate memory for extradata.\n");
        return AVERROR(ENOMEM);
    }
@@ -327,7 +328,7 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
        /* any frames to encode? */
        if (c->mc_lifetime) {
            int alloc_size = charset_size + c->mc_lifetime*(screen_size + colram_size);
-            if ((ret = ff_alloc_packet2(avctx, pkt, alloc_size)) < 0)
+            if ((ret = ff_alloc_packet2(avctx, pkt, alloc_size, 0)) < 0)
                return ret;
            buf = pkt->data;

@@ -405,7 +406,7 @@ AVCodec ff_a64multi_encoder = {
    .encode2        = a64multi_encode_frame,
    .close          = a64multi_close_encoder,
    .pix_fmts       = (const enum AVPixelFormat[]) {AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE},
-    .capabilities   = CODEC_CAP_DELAY,
+    .capabilities   = AV_CODEC_CAP_DELAY,
 };
 #endif
 #if CONFIG_A64MULTI5_ENCODER
@@ -419,6 +420,6 @@ AVCodec ff_a64multi5_encoder = {
    .encode2        = a64multi_encode_frame,
    .close          = a64multi_close_encoder,
    .pix_fmts       = (const enum AVPixelFormat[]) {AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE},
-    .capabilities   = CODEC_CAP_DELAY,
+    .capabilities   = AV_CODEC_CAP_DELAY,
 };
 #endif
@@ -30,9 +30,14 @@
 #ifndef AVCODEC_AAC_H
 #define AVCODEC_AAC_H

+
+#include "aac_defines.h"
 #include "libavutil/float_dsp.h"
+#include "libavutil/fixed_dsp.h"
 #include "avcodec.h"
+#if !USE_FIXED
 #include "imdct15.h"
+#endif
 #include "fft.h"
 #include "mpeg4audio.h"
 #include "sbr.h"
@@ -45,6 +50,8 @@
 #define TNS_MAX_ORDER 20
 #define MAX_LTP_LONG_SFB 40

+#define CLIP_AVOIDANCE_FACTOR 0.95f
+
 enum RawDataBlockType {
    TYPE_SCE,
    TYPE_CPE,
@@ -76,9 +83,10 @@ enum BandType {
    ZERO_BT        = 0,     ///< Scalefactors and spectral data are all zero.
    FIRST_PAIR_BT  = 5,     ///< This and later band types encode two values (rather than four) with one code word.
    ESC_BT         = 11,    ///< Spectral data are coded with an escape sequence.
+    RESERVED_BT    = 12,    ///< Band types following are encoded differently from others.
    NOISE_BT       = 13,    ///< Spectral data are scaled white noise not coded in the bitstream.
-    INTENSITY_BT2  = 14,    ///< Scalefactor data are intensity stereo positions.
-    INTENSITY_BT   = 15,    ///< Scalefactor data are intensity stereo positions.
+    INTENSITY_BT2  = 14,    ///< Scalefactor data are intensity stereo positions (out of phase).
+    INTENSITY_BT   = 15,    ///< Scalefactor data are intensity stereo positions (in phase).
 };

 #define IS_CODEBOOK_UNSIGNED(x) (((x) - 1) & 10)
@@ -125,12 +133,14 @@ typedef struct OutputConfiguration {
 * Predictor State
 */
 typedef struct PredictorState {
-    float cor0;
-    float cor1;
-    float var0;
-    float var1;
-    float r0;
-    float r1;
+    AAC_FLOAT cor0;
+    AAC_FLOAT cor1;
+    AAC_FLOAT var0;
+    AAC_FLOAT var1;
+    AAC_FLOAT r0;
+    AAC_FLOAT r1;
+    AAC_FLOAT k1;
+    AAC_FLOAT x_est;
 } PredictorState;

 #define MAX_PREDICTORS 672
@@ -141,13 +151,17 @@ typedef struct PredictorState {
 #define SCALE_MAX_DIFF   60    ///< maximum scalefactor difference allowed by standard
 #define SCALE_DIFF_ZERO  60    ///< codebook index corresponding to zero scalefactor indices difference

+#define NOISE_PRE       256    ///< preamble for NOISE_BT, put in bitstream with the first noise band
+#define NOISE_PRE_BITS    9    ///< length of preamble
+#define NOISE_OFFSET     90    ///< subtracted from global gain, used as offset for the preamble
+
 /**
 * Long Term Prediction
 */
 typedef struct LongTermPrediction {
    int8_t present;
    int16_t lag;
-    float coef;
+    INTFLOAT coef;
    int8_t used[MAX_LTP_LONG_SFB];
 } LongTermPrediction;

@@ -169,7 +183,10 @@ typedef struct IndividualChannelStream {
    int predictor_present;
    int predictor_initialized;
    int predictor_reset_group;
+    int predictor_reset_count[31];  ///< used by encoder to count prediction resets
    uint8_t prediction_used[41];
+    uint8_t window_clipping[8]; ///< set if a certain window is near clipping
+    float clip_avoidance_factor; ///< set if any window is near clipping to the necessary atennuation factor to avoid it
 } IndividualChannelStream;

 /**
@@ -181,7 +198,8 @@ typedef struct TemporalNoiseShaping {
    int length[8][4];
    int direction[8][4];
    int order[8][4];
-    float coef[8][4][TNS_MAX_ORDER];
+    int coef_idx[8][4][TNS_MAX_ORDER];
+    INTFLOAT coef[8][4][TNS_MAX_ORDER];
 } TemporalNoiseShaping;

 /**
@@ -218,7 +236,7 @@ typedef struct ChannelCoupling {
    int ch_select[8];      /**< [0] shared list of gains; [1] list of gains for right channel;
                            *   [2] list of gains for left channel; [3] lists of gains for both channels
                            */
-    float gain[16][120];
+    INTFLOAT gain[16][120];
 } ChannelCoupling;

 /**
@@ -229,17 +247,21 @@ typedef struct SingleChannelElement {
    TemporalNoiseShaping tns;
    Pulse pulse;
    enum BandType band_type[128];                   ///< band types
+    enum BandType band_alt[128];                    ///< alternative band type (used by encoder)
    int band_type_run_end[120];                     ///< band type run end points
-    float sf[120];                                  ///< scalefactors
+    INTFLOAT sf[120];                               ///< scalefactors
    int sf_idx[128];                                ///< scalefactor indices (used by encoder)
    uint8_t zeroes[128];                            ///< band is not coded (used by encoder)
-    DECLARE_ALIGNED(32, float,   pcoeffs)[1024];    ///< coefficients for IMDCT, pristine
-    DECLARE_ALIGNED(32, float,   coeffs)[1024];     ///< coefficients for IMDCT, maybe processed
-    DECLARE_ALIGNED(32, float,   saved)[1536];      ///< overlap
-    DECLARE_ALIGNED(32, float,   ret_buf)[2048];    ///< PCM output buffer
-    DECLARE_ALIGNED(16, float,   ltp_state)[3072];  ///< time signal for LTP
+    float  is_ener[128];                            ///< Intensity stereo pos (used by encoder)
+    float pns_ener[128];                            ///< Noise energy values (used by encoder)
+    DECLARE_ALIGNED(32, INTFLOAT, pcoeffs)[1024];   ///< coefficients for IMDCT, pristine
+    DECLARE_ALIGNED(32, INTFLOAT, coeffs)[1024];    ///< coefficients for IMDCT, maybe processed
+    DECLARE_ALIGNED(32, INTFLOAT, saved)[1536];     ///< overlap
+    DECLARE_ALIGNED(32, INTFLOAT, ret_buf)[2048];   ///< PCM output buffer
+    DECLARE_ALIGNED(16, INTFLOAT, ltp_state)[3072]; ///< time signal for LTP
+    DECLARE_ALIGNED(32, AAC_FLOAT, prcoeffs)[1024]; ///< Main prediction coefs (used by encoder)
    PredictorState predictor_state[MAX_PREDICTORS];
-    float *ret;                                     ///< PCM output
+    INTFLOAT *ret;                                  ///< PCM output
 } SingleChannelElement;

 /**
@@ -250,7 +272,9 @@ typedef struct ChannelElement {
    // CPE specific
    int common_window;        ///< Set if channels share a common 'IndividualChannelStream' in bitstream.
    int     ms_mode;          ///< Signals mid/side stereo flags coding mode (used by encoder)
+    uint8_t is_mode;          ///< Set if any bands have been encoded using intensity stereo (used by encoder)
    uint8_t ms_mask[128];     ///< Set if mid/side stereo is used for each scalefactor window band
+    uint8_t is_mask[128];     ///< Set if intensity stereo is used (used by encoder)
    // shared
    SingleChannelElement ch[2];
    // CCE specific
@@ -284,7 +308,7 @@ struct AACContext {
     * (We do not want to have these on the stack.)
     * @{
     */
-    DECLARE_ALIGNED(32, float, buf_mdct)[1024];
+    DECLARE_ALIGNED(32, INTFLOAT, buf_mdct)[1024];
    /** @} */

    /**
@@ -295,8 +319,12 @@ struct AACContext {
    FFTContext mdct_small;
    FFTContext mdct_ld;
    FFTContext mdct_ltp;
+#if USE_FIXED
+    AVFixedDSPContext *fdsp;
+#else
    IMDCT15Context *mdct480;
    AVFloatDSPContext *fdsp;
+#endif /* USE_FIXED */
    int random_state;
    /** @} */

@@ -316,7 +344,7 @@ struct AACContext {
    int dmono_mode;      ///< 0->not dmono, 1->use first channel, 2->use second channel
    /** @} */

-    DECLARE_ALIGNED(32, float, temp)[128];
+    DECLARE_ALIGNED(32, INTFLOAT, temp)[128];

    OutputConfiguration oc[2];
    int warned_num_aac_frames;
@@ -324,11 +352,13 @@ struct AACContext {
    /* aacdec functions pointers */
    void (*imdct_and_windowing)(AACContext *ac, SingleChannelElement *sce);
    void (*apply_ltp)(AACContext *ac, SingleChannelElement *sce);
-    void (*apply_tns)(float coef[1024], TemporalNoiseShaping *tns,
+    void (*apply_tns)(INTFLOAT coef[1024], TemporalNoiseShaping *tns,
                      IndividualChannelStream *ics, int decode);
-    void (*windowing_and_mdct_ltp)(AACContext *ac, float *out,
-                                   float *in, IndividualChannelStream *ics);
+    void (*windowing_and_mdct_ltp)(AACContext *ac, INTFLOAT *out,
+                                   INTFLOAT *in, IndividualChannelStream *ics);
    void (*update_ltp)(AACContext *ac, SingleChannelElement *sce);
+    void (*vector_pow43)(int *coefs, int len);
+    void (*subband_scale)(int *dst, int *src, int scale, int offset, int len);

 };

@@ -89,7 +89,7 @@ static int aac_adtstoasc_filter(AVBitStreamFilterContext *bsfc,
        }
        av_free(avctx->extradata);
        avctx->extradata_size = 2 + pce_size;
-        avctx->extradata = av_mallocz(avctx->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
+        avctx->extradata = av_mallocz(avctx->extradata_size + AV_INPUT_BUFFER_PADDING_SIZE);
        if (!avctx->extradata) {
            avctx->extradata_size = 0;
            return AVERROR(ENOMEM);
@@ -0,0 +1,114 @@
+/*
+ * AAC defines
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_AAC_DEFINES_H
+#define AVCODEC_AAC_DEFINES_H
+
+#ifndef USE_FIXED
+#define USE_FIXED 0
+#endif
+
+#if USE_FIXED
+
+#include "libavutil/softfloat.h"
+
+#define FFT_FLOAT    0
+#define FFT_FIXED_32 1
+
+#define AAC_RENAME(x)       x ## _fixed
+#define AAC_RENAME_32(x)    x ## _fixed_32
+#define INTFLOAT int
+#define INT64FLOAT          int64_t
+#define SHORTFLOAT int16_t
+#define AAC_FLOAT SoftFloat
+#define AAC_SIGNE           int
+#define FIXR(a)             ((int)((a) * 1 + 0.5))
+#define FIXR10(a)           ((int)((a) * 1024.0 + 0.5))
+#define Q23(a)              (int)((a) * 8388608.0 + 0.5)
+#define Q30(x)              (int)((x)*1073741824.0 + 0.5)
+#define Q31(x)              (int)((x)*2147483648.0 + 0.5)
+#define RANGE15(x)          x
+#define GET_GAIN(x, y)      (-(y) * (1 << (x))) + 1024
+#define AAC_MUL16(x, y)     (int)(((int64_t)(x) * (y) + 0x8000) >> 16)
+#define AAC_MUL26(x, y)     (int)(((int64_t)(x) * (y) + 0x2000000) >> 26)
+#define AAC_MUL30(x, y)     (int)(((int64_t)(x) * (y) + 0x20000000) >> 30)
+#define AAC_MUL31(x, y)     (int)(((int64_t)(x) * (y) + 0x40000000) >> 31)
+#define AAC_MADD28(x, y, a, b) (int)((((int64_t)(x) * (y)) + \
+                                      ((int64_t)(a) * (b)) + \
+                                        0x8000000) >> 28)
+#define AAC_MADD30(x, y, a, b) (int)((((int64_t)(x) * (y)) + \
+                                      ((int64_t)(a) * (b)) + \
+                                        0x20000000) >> 30)
+#define AAC_MADD30_V8(x, y, a, b, c, d, e, f) (int)((((int64_t)(x) * (y)) + \
+                                                     ((int64_t)(a) * (b)) + \
+                                                     ((int64_t)(c) * (d)) + \
+                                                     ((int64_t)(e) * (f)) + \
+                                                       0x20000000) >> 30)
+#define AAC_MSUB30(x, y, a, b) (int)((((int64_t)(x) * (y)) - \
+                                      ((int64_t)(a) * (b)) + \
+                                        0x20000000) >> 30)
+#define AAC_MSUB30_V8(x, y, a, b, c, d, e, f) (int)((((int64_t)(x) * (y)) + \
+                                                     ((int64_t)(a) * (b)) - \
+                                                     ((int64_t)(c) * (d)) - \
+                                                     ((int64_t)(e) * (f)) + \
+                                                       0x20000000) >> 30)
+#define AAC_MSUB31_V3(x, y, z)    (int)((((int64_t)(x) * (z)) - \
+                                      ((int64_t)(y) * (z)) + \
+                                        0x40000000) >> 31)
+#define AAC_HALF_SUM(x, y)  (((x) >> 1) + ((y) >> 1))
+#define AAC_SRA_R(x, y)     (int)(((x) + (1 << ((y) - 1))) >> (y))
+
+#else
+
+#define FFT_FLOAT    1
+#define FFT_FIXED_32 0
+
+#define AAC_RENAME(x)       x
+#define AAC_RENAME_32(x)    x
+#define INTFLOAT float
+#define INT64FLOAT          float
+#define SHORTFLOAT float
+#define AAC_FLOAT float
+#define AAC_SIGNE           unsigned
+#define FIXR(x)             ((float)(x))
+#define FIXR10(x)           ((float)(x))
+#define Q23(x)              x
+#define Q30(x)              x
+#define Q31(x)              x
+#define RANGE15(x)          (32768.0 * (x))
+#define GET_GAIN(x, y)      powf((x), -(y))
+#define AAC_MUL16(x, y)     ((x) * (y))
+#define AAC_MUL26(x, y)     ((x) * (y))
+#define AAC_MUL30(x, y)     ((x) * (y))
+#define AAC_MUL31(x, y)     ((x) * (y))
+#define AAC_MADD28(x, y, a, b) ((x) * (y) + (a) * (b))
+#define AAC_MADD30(x, y, a, b) ((x) * (y) + (a) * (b))
+#define AAC_MADD30_V8(x, y, a, b, c, d, e, f) ((x) * (y) + (a) * (b) + \
+                                               (c) * (d) + (e) * (f))
+#define AAC_MSUB30(x, y, a, b) ((x) * (y) - (a) * (b))
+#define AAC_MSUB30_V8(x, y, a, b, c, d, e, f) ((x) * (y) + (a) * (b) - \
+                                               (c) * (d) - (e) * (f))
+#define AAC_MSUB31_V3(x, y, z)    ((x) - (y)) * (z)
+#define AAC_HALF_SUM(x, y)  ((x) + (y)) * 0.5f
+#define AAC_SRA_R(x, y)     (x)
+
+#endif /* USE_FIXED */
+
+#endif /* AVCODEC_AAC_DEFINES_H */
@@ -34,7 +34,7 @@ static int aac_sync(uint64_t state, AACAC3ParseContext *hdr_info,
    int size;
    union {
        uint64_t u64;
-        uint8_t  u8[8 + FF_INPUT_BUFFER_PADDING_SIZE];
+        uint8_t  u8[8 + AV_INPUT_BUFFER_PADDING_SIZE];
    } tmp;

    tmp.u64 = av_be2ne64(state);
@@ -33,5 +33,7 @@ int main(void)

    WRITE_ARRAY("const", float, ff_aac_pow2sf_tab);

+    WRITE_ARRAY("const", float, ff_aac_pow34sf_tab);
+
    return 0;
 }
@@ -30,12 +30,15 @@
 #else
 #include "libavutil/mathematics.h"
 float ff_aac_pow2sf_tab[428];
+float ff_aac_pow34sf_tab[428];

 av_cold void ff_aac_tableinit(void)
 {
    int i;
-    for (i = 0; i < 428; i++)
+    for (i = 0; i < 428; i++) {
        ff_aac_pow2sf_tab[i] = pow(2, (i - POW_SF2_ZERO) / 4.0);
+        ff_aac_pow34sf_tab[i] = pow(ff_aac_pow2sf_tab[i], 3.0/4.0);
+    }
 }
 #endif /* CONFIG_HARDCODED_TABLES */

@@ -28,9 +28,11 @@
 #if CONFIG_HARDCODED_TABLES
 #define ff_aac_tableinit()
 extern const float ff_aac_pow2sf_tab[428];
+extern const float ff_aac_pow34sf_tab[428];
 #else
 void ff_aac_tableinit(void);
 extern       float ff_aac_pow2sf_tab[428];
+extern       float ff_aac_pow34sf_tab[428];
 #endif /* CONFIG_HARDCODED_TABLES */

 #endif /* AVCODEC_AAC_TABLEGEN_DECL_H */
@@ -39,263 +39,26 @@
 #include "aac.h"
 #include "aacenc.h"
 #include "aactab.h"
+#include "aacenctab.h"
+#include "aacenc_utils.h"
+#include "aacenc_quantization.h"
+#include "aac_tablegen_decl.h"

-/** bits needed to code codebook run value for long windows */
-static const uint8_t run_value_bits_long[64] = {
-     5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,
-     5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5, 10,
-    10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10,
-    10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15
-};
+#include "aacenc_is.h"
+#include "aacenc_tns.h"
+#include "aacenc_pred.h"

-/** bits needed to code codebook run value for short windows */
-static const uint8_t run_value_bits_short[16] = {
-    3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
-};
+/** Frequency in Hz for lower limit of noise substitution **/
+#define NOISE_LOW_LIMIT 4500

-static const uint8_t * const run_value_bits[2] = {
-    run_value_bits_long, run_value_bits_short
-};
+/* Energy spread threshold value below which no PNS is used, this corresponds to
+ * typically around 17Khz, after which PNS usage decays ending at 19Khz */
+#define NOISE_SPREAD_THRESHOLD 0.5f

-
-/**
- * Quantize one coefficient.
- * @return absolute value of the quantized coefficient
- * @see 3GPP TS26.403 5.6.2 "Scalefactor determination"
- */
-static av_always_inline int quant(float coef, const float Q)
-{
-    float a = coef * Q;
-    return sqrtf(a * sqrtf(a)) + 0.4054;
-}
-
-static void quantize_bands(int *out, const float *in, const float *scaled,
-                           int size, float Q34, int is_signed, int maxval)
-{
-    int i;
-    double qc;
-    for (i = 0; i < size; i++) {
-        qc = scaled[i] * Q34;
-        out[i] = (int)FFMIN(qc + 0.4054, (double)maxval);
-        if (is_signed && in[i] < 0.0f) {
-            out[i] = -out[i];
-        }
-    }
-}
-
-static void abs_pow34_v(float *out, const float *in, const int size)
-{
-#ifndef USE_REALLY_FULL_SEARCH
-    int i;
-    for (i = 0; i < size; i++) {
-        float a = fabsf(in[i]);
-        out[i] = sqrtf(a * sqrtf(a));
-    }
-#endif /* USE_REALLY_FULL_SEARCH */
-}
-
-static const uint8_t aac_cb_range [12] = {0, 3, 3, 3, 3, 9, 9, 8, 8, 13, 13, 17};
-static const uint8_t aac_cb_maxval[12] = {0, 1, 1, 2, 2, 4, 4, 7, 7, 12, 12, 16};
-
-/**
- * Calculate rate distortion cost for quantizing with given codebook
- *
- * @return quantization distortion
- */
-static av_always_inline float quantize_and_encode_band_cost_template(
-                                struct AACEncContext *s,
-                                PutBitContext *pb, const float *in,
-                                const float *scaled, int size, int scale_idx,
-                                int cb, const float lambda, const float uplim,
-                                int *bits, int BT_ZERO, int BT_UNSIGNED,
-                                int BT_PAIR, int BT_ESC)
-{
-    const int q_idx = POW_SF2_ZERO - scale_idx + SCALE_ONE_POS - SCALE_DIV_512;
-    const float Q   = ff_aac_pow2sf_tab [q_idx];
-    const float Q34 = ff_aac_pow34sf_tab[q_idx];
-    const float IQ  = ff_aac_pow2sf_tab [POW_SF2_ZERO + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
-    const float CLIPPED_ESCAPE = 165140.0f*IQ;
-    int i, j;
-    float cost = 0;
-    const int dim = BT_PAIR ? 2 : 4;
-    int resbits = 0;
-    const int range  = aac_cb_range[cb];
-    const int maxval = aac_cb_maxval[cb];
-    int off;
-
-    if (BT_ZERO) {
-        for (i = 0; i < size; i++)
-            cost += in[i]*in[i];
-        if (bits)
-            *bits = 0;
-        return cost * lambda;
-    }
-    if (!scaled) {
-        abs_pow34_v(s->scoefs, in, size);
-        scaled = s->scoefs;
-    }
-    quantize_bands(s->qcoefs, in, scaled, size, Q34, !BT_UNSIGNED, maxval);
-    if (BT_UNSIGNED) {
-        off = 0;
-    } else {
-        off = maxval;
-    }
-    for (i = 0; i < size; i += dim) {
-        const float *vec;
-        int *quants = s->qcoefs + i;
-        int curidx = 0;
-        int curbits;
-        float rd = 0.0f;
-        for (j = 0; j < dim; j++) {
-            curidx *= range;
-            curidx += quants[j] + off;
-        }
-        curbits =  ff_aac_spectral_bits[cb-1][curidx];
-        vec     = &ff_aac_codebook_vectors[cb-1][curidx*dim];
-        if (BT_UNSIGNED) {
-            for (j = 0; j < dim; j++) {
-                float t = fabsf(in[i+j]);
-                float di;
-                if (BT_ESC && vec[j] == 64.0f) { //FIXME: slow
-                    if (t >= CLIPPED_ESCAPE) {
-                        di = t - CLIPPED_ESCAPE;
-                        curbits += 21;
-                    } else {
-                        int c = av_clip_uintp2(quant(t, Q), 13);
-                        di = t - c*cbrtf(c)*IQ;
-                        curbits += av_log2(c)*2 - 4 + 1;
-                    }
-                } else {
-                    di = t - vec[j]*IQ;
-                }
-                if (vec[j] != 0.0f)
-                    curbits++;
-                rd += di*di;
-            }
-        } else {
-            for (j = 0; j < dim; j++) {
-                float di = in[i+j] - vec[j]*IQ;
-                rd += di*di;
-            }
-        }
-        cost    += rd * lambda + curbits;
-        resbits += curbits;
-        if (cost >= uplim)
-            return uplim;
-        if (pb) {
-            put_bits(pb, ff_aac_spectral_bits[cb-1][curidx], ff_aac_spectral_codes[cb-1][curidx]);
-            if (BT_UNSIGNED)
-                for (j = 0; j < dim; j++)
-                    if (ff_aac_codebook_vectors[cb-1][curidx*dim+j] != 0.0f)
-                        put_bits(pb, 1, in[i+j] < 0.0f);
-            if (BT_ESC) {
-                for (j = 0; j < 2; j++) {
-                    if (ff_aac_codebook_vectors[cb-1][curidx*2+j] == 64.0f) {
-                        int coef = av_clip_uintp2(quant(fabsf(in[i+j]), Q), 13);
-                        int len = av_log2(coef);
-
-                        put_bits(pb, len - 4 + 1, (1 << (len - 4 + 1)) - 2);
-                        put_bits(pb, len, coef & ((1 << len) - 1));
-                    }
-                }
-            }
-        }
-    }
-
-    if (bits)
-        *bits = resbits;
-    return cost;
-}
-
-#define QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NAME, BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC) \
-static float quantize_and_encode_band_cost_ ## NAME(                                        \
-                                struct AACEncContext *s,                                \
-                                PutBitContext *pb, const float *in,                     \
-                                const float *scaled, int size, int scale_idx,           \
-                                int cb, const float lambda, const float uplim,          \
-                                int *bits) {                                            \
-    return quantize_and_encode_band_cost_template(                                      \
-                                s, pb, in, scaled, size, scale_idx,                     \
-                                BT_ESC ? ESC_BT : cb, lambda, uplim, bits,              \
-                                BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC);                 \
-}
-
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ZERO,  1, 0, 0, 0)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SQUAD, 0, 0, 0, 0)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UQUAD, 0, 1, 0, 0)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SPAIR, 0, 0, 1, 0)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UPAIR, 0, 1, 1, 0)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ESC,   0, 1, 1, 1)
-
-static float (*const quantize_and_encode_band_cost_arr[])(
-                                struct AACEncContext *s,
-                                PutBitContext *pb, const float *in,
-                                const float *scaled, int size, int scale_idx,
-                                int cb, const float lambda, const float uplim,
-                                int *bits) = {
-    quantize_and_encode_band_cost_ZERO,
-    quantize_and_encode_band_cost_SQUAD,
-    quantize_and_encode_band_cost_SQUAD,
-    quantize_and_encode_band_cost_UQUAD,
-    quantize_and_encode_band_cost_UQUAD,
-    quantize_and_encode_band_cost_SPAIR,
-    quantize_and_encode_band_cost_SPAIR,
-    quantize_and_encode_band_cost_UPAIR,
-    quantize_and_encode_band_cost_UPAIR,
-    quantize_and_encode_band_cost_UPAIR,
-    quantize_and_encode_band_cost_UPAIR,
-    quantize_and_encode_band_cost_ESC,
-};
-
-#define quantize_and_encode_band_cost(                                  \
-                                s, pb, in, scaled, size, scale_idx, cb, \
-                                lambda, uplim, bits)                    \
-    quantize_and_encode_band_cost_arr[cb](                              \
-                                s, pb, in, scaled, size, scale_idx, cb, \
-                                lambda, uplim, bits)
-
-static float quantize_band_cost(struct AACEncContext *s, const float *in,
-                                const float *scaled, int size, int scale_idx,
-                                int cb, const float lambda, const float uplim,
-                                int *bits)
-{
-    return quantize_and_encode_band_cost(s, NULL, in, scaled, size, scale_idx,
-                                         cb, lambda, uplim, bits);
-}
-
-static void quantize_and_encode_band(struct AACEncContext *s, PutBitContext *pb,
-                                     const float *in, int size, int scale_idx,
-                                     int cb, const float lambda)
-{
-    quantize_and_encode_band_cost(s, pb, in, NULL, size, scale_idx, cb, lambda,
-                                  INFINITY, NULL);
-}
-
-static float find_max_val(int group_len, int swb_size, const float *scaled) {
-    float maxval = 0.0f;
-    int w2, i;
-    for (w2 = 0; w2 < group_len; w2++) {
-        for (i = 0; i < swb_size; i++) {
-            maxval = FFMAX(maxval, scaled[w2*128+i]);
-        }
-    }
-    return maxval;
-}
-
-static int find_min_book(float maxval, int sf) {
-    float Q = ff_aac_pow2sf_tab[POW_SF2_ZERO - sf + SCALE_ONE_POS - SCALE_DIV_512];
-    float Q34 = sqrtf(Q * sqrtf(Q));
-    int qmaxval, cb;
-    qmaxval = maxval * Q34 + 0.4054f;
-    if      (qmaxval ==  0) cb = 0;
-    else if (qmaxval ==  1) cb = 1;
-    else if (qmaxval ==  2) cb = 3;
-    else if (qmaxval <=  4) cb = 5;
-    else if (qmaxval <=  7) cb = 7;
-    else if (qmaxval <= 12) cb = 9;
-    else                    cb = 11;
-    return cb;
-}
+/* This constant gets divided by lambda to return ~1.65 which when multiplied
+ * by the band->threshold and compared to band->energy is the boundary between
+ * excessive PNS and little PNS usage. */
+#define NOISE_LAMBDA_NUMERATOR 252.1f

 /**
 * structure used in optimal codebook search
@@ -312,7 +75,7 @@ typedef struct BandCodingPath {
 static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce,
                                     int win, int group_len, const float lambda)
 {
-    BandCodingPath path[120][12];
+    BandCodingPath path[120][CB_TOT_ALL];
    int w, swb, cb, start, size;
    int i, j;
    const int max_sfb  = sce->ics.max_sfb;
@@ -325,7 +88,7 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce

    abs_pow34_v(s->scoefs, sce->coeffs, 1024);
    start = win*128;
-    for (cb = 0; cb < 12; cb++) {
+    for (cb = 0; cb < CB_TOT_ALL; cb++) {
        path[0][cb].cost     = 0.0f;
        path[0][cb].prev_idx = -1;
        path[0][cb].run      = 0;
@@ -333,7 +96,7 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
    for (swb = 0; swb < max_sfb; swb++) {
        size = sce->ics.swb_sizes[swb];
        if (sce->zeroes[win*16 + swb]) {
-            for (cb = 0; cb < 12; cb++) {
+            for (cb = 0; cb < CB_TOT_ALL; cb++) {
                path[swb+1][cb].prev_idx = cb;
                path[swb+1][cb].cost     = path[swb][cb].cost;
                path[swb+1][cb].run      = path[swb][cb].run + 1;
@@ -343,15 +106,22 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
            int mincb = next_mincb;
            next_minrd = INFINITY;
            next_mincb = 0;
-            for (cb = 0; cb < 12; cb++) {
+            for (cb = 0; cb < CB_TOT_ALL; cb++) {
                float cost_stay_here, cost_get_here;
                float rd = 0.0f;
+                if (cb >= 12 && sce->band_type[win*16+swb] < aac_cb_out_map[cb] ||
+                    cb  < aac_cb_in_map[sce->band_type[win*16+swb]] && sce->band_type[win*16+swb] > aac_cb_out_map[cb]) {
+                    path[swb+1][cb].prev_idx = -1;
+                    path[swb+1][cb].cost     = INFINITY;
+                    path[swb+1][cb].run      = path[swb][cb].run + 1;
+                    continue;
+                }
                for (w = 0; w < group_len; w++) {
                    FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(win+w)*16+swb];
-                    rd += quantize_band_cost(s, sce->coeffs + start + w*128,
-                                             s->scoefs + start + w*128, size,
-                                             sce->sf_idx[(win+w)*16+swb], cb,
-                                             lambda / band->threshold, INFINITY, NULL);
+                    rd += quantize_band_cost(s, &sce->coeffs[start + w*128],
+                                             &s->scoefs[start + w*128], size,
+                                             sce->sf_idx[(win+w)*16+swb], aac_cb_out_map[cb],
+                                             lambda / band->threshold, INFINITY, NULL, 0);
                }
                cost_stay_here = path[swb][cb].cost + rd;
                cost_get_here  = minrd              + rd + run_bits + 4;
@@ -379,11 +149,12 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
    //convert resulting path from backward-linked list
    stack_len = 0;
    idx       = 0;
-    for (cb = 1; cb < 12; cb++)
+    for (cb = 1; cb < CB_TOT_ALL; cb++)
        if (path[max_sfb][cb].cost < path[max_sfb][idx].cost)
            idx = cb;
    ppos = max_sfb;
    while (ppos > 0) {
+        av_assert1(idx >= 0);
        cb = idx;
        stackrun[stack_len] = path[ppos][cb].run;
        stackcb [stack_len] = cb;
@@ -394,12 +165,13 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
    //perform actual band info encoding
    start = 0;
    for (i = stack_len - 1; i >= 0; i--) {
-        put_bits(&s->pb, 4, stackcb[i]);
+        cb = aac_cb_out_map[stackcb[i]];
+        put_bits(&s->pb, 4, cb);
        count = stackrun[i];
-        memset(sce->zeroes + win*16 + start, !stackcb[i], count);
+        memset(sce->zeroes + win*16 + start, !cb, count);
        //XXX: memset when band_type is also uint8_t
        for (j = 0; j < count; j++) {
-            sce->band_type[win*16 + start] =  stackcb[i];
+            sce->band_type[win*16 + start] = cb;
            start++;
        }
        while (count >= run_esc) {
@@ -413,7 +185,7 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
 static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
                                  int win, int group_len, const float lambda)
 {
-    BandCodingPath path[120][12];
+    BandCodingPath path[120][CB_TOT_ALL];
    int w, swb, cb, start, size;
    int i, j;
    const int max_sfb  = sce->ics.max_sfb;
@@ -426,7 +198,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,

    abs_pow34_v(s->scoefs, sce->coeffs, 1024);
    start = win*128;
-    for (cb = 0; cb < 12; cb++) {
+    for (cb = 0; cb < CB_TOT_ALL; cb++) {
        path[0][cb].cost     = run_bits+4;
        path[0][cb].prev_idx = -1;
        path[0][cb].run      = 0;
@@ -450,7 +222,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
            }
            next_minbits = path[swb+1][0].cost;
            next_mincb = 0;
-            for (cb = 1; cb < 12; cb++) {
+            for (cb = 1; cb < CB_TOT_ALL; cb++) {
                path[swb+1][cb].cost = 61450;
                path[swb+1][cb].prev_idx = -1;
                path[swb+1][cb].run = 0;
@@ -459,6 +231,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
            float minbits = next_minbits;
            int mincb = next_mincb;
            int startcb = sce->band_type[win*16+swb];
+            startcb = aac_cb_in_map[startcb];
            next_minbits = INFINITY;
            next_mincb = 0;
            for (cb = 0; cb < startcb; cb++) {
@@ -466,14 +239,21 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
                path[swb+1][cb].prev_idx = -1;
                path[swb+1][cb].run = 0;
            }
-            for (cb = startcb; cb < 12; cb++) {
+            for (cb = startcb; cb < CB_TOT_ALL; cb++) {
                float cost_stay_here, cost_get_here;
                float bits = 0.0f;
+                if (cb >= 12 && sce->band_type[win*16+swb] != aac_cb_out_map[cb]) {
+                    path[swb+1][cb].cost = 61450;
+                    path[swb+1][cb].prev_idx = -1;
+                    path[swb+1][cb].run = 0;
+                    continue;
+                }
                for (w = 0; w < group_len; w++) {
-                    bits += quantize_band_cost(s, sce->coeffs + start + w*128,
-                                               s->scoefs + start + w*128, size,
-                                               sce->sf_idx[(win+w)*16+swb], cb,
-                                               0, INFINITY, NULL);
+                    bits += quantize_band_cost(s, &sce->coeffs[start + w*128],
+                                               &s->scoefs[start + w*128], size,
+                                               sce->sf_idx[win*16+swb],
+                                               aac_cb_out_map[cb],
+                                               0, INFINITY, NULL, 0);
                }
                cost_stay_here = path[swb][cb].cost + bits;
                cost_get_here  = minbits            + bits + run_bits + 4;
@@ -501,7 +281,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
    //convert resulting path from backward-linked list
    stack_len = 0;
    idx       = 0;
-    for (cb = 1; cb < 12; cb++)
+    for (cb = 1; cb < CB_TOT_ALL; cb++)
        if (path[max_sfb][cb].cost < path[max_sfb][idx].cost)
            idx = cb;
    ppos = max_sfb;
@@ -517,12 +297,13 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
    //perform actual band info encoding
    start = 0;
    for (i = stack_len - 1; i >= 0; i--) {
-        put_bits(&s->pb, 4, stackcb[i]);
+        cb = aac_cb_out_map[stackcb[i]];
+        put_bits(&s->pb, 4, cb);
        count = stackrun[i];
-        memset(sce->zeroes + win*16 + start, !stackcb[i], count);
+        memset(sce->zeroes + win*16 + start, !cb, count);
        //XXX: memset when band_type is also uint8_t
        for (j = 0; j < count; j++) {
-            sce->band_type[win*16 + start] =  stackcb[i];
+            sce->band_type[win*16 + start] = cb;
            start++;
        }
        while (count >= run_esc) {
@@ -533,16 +314,6 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
    }
 }

-/** Return the minimum scalefactor where the quantized coef does not clip. */
-static av_always_inline uint8_t coef2minsf(float coef) {
-    return av_clip_uint8(log2f(coef)*4 - 69 + SCALE_ONE_POS - SCALE_DIV_512);
-}
-
-/** Return the maximum scalefactor where the quantized coef is not zero. */
-static av_always_inline uint8_t coef2maxsf(float coef) {
-    return av_clip_uint8(log2f(coef)*4 +  6 + SCALE_ONE_POS - SCALE_DIV_512);
-}
-
 typedef struct TrellisPath {
    float cost;
    int prev;
@@ -551,6 +322,43 @@ typedef struct TrellisPath {
 #define TRELLIS_STAGES 121
 #define TRELLIS_STATES (SCALE_MAX_DIFF+1)

+static void set_special_band_scalefactors(AACEncContext *s, SingleChannelElement *sce)
+{
+    int w, g, start = 0;
+    int minscaler_n = sce->sf_idx[0], minscaler_i = sce->sf_idx[0];
+    int bands = 0;
+
+    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+        start = 0;
+        for (g = 0;  g < sce->ics.num_swb; g++) {
+            if (sce->band_type[w*16+g] == INTENSITY_BT || sce->band_type[w*16+g] == INTENSITY_BT2) {
+                sce->sf_idx[w*16+g] = av_clip(ceilf(log2f(sce->is_ener[w*16+g])*2), -155, 100);
+                minscaler_i = FFMIN(minscaler_i, sce->sf_idx[w*16+g]);
+                bands++;
+            } else if (sce->band_type[w*16+g] == NOISE_BT) {
+                sce->sf_idx[w*16+g] = av_clip(4+log2f(sce->pns_ener[w*16+g])*2, -100, 155);
+                minscaler_n = FFMIN(minscaler_n, sce->sf_idx[w*16+g]);
+                bands++;
+            }
+            start += sce->ics.swb_sizes[g];
+        }
+    }
+
+    if (!bands)
+        return;
+
+    /* Clip the scalefactor indices */
+    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+        for (g = 0;  g < sce->ics.num_swb; g++) {
+            if (sce->band_type[w*16+g] == INTENSITY_BT || sce->band_type[w*16+g] == INTENSITY_BT2) {
+                sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler_i, minscaler_i + SCALE_MAX_DIFF);
+            } else if (sce->band_type[w*16+g] == NOISE_BT) {
+                sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler_n, minscaler_n + SCALE_MAX_DIFF);
+            }
+        }
+    }
+}
+
 static void search_for_quantizers_anmr(AVCodecContext *avctx, AACEncContext *s,
                                       SingleChannelElement *sce,
                                       const float lambda)
@@ -616,7 +424,7 @@ static void search_for_quantizers_anmr(AVCodecContext *avctx, AACEncContext *s,
    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
        start = w*128;
        for (g = 0; g < sce->ics.num_swb; g++) {
-            const float *coefs = sce->coeffs + start;
+            const float *coefs = &sce->coeffs[start];
            float qmin, qmax;
            int nz = 0;

@@ -655,7 +463,7 @@ static void search_for_quantizers_anmr(AVCodecContext *avctx, AACEncContext *s,
                    for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
                        FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
                        dist += quantize_band_cost(s, coefs + w2*128, s->scoefs + start + w2*128, sce->ics.swb_sizes[g],
-                                                   q + q0, cb, lambda / band->threshold, INFINITY, NULL);
+                                                   q + q0, cb, lambda / band->threshold, INFINITY, NULL, 0);
                    }
                    minrd = FFMIN(minrd, dist);

@@ -691,7 +499,7 @@ static void search_for_quantizers_anmr(AVCodecContext *avctx, AACEncContext *s,
    }
    while (idx) {
        sce->sf_idx[bandaddr[idx]] = minq + q0;
-        minq = paths[idx][minq].prev;
+        minq = FFMAX(paths[idx][minq].prev, 0);
        idx--;
    }
    //set the same quantizers inside window groups
@@ -711,7 +519,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
 {
    int start = 0, i, w, w2, g;
    int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->channels * (lambda / 120.f);
-    float dists[128] = { 0 }, uplims[128];
+    float dists[128] = { 0 }, uplims[128] = { 0 };
    float maxvals[128];
    int fflag, minscaler;
    int its  = 0;
@@ -726,10 +534,11 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
        for (g = 0;  g < sce->ics.num_swb; g++) {
            int nz = 0;
-            float uplim = 0.0f;
+            float uplim = 0.0f, energy = 0.0f;
            for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
                FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
-                uplim += band->threshold;
+                uplim  += band->threshold;
+                energy += band->energy;
                if (band->energy <= band->threshold || band->threshold == 0.0f) {
                    sce->zeroes[(w+w2)*16+g] = 1;
                    continue;
@@ -779,8 +588,8 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
            for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
                start = w*128;
                for (g = 0;  g < sce->ics.num_swb; g++) {
-                    const float *coefs = sce->coeffs + start;
-                    const float *scaled = s->scoefs + start;
+                    const float *coefs = &sce->coeffs[start];
+                    const float *scaled = &s->scoefs[start];
                    int bits = 0;
                    int cb;
                    float dist = 0.0f;
@@ -800,7 +609,8 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
                                                   cb,
                                                   1.0f,
                                                   INFINITY,
-                                                   &b);
+                                                   &b,
+                                                   0);
                        bits += b;
                    }
                    dists[w*16+g] = dist - bits;
@@ -828,6 +638,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,

        fflag = 0;
        minscaler = av_clip(minscaler, 60, 255 - SCALE_MAX_DIFF);
+
        for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
            for (g = 0; g < sce->ics.num_swb; g++) {
                int prevsc = sce->sf_idx[w*16+g];
@@ -873,7 +684,7 @@ static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
        }
    } else {
        for (w = 0; w < 8; w++) {
-            const float *coeffs = sce->coeffs + w*128;
+            const float *coeffs = &sce->coeffs[w*128];
            curband = start = 0;
            for (i = 0; i < 128; i++) {
                if (i - start >= sce->ics.swb_sizes[curband]) {
@@ -898,7 +709,7 @@ static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
        start = w*128;
        for (g = 0; g < sce->ics.num_swb; g++) {
-            float *coefs   = sce->coeffs + start;
+            float *coefs   = &sce->coeffs[start];
            const int size = sce->ics.swb_sizes[g];
            int start2 = start, end2 = start + size, peakpos = start;
            float maxval = -1, thr = 0.0f, t;
@@ -939,8 +750,8 @@ static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
        start = w*128;
        for (g = 0;  g < sce->ics.num_swb; g++) {
-            const float *coefs  = sce->coeffs + start;
-            const float *scaled = s->scoefs   + start;
+            const float *coefs  = &sce->coeffs[start];
+            const float *scaled = &s->scoefs[start];
            const int size      = sce->ics.swb_sizes[g];
            int scf, prev_scf, step;
            int min_scf = -1, max_scf = 256;
@@ -965,11 +776,12 @@ static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
                                               ESC_BT,
                                               lambda,
                                               INFINITY,
-                                               &b);
+                                               &b,
+                                               0);
                    dist -= b;
                }
                dist *= 1.0f / 512.0f / lambda;
-                quant_max = quant(maxq[w*16+g], ff_aac_pow2sf_tab[POW_SF2_ZERO - scf + SCALE_ONE_POS - SCALE_DIV_512]);
+                quant_max = quant(maxq[w*16+g], ff_aac_pow2sf_tab[POW_SF2_ZERO - scf + SCALE_ONE_POS - SCALE_DIV_512], ROUND_STANDARD);
                if (quant_max >= 8191) { // too much, return to the previous quantizer
                    sce->sf_idx[w*16+g] = prev_scf;
                    break;
@@ -1049,17 +861,50 @@ static void search_for_quantizers_fast(AVCodecContext *avctx, AACEncContext *s,
                sce->sf_idx[(w+w2)*16+g] = sce->sf_idx[w*16+g];
 }

-static void search_for_ms(AACEncContext *s, ChannelElement *cpe,
-                          const float lambda)
+static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce)
+{
+    int start = 0, w, w2, g;
+    const float lambda = s->lambda;
+    const float freq_mult = avctx->sample_rate/(1024.0f/sce->ics.num_windows)/2.0f;
+    const float spread_threshold = NOISE_SPREAD_THRESHOLD*(lambda/120.f);
+    const float thr_mult = NOISE_LAMBDA_NUMERATOR/lambda;
+
+    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+        start = 0;
+        for (g = 0;  g < sce->ics.num_swb; g++) {
+            if (start*freq_mult > NOISE_LOW_LIMIT*(lambda/170.0f)) {
+                float energy = 0.0f, threshold = 0.0f, spread = 0.0f;
+                for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
+                    FFPsyBand *band = &s->psy.ch[s->cur_channel+0].psy_bands[(w+w2)*16+g];
+                    energy += band->energy;
+                    threshold += band->threshold;
+                    spread += band->spread;
+                }
+                if (spread > spread_threshold*sce->ics.group_len[w] &&
+                    ((sce->zeroes[w*16+g] && energy >= threshold) ||
+                    energy < threshold*thr_mult*sce->ics.group_len[w])) {
+                    sce->band_type[w*16+g] = NOISE_BT;
+                    sce->pns_ener[w*16+g] = energy / sce->ics.group_len[w];
+                    sce->zeroes[w*16+g] = 0;
+                }
+            }
+            start += sce->ics.swb_sizes[g];
+        }
+    }
+}
+
+static void search_for_ms(AACEncContext *s, ChannelElement *cpe)
 {
    int start = 0, i, w, w2, g;
    float M[128], S[128];
    float *L34 = s->scoefs, *R34 = s->scoefs + 128, *M34 = s->scoefs + 128*2, *S34 = s->scoefs + 128*3;
+    const float lambda = s->lambda;
    SingleChannelElement *sce0 = &cpe->ch[0];
    SingleChannelElement *sce1 = &cpe->ch[1];
    if (!cpe->common_window)
        return;
    for (w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]) {
+        start = 0;
        for (g = 0;  g < sce0->ics.num_swb; g++) {
            if (!cpe->ch[0].zeroes[w*16+g] && !cpe->ch[1].zeroes[w*16+g]) {
                float dist1 = 0.0f, dist2 = 0.0f;
@@ -1069,39 +914,39 @@ static void search_for_ms(AACEncContext *s, ChannelElement *cpe,
                    float minthr = FFMIN(band0->threshold, band1->threshold);
                    float maxthr = FFMAX(band0->threshold, band1->threshold);
                    for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
-                        M[i] = (sce0->pcoeffs[start+w2*128+i]
-                              + sce1->pcoeffs[start+w2*128+i]) * 0.5;
+                        M[i] = (sce0->coeffs[start+(w+w2)*128+i]
+                              + sce1->coeffs[start+(w+w2)*128+i]) * 0.5;
                        S[i] =  M[i]
-                              - sce1->pcoeffs[start+w2*128+i];
+                              - sce1->coeffs[start+(w+w2)*128+i];
                    }
-                    abs_pow34_v(L34, sce0->coeffs+start+w2*128, sce0->ics.swb_sizes[g]);
-                    abs_pow34_v(R34, sce1->coeffs+start+w2*128, sce0->ics.swb_sizes[g]);
+                    abs_pow34_v(L34, sce0->coeffs+start+(w+w2)*128, sce0->ics.swb_sizes[g]);
+                    abs_pow34_v(R34, sce1->coeffs+start+(w+w2)*128, sce0->ics.swb_sizes[g]);
                    abs_pow34_v(M34, M,                         sce0->ics.swb_sizes[g]);
                    abs_pow34_v(S34, S,                         sce0->ics.swb_sizes[g]);
-                    dist1 += quantize_band_cost(s, sce0->coeffs + start + w2*128,
+                    dist1 += quantize_band_cost(s, &sce0->coeffs[start + (w+w2)*128],
                                                L34,
                                                sce0->ics.swb_sizes[g],
                                                sce0->sf_idx[(w+w2)*16+g],
                                                sce0->band_type[(w+w2)*16+g],
-                                                lambda / band0->threshold, INFINITY, NULL);
-                    dist1 += quantize_band_cost(s, sce1->coeffs + start + w2*128,
+                                                lambda / band0->threshold, INFINITY, NULL, 0);
+                    dist1 += quantize_band_cost(s, &sce1->coeffs[start + (w+w2)*128],
                                                R34,
                                                sce1->ics.swb_sizes[g],
                                                sce1->sf_idx[(w+w2)*16+g],
                                                sce1->band_type[(w+w2)*16+g],
-                                                lambda / band1->threshold, INFINITY, NULL);
+                                                lambda / band1->threshold, INFINITY, NULL, 0);
                    dist2 += quantize_band_cost(s, M,
                                                M34,
                                                sce0->ics.swb_sizes[g],
                                                sce0->sf_idx[(w+w2)*16+g],
                                                sce0->band_type[(w+w2)*16+g],
-                                                lambda / maxthr, INFINITY, NULL);
+                                                lambda / maxthr, INFINITY, NULL, 0);
                    dist2 += quantize_band_cost(s, S,
                                                S34,
                                                sce1->ics.swb_sizes[g],
                                                sce1->sf_idx[(w+w2)*16+g],
                                                sce1->band_type[(w+w2)*16+g],
-                                                lambda / minthr, INFINITY, NULL);
+                                                lambda / minthr, INFINITY, NULL, 0);
                }
                cpe->ms_mask[w*16+g] = dist2 < dist1;
            }
@@ -1115,24 +960,64 @@ AACCoefficientsEncoder ff_aac_coders[AAC_CODER_NB] = {
        search_for_quantizers_faac,
        encode_window_bands_info,
        quantize_and_encode_band,
+        ff_aac_encode_tns_info,
+        ff_aac_encode_main_pred,
+        ff_aac_adjust_common_prediction,
+        ff_aac_apply_main_pred,
+        ff_aac_apply_tns,
+        set_special_band_scalefactors,
+        search_for_pns,
+        ff_aac_search_for_tns,
        search_for_ms,
+        ff_aac_search_for_is,
+        ff_aac_search_for_pred,
    },
    [AAC_CODER_ANMR] = {
        search_for_quantizers_anmr,
        encode_window_bands_info,
        quantize_and_encode_band,
+        ff_aac_encode_tns_info,
+        ff_aac_encode_main_pred,
+        ff_aac_adjust_common_prediction,
+        ff_aac_apply_main_pred,
+        ff_aac_apply_tns,
+        set_special_band_scalefactors,
+        search_for_pns,
+        ff_aac_search_for_tns,
        search_for_ms,
+        ff_aac_search_for_is,
+        ff_aac_search_for_pred,
    },
    [AAC_CODER_TWOLOOP] = {
        search_for_quantizers_twoloop,
        codebook_trellis_rate,
        quantize_and_encode_band,
+        ff_aac_encode_tns_info,
+        ff_aac_encode_main_pred,
+        ff_aac_adjust_common_prediction,
+        ff_aac_apply_main_pred,
+        ff_aac_apply_tns,
+        set_special_band_scalefactors,
+        search_for_pns,
+        ff_aac_search_for_tns,
        search_for_ms,
+        ff_aac_search_for_is,
+        ff_aac_search_for_pred,
    },
    [AAC_CODER_FAST] = {
        search_for_quantizers_fast,
        encode_window_bands_info,
        quantize_and_encode_band,
+        ff_aac_encode_tns_info,
+        ff_aac_encode_main_pred,
+        ff_aac_adjust_common_prediction,
+        ff_aac_apply_main_pred,
+        ff_aac_apply_tns,
+        set_special_band_scalefactors,
+        search_for_pns,
+        ff_aac_search_for_tns,
        search_for_ms,
+        ff_aac_search_for_is,
+        ff_aac_search_for_pred,
    },
 };
@@ -0,0 +1,458 @@
+/*
+ * Copyright (c) 2013
+ *      MIPS Technologies, Inc., California.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ *    notice, this list of conditions and the following disclaimer in the
+ *    documentation and/or other materials provided with the distribution.
+ * 3. Neither the name of the MIPS Technologies, Inc., nor the names of its
+ *    contributors may be used to endorse or promote products derived from
+ *    this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ * ARE DISCLAIMED.  IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE
+ * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+ * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ *
+ * AAC decoder fixed-point implementation
+ *
+ * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
+ * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC decoder
+ * @author Oded Shimon  ( ods15 ods15 dyndns org )
+ * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
+ *
+ * Fixed point implementation
+ * @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
+ */
+
+#define FFT_FLOAT 0
+#define FFT_FIXED_32 1
+#define USE_FIXED 1
+
+#include "libavutil/fixed_dsp.h"
+#include "libavutil/opt.h"
+#include "avcodec.h"
+#include "internal.h"
+#include "get_bits.h"
+#include "fft.h"
+#include "lpc.h"
+#include "kbdwin.h"
+#include "sinewin.h"
+
+#include "aac.h"
+#include "aactab.h"
+#include "aacdectab.h"
+#include "cbrt_tablegen.h"
+#include "sbr.h"
+#include "aacsbr.h"
+#include "mpeg4audio.h"
+#include "aacadtsdec.h"
+#include "libavutil/intfloat.h"
+
+#include <math.h>
+#include <string.h>
+
+static av_always_inline void reset_predict_state(PredictorState *ps)
+{
+    ps->r0.mant   = 0;
+    ps->r0.exp   = 0;
+    ps->r1.mant   = 0;
+    ps->r1.exp   = 0;
+    ps->cor0.mant = 0;
+    ps->cor0.exp = 0;
+    ps->cor1.mant = 0;
+    ps->cor1.exp = 0;
+    ps->var0.mant = 0x20000000;
+    ps->var0.exp = 1;
+    ps->var1.mant = 0x20000000;
+    ps->var1.exp = 1;
+}
+
+static const int exp2tab[4] = { Q31(1.0000000000/2), Q31(1.1892071150/2), Q31(1.4142135624/2), Q31(1.6817928305/2) };  // 2^0, 2^0.25, 2^0.5, 2^0.75
+
+static inline int *DEC_SPAIR(int *dst, unsigned idx)
+{
+    dst[0] = (idx & 15) - 4;
+    dst[1] = (idx >> 4 & 15) - 4;
+
+    return dst + 2;
+}
+
+static inline int *DEC_SQUAD(int *dst, unsigned idx)
+{
+    dst[0] = (idx & 3) - 1;
+    dst[1] = (idx >> 2 & 3) - 1;
+    dst[2] = (idx >> 4 & 3) - 1;
+    dst[3] = (idx >> 6 & 3) - 1;
+
+    return dst + 4;
+}
+
+static inline int *DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
+{
+    dst[0] = (idx & 15) * (1 - (sign & 0xFFFFFFFE));
+    dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) * 2));
+
+    return dst + 2;
+}
+
+static inline int *DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
+{
+    unsigned nz = idx >> 12;
+
+    dst[0] = (idx & 3) * (1 + (((int)sign >> 31) * 2));
+    sign <<= nz & 1;
+    nz >>= 1;
+    dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) * 2));
+    sign <<= nz & 1;
+    nz >>= 1;
+    dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) * 2));
+    sign <<= nz & 1;
+    nz >>= 1;
+    dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) * 2));
+
+    return dst + 4;
+}
+
+static void vector_pow43(int *coefs, int len)
+{
+    int i, coef;
+
+    for (i=0; i<len; i++) {
+        coef = coefs[i];
+        if (coef < 0)
+            coef = -(int)cbrt_tab[-coef];
+        else
+            coef = (int)cbrt_tab[coef];
+        coefs[i] = coef;
+    }
+}
+
+static void subband_scale(int *dst, int *src, int scale, int offset, int len)
+{
+    int ssign = scale < 0 ? -1 : 1;
+    int s = FFABS(scale);
+    unsigned int round;
+    int i, out, c = exp2tab[s & 3];
+
+    s = offset - (s >> 2);
+
+    if (s > 31) {
+        for (i=0; i<len; i++) {
+            dst[i] = 0;
+        }
+    } else if (s > 0) {
+        round = 1 << (s-1);
+        for (i=0; i<len; i++) {
+            out = (int)(((int64_t)src[i] * c) >> 32);
+            dst[i] = ((int)(out+round) >> s) * ssign;
+        }
+    } else if (s > -32) {
+        s = s + 32;
+        round = 1 << (s-1);
+        for (i=0; i<len; i++) {
+            out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
+            dst[i] = out * (unsigned)ssign;
+        }
+    } else {
+        av_log(NULL, AV_LOG_ERROR, "Overflow in subband_scale()\n");
+    }
+}
+
+static void noise_scale(int *coefs, int scale, int band_energy, int len)
+{
+    int ssign = scale < 0 ? -1 : 1;
+    int s = FFABS(scale);
+    unsigned int round;
+    int i, out, c = exp2tab[s & 3];
+    int nlz = 0;
+
+    while (band_energy > 0x7fff) {
+        band_energy >>= 1;
+        nlz++;
+    }
+    c /= band_energy;
+    s = 21 + nlz - (s >> 2);
+
+    if (s > 31) {
+        for (i=0; i<len; i++) {
+            coefs[i] = 0;
+        }
+    } else if (s >= 0) {
+        round = s ? 1 << (s-1) : 0;
+        for (i=0; i<len; i++) {
+            out = (int)(((int64_t)coefs[i] * c) >> 32);
+            coefs[i] = ((int)(out+round) >> s) * ssign;
+        }
+    }
+    else {
+        s = s + 32;
+        round = 1 << (s-1);
+        for (i=0; i<len; i++) {
+            out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
+            coefs[i] = out * ssign;
+        }
+    }
+}
+
+static av_always_inline SoftFloat flt16_round(SoftFloat pf)
+{
+    SoftFloat tmp;
+    int s;
+
+    tmp.exp = pf.exp;
+    s = pf.mant >> 31;
+    tmp.mant = (pf.mant ^ s) - s;
+    tmp.mant = (tmp.mant + 0x00200000U) & 0xFFC00000U;
+    tmp.mant = (tmp.mant ^ s) - s;
+
+    return tmp;
+}
+
+static av_always_inline SoftFloat flt16_even(SoftFloat pf)
+{
+    SoftFloat tmp;
+    int s;
+
+    tmp.exp = pf.exp;
+    s = pf.mant >> 31;
+    tmp.mant = (pf.mant ^ s) - s;
+    tmp.mant = (tmp.mant + 0x001FFFFFU + (tmp.mant & 0x00400000U >> 16)) & 0xFFC00000U;
+    tmp.mant = (tmp.mant ^ s) - s;
+
+    return tmp;
+}
+
+static av_always_inline SoftFloat flt16_trunc(SoftFloat pf)
+{
+    SoftFloat pun;
+    int s;
+
+    pun.exp = pf.exp;
+    s = pf.mant >> 31;
+    pun.mant = (pf.mant ^ s) - s;
+    pun.mant = pun.mant & 0xFFC00000U;
+    pun.mant = (pun.mant ^ s) - s;
+
+    return pun;
+}
+
+static av_always_inline void predict(PredictorState *ps, int *coef,
+                                     int output_enable)
+{
+    const SoftFloat a     = { 1023410176, 0 };  // 61.0 / 64
+    const SoftFloat alpha = {  973078528, 0 };  // 29.0 / 32
+    SoftFloat e0, e1;
+    SoftFloat pv;
+    SoftFloat k1, k2;
+    SoftFloat   r0 = ps->r0,     r1 = ps->r1;
+    SoftFloat cor0 = ps->cor0, cor1 = ps->cor1;
+    SoftFloat var0 = ps->var0, var1 = ps->var1;
+    SoftFloat tmp;
+
+    if (var0.exp > 1 || (var0.exp == 1 && var0.mant > 0x20000000)) {
+        k1 = av_mul_sf(cor0, flt16_even(av_div_sf(a, var0)));
+    }
+    else {
+        k1.mant = 0;
+        k1.exp = 0;
+    }
+
+    if (var1.exp > 1 || (var1.exp == 1 && var1.mant > 0x20000000)) {
+        k2 = av_mul_sf(cor1, flt16_even(av_div_sf(a, var1)));
+    }
+    else {
+        k2.mant = 0;
+        k2.exp = 0;
+    }
+
+    tmp = av_mul_sf(k1, r0);
+    pv = flt16_round(av_add_sf(tmp, av_mul_sf(k2, r1)));
+    if (output_enable) {
+        int shift = 28 - pv.exp;
+
+        if (shift < 31) {
+            if (shift > 0) {
+                *coef += (unsigned)((pv.mant + (1 << (shift - 1))) >> shift);
+            } else
+                *coef += (unsigned)pv.mant << -shift;
+        }
+    }
+
+    e0 = av_int2sf(*coef, 2);
+    e1 = av_sub_sf(e0, tmp);
+
+    ps->cor1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor1), av_mul_sf(r1, e1)));
+    tmp = av_add_sf(av_mul_sf(r1, r1), av_mul_sf(e1, e1));
+    tmp.exp--;
+    ps->var1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var1), tmp));
+    ps->cor0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor0), av_mul_sf(r0, e0)));
+    tmp = av_add_sf(av_mul_sf(r0, r0), av_mul_sf(e0, e0));
+    tmp.exp--;
+    ps->var0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var0), tmp));
+
+    ps->r1 = flt16_trunc(av_mul_sf(a, av_sub_sf(r0, av_mul_sf(k1, e0))));
+    ps->r0 = flt16_trunc(av_mul_sf(a, e0));
+}
+
+
+static const int cce_scale_fixed[8] = {
+    Q30(1.0),          //2^(0/8)
+    Q30(1.0905077327), //2^(1/8)
+    Q30(1.1892071150), //2^(2/8)
+    Q30(1.2968395547), //2^(3/8)
+    Q30(1.4142135624), //2^(4/8)
+    Q30(1.5422108254), //2^(5/8)
+    Q30(1.6817928305), //2^(6/8)
+    Q30(1.8340080864), //2^(7/8)
+};
+
+/**
+ * Apply dependent channel coupling (applied before IMDCT).
+ *
+ * @param   index   index into coupling gain array
+ */
+static void apply_dependent_coupling_fixed(AACContext *ac,
+                                     SingleChannelElement *target,
+                                     ChannelElement *cce, int index)
+{
+    IndividualChannelStream *ics = &cce->ch[0].ics;
+    const uint16_t *offsets = ics->swb_offset;
+    int *dest = target->coeffs;
+    const int *src = cce->ch[0].coeffs;
+    int g, i, group, k, idx = 0;
+    if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
+        av_log(ac->avctx, AV_LOG_ERROR,
+               "Dependent coupling is not supported together with LTP\n");
+        return;
+    }
+    for (g = 0; g < ics->num_window_groups; g++) {
+        for (i = 0; i < ics->max_sfb; i++, idx++) {
+            if (cce->ch[0].band_type[idx] != ZERO_BT) {
+                const int gain = cce->coup.gain[index][idx];
+                int shift, round, c, tmp;
+
+                if (gain < 0) {
+                    c = -cce_scale_fixed[-gain & 7];
+                    shift = (-gain-1024) >> 3;
+                }
+                else {
+                    c = cce_scale_fixed[gain & 7];
+                    shift = (gain-1024) >> 3;
+                }
+
+                if (shift < -31) {
+                    // Nothing to do
+                } else if (shift < 0) {
+                    shift = -shift;
+                    round = 1 << (shift - 1);
+
+                    for (group = 0; group < ics->group_len[g]; group++) {
+                        for (k = offsets[i]; k < offsets[i + 1]; k++) {
+                            tmp = (int)(((int64_t)src[group * 128 + k] * c + \
+                                       (int64_t)0x1000000000) >> 37);
+                            dest[group * 128 + k] += (tmp + (int64_t)round) >> shift;
+                        }
+                    }
+                }
+                else {
+                    for (group = 0; group < ics->group_len[g]; group++) {
+                        for (k = offsets[i]; k < offsets[i + 1]; k++) {
+                            tmp = (int)(((int64_t)src[group * 128 + k] * c + \
+                                        (int64_t)0x1000000000) >> 37);
+                            dest[group * 128 + k] += tmp * (1U << shift);
+                        }
+                    }
+                }
+            }
+        }
+        dest += ics->group_len[g] * 128;
+        src  += ics->group_len[g] * 128;
+    }
+}
+
+/**
+ * Apply independent channel coupling (applied after IMDCT).
+ *
+ * @param   index   index into coupling gain array
+ */
+static void apply_independent_coupling_fixed(AACContext *ac,
+                                       SingleChannelElement *target,
+                                       ChannelElement *cce, int index)
+{
+    int i, c, shift, round, tmp;
+    const int gain = cce->coup.gain[index][0];
+    const int *src = cce->ch[0].ret;
+    unsigned int *dest = target->ret;
+    const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
+
+    c = cce_scale_fixed[gain & 7];
+    shift = (gain-1024) >> 3;
+    if (shift < 0) {
+        shift = -shift;
+        round = 1 << (shift - 1);
+
+        for (i = 0; i < len; i++) {
+            tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
+            dest[i] += (tmp + round) >> shift;
+        }
+    }
+    else {
+      for (i = 0; i < len; i++) {
+          tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
+          dest[i] += tmp * (1U << shift);
+      }
+    }
+}
+
+#include "aacdec_template.c"
+
+AVCodec ff_aac_fixed_decoder = {
+    .name            = "aac_fixed",
+    .long_name       = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
+    .type            = AVMEDIA_TYPE_AUDIO,
+    .id              = AV_CODEC_ID_AAC,
+    .priv_data_size  = sizeof(AACContext),
+    .init            = aac_decode_init,
+    .close           = aac_decode_close,
+    .decode          = aac_decode_frame,
+    .sample_fmts     = (const enum AVSampleFormat[]) {
+        AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
+    },
+    .capabilities    = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
+    .channel_layouts = aac_channel_layout,
+    .flush = flush,
+};
@@ -38,49 +38,14 @@
 /* @name ltp_coef
 * Table of the LTP coefficients
 */
-static const float ltp_coef[8] = {
-    0.570829, 0.696616, 0.813004, 0.911304,
-    0.984900, 1.067894, 1.194601, 1.369533,
+static const INTFLOAT ltp_coef[8] = {
+    Q30(0.570829f), Q30(0.696616f), Q30(0.813004f), Q30(0.911304f),
+    Q30(0.984900f), Q30(1.067894f), Q30(1.194601f), Q30(1.369533f),
 };

-/* @name tns_tmp2_map
- * Tables of the tmp2[] arrays of LPC coefficients used for TNS.
- * The suffix _M_N[] indicate the values of coef_compress and coef_res
- * respectively.
- * @{
- */
-static const float tns_tmp2_map_1_3[4] = {
-     0.00000000, -0.43388373,  0.64278758,  0.34202015,
-};
+static const int8_t tags_per_config[16] = { 0, 1, 1, 2, 3, 3, 4, 5, 0, 0, 0, 4, 5, 0, 5, 0 };

-static const float tns_tmp2_map_0_3[8] = {
-     0.00000000, -0.43388373, -0.78183150, -0.97492790,
-     0.98480773,  0.86602539,  0.64278758,  0.34202015,
-};
-
-static const float tns_tmp2_map_1_4[8] = {
-     0.00000000, -0.20791170, -0.40673664, -0.58778524,
-     0.67369562,  0.52643216,  0.36124167,  0.18374951,
-};
-
-static const float tns_tmp2_map_0_4[16] = {
-     0.00000000, -0.20791170, -0.40673664, -0.58778524,
-    -0.74314481, -0.86602539, -0.95105654, -0.99452192,
-     0.99573416,  0.96182561,  0.89516330,  0.79801720,
-     0.67369562,  0.52643216,  0.36124167,  0.18374951,
-};
-
-static const float * const tns_tmp2_map[4] = {
-    tns_tmp2_map_0_3,
-    tns_tmp2_map_0_4,
-    tns_tmp2_map_1_3,
-    tns_tmp2_map_1_4
-};
-// @}
-
-static const int8_t tags_per_config[16] = { 0, 1, 1, 2, 3, 3, 4, 5, 0, 0, 0, 0, 0, 0, 0, 0 };
-
-static const uint8_t aac_channel_layout_map[7][5][3] = {
+static const uint8_t aac_channel_layout_map[16][5][3] = {
    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, },
    { { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, },
    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, },
@@ -88,9 +53,16 @@ static const uint8_t aac_channel_layout_map[7][5][3] = {
    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_BACK }, },
    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE  }, },
    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_FRONT }, { TYPE_CPE, 2, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE  }, },
+    { { 0, } },
+    { { 0, } },
+    { { 0, } },
+    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_BACK }, { TYPE_SCE, 1, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE  }, },
+    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_SIDE }, { TYPE_CPE, 2, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE  }, },
+    { { 0, } },
+    /* TODO: Add 7+1 TOP configuration */
 };

-static const uint64_t aac_channel_layout[8] = {
+static const uint64_t aac_channel_layout[16] = {
    AV_CH_LAYOUT_MONO,
    AV_CH_LAYOUT_STEREO,
    AV_CH_LAYOUT_SURROUND,
@@ -99,6 +71,12 @@ static const uint64_t aac_channel_layout[8] = {
    AV_CH_LAYOUT_5POINT1_BACK,
    AV_CH_LAYOUT_7POINT1_WIDE_BACK,
    0,
+    0,
+    0,
+    AV_CH_LAYOUT_6POINT1,
+    AV_CH_LAYOUT_7POINT1,
+    0,
+    /* AV_CH_LAYOUT_7POINT1_TOP, */
 };

 #endif /* AVCODEC_AACDECTAB_H */
@@ -27,7 +27,6 @@
 /***********************************
 *              TODOs:
 * add sane pulse detection
- * add temporal noise shaping
 ***********************************/

 #include "libavutil/float_dsp.h"
@@ -42,127 +41,11 @@
 #include "aac.h"
 #include "aactab.h"
 #include "aacenc.h"
+#include "aacenctab.h"
+#include "aacenc_utils.h"

 #include "psymodel.h"

-#define AAC_MAX_CHANNELS 6
-
-#define ERROR_IF(cond, ...) \
-    if (cond) { \
-        av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
-        return AVERROR(EINVAL); \
-    }
-
-#define WARN_IF(cond, ...) \
-    if (cond) { \
-        av_log(avctx, AV_LOG_WARNING, __VA_ARGS__); \
-    }
-
-float ff_aac_pow34sf_tab[428];
-
-static const uint8_t swb_size_1024_96[] = {
-    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
-    12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
-    64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
-};
-
-static const uint8_t swb_size_1024_64[] = {
-    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
-    12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
-    40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
-};
-
-static const uint8_t swb_size_1024_48[] = {
-    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
-    12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
-    32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
-    96
-};
-
-static const uint8_t swb_size_1024_32[] = {
-    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
-    12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
-    32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
-};
-
-static const uint8_t swb_size_1024_24[] = {
-    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
-    12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
-    32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
-};
-
-static const uint8_t swb_size_1024_16[] = {
-    8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
-    12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
-    32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
-};
-
-static const uint8_t swb_size_1024_8[] = {
-    12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
-    16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
-    32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
-};
-
-static const uint8_t *swb_size_1024[] = {
-    swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
-    swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
-    swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
-    swb_size_1024_16, swb_size_1024_16, swb_size_1024_8,
-    swb_size_1024_8
-};
-
-static const uint8_t swb_size_128_96[] = {
-    4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
-};
-
-static const uint8_t swb_size_128_48[] = {
-    4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
-};
-
-static const uint8_t swb_size_128_24[] = {
-    4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
-};
-
-static const uint8_t swb_size_128_16[] = {
-    4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
-};
-
-static const uint8_t swb_size_128_8[] = {
-    4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
-};
-
-static const uint8_t *swb_size_128[] = {
-    /* the last entry on the following row is swb_size_128_64 but is a
-       duplicate of swb_size_128_96 */
-    swb_size_128_96, swb_size_128_96, swb_size_128_96,
-    swb_size_128_48, swb_size_128_48, swb_size_128_48,
-    swb_size_128_24, swb_size_128_24, swb_size_128_16,
-    swb_size_128_16, swb_size_128_16, swb_size_128_8,
-    swb_size_128_8
-};
-
-/** default channel configurations */
-static const uint8_t aac_chan_configs[6][5] = {
- {1, TYPE_SCE},                               // 1 channel  - single channel element
- {1, TYPE_CPE},                               // 2 channels - channel pair
- {2, TYPE_SCE, TYPE_CPE},                     // 3 channels - center + stereo
- {3, TYPE_SCE, TYPE_CPE, TYPE_SCE},           // 4 channels - front center + stereo + back center
- {3, TYPE_SCE, TYPE_CPE, TYPE_CPE},           // 5 channels - front center + stereo + back stereo
- {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
-};
-
-/**
- * Table to remap channels from libavcodec's default order to AAC order.
- */
-static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
-    { 0 },
-    { 0, 1 },
-    { 2, 0, 1 },
-    { 2, 0, 1, 3 },
-    { 2, 0, 1, 3, 4 },
-    { 2, 0, 1, 4, 5, 3 },
-};
-
 /**
 * Make AAC audio config object.
 * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
@@ -173,7 +56,7 @@ static void put_audio_specific_config(AVCodecContext *avctx)
    AACEncContext *s = avctx->priv_data;

    init_put_bits(&pb, avctx->extradata, avctx->extradata_size);
-    put_bits(&pb, 5, 2); //object type - AAC-LC
+    put_bits(&pb, 5, s->profile+1); //profile
    put_bits(&pb, 4, s->samplerate_index); //sample rate index
    put_bits(&pb, 4, s->channels);
    //GASpecificConfig
@@ -265,7 +148,7 @@ static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
        s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
    else
        for (i = 0; i < 1024; i += 128)
-            s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2);
+            s->mdct128.mdct_calc(&s->mdct128, &sce->coeffs[i], output + i*2);
    memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
    memcpy(sce->pcoeffs, sce->coeffs, sizeof(sce->pcoeffs));
 }
@@ -283,7 +166,7 @@ static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
    put_bits(&s->pb, 1, info->use_kb_window[0]);
    if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
        put_bits(&s->pb, 6, info->max_sfb);
-        put_bits(&s->pb, 1, 0);            // no prediction
+        put_bits(&s->pb, 1, !!info->predictor_present);
    } else {
        put_bits(&s->pb, 4, info->max_sfb);
        for (w = 1; w < 8; w++)
@@ -312,26 +195,14 @@ static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
 static void adjust_frame_information(ChannelElement *cpe, int chans)
 {
    int i, w, w2, g, ch;
-    int start, maxsfb, cmaxsfb;
+    int maxsfb, cmaxsfb;

    for (ch = 0; ch < chans; ch++) {
        IndividualChannelStream *ics = &cpe->ch[ch].ics;
-        start = 0;
        maxsfb = 0;
        cpe->ch[ch].pulse.num_pulse = 0;
        for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
-            for (w2 = 0; w2 < ics->group_len[w]; w2++) {
-                start = (w+w2) * 128;
-                for (g = 0; g < ics->num_swb; g++) {
-                    //apply M/S
-                    if (cpe->common_window && !ch && cpe->ms_mask[w*16 + g]) {
-                        for (i = 0; i < ics->swb_sizes[g]; i++) {
-                            cpe->ch[0].coeffs[start+i] = (cpe->ch[0].pcoeffs[start+i] + cpe->ch[1].pcoeffs[start+i]) * 0.5f;
-                            cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].pcoeffs[start+i];
-                        }
-                    }
-                    start += ics->swb_sizes[g];
-                }
+            for (w2 =  0; w2 < ics->group_len[w]; w2++) {
                for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w*16+cmaxsfb-1]; cmaxsfb--)
                    ;
                maxsfb = FFMAX(maxsfb, cmaxsfb);
@@ -371,6 +242,59 @@ static void adjust_frame_information(ChannelElement *cpe, int chans)
    }
 }

+static void apply_intensity_stereo(ChannelElement *cpe)
+{
+    int w, w2, g, i;
+    IndividualChannelStream *ics = &cpe->ch[0].ics;
+    if (!cpe->common_window)
+        return;
+    for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
+        for (w2 =  0; w2 < ics->group_len[w]; w2++) {
+            int start = (w+w2) * 128;
+            for (g = 0; g < ics->num_swb; g++) {
+                int p  = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14);
+                float scale = cpe->ch[0].is_ener[w*16+g];
+                if (!cpe->is_mask[w*16 + g]) {
+                    start += ics->swb_sizes[g];
+                    continue;
+                }
+                for (i = 0; i < ics->swb_sizes[g]; i++) {
+                    float sum = (cpe->ch[0].coeffs[start+i] + p*cpe->ch[1].coeffs[start+i])*scale;
+                    cpe->ch[0].coeffs[start+i] = sum;
+                    cpe->ch[1].coeffs[start+i] = 0.0f;
+                }
+                start += ics->swb_sizes[g];
+            }
+        }
+    }
+}
+
+static void apply_mid_side_stereo(ChannelElement *cpe)
+{
+    int w, w2, g, i;
+    IndividualChannelStream *ics = &cpe->ch[0].ics;
+    if (!cpe->common_window)
+        return;
+    for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
+        for (w2 =  0; w2 < ics->group_len[w]; w2++) {
+            int start = (w+w2) * 128;
+            for (g = 0; g < ics->num_swb; g++) {
+                if (!cpe->ms_mask[w*16 + g]) {
+                    start += ics->swb_sizes[g];
+                    continue;
+                }
+                for (i = 0; i < ics->swb_sizes[g]; i++) {
+                    float L = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) * 0.5f;
+                    float R = L - cpe->ch[1].coeffs[start+i];
+                    cpe->ch[0].coeffs[start+i] = L;
+                    cpe->ch[1].coeffs[start+i] = R;
+                }
+                start += ics->swb_sizes[g];
+            }
+        }
+    }
+}
+
 /**
 * Encode scalefactor band coding type.
 */
@@ -378,6 +302,9 @@ static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
 {
    int w;

+    if (s->coder->set_special_band_scalefactors)
+        s->coder->set_special_band_scalefactors(s, sce);
+
    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
        s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
 }
@@ -388,15 +315,30 @@ static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
 static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
                                 SingleChannelElement *sce)
 {
-    int off = sce->sf_idx[0], diff;
+    int diff, off_sf = sce->sf_idx[0], off_pns = sce->sf_idx[0] - NOISE_OFFSET;
+    int off_is = 0, noise_flag = 1;
    int i, w;

    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
        for (i = 0; i < sce->ics.max_sfb; i++) {
            if (!sce->zeroes[w*16 + i]) {
-                diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
+                if (sce->band_type[w*16 + i] == NOISE_BT) {
+                    diff = sce->sf_idx[w*16 + i] - off_pns;
+                    off_pns = sce->sf_idx[w*16 + i];
+                    if (noise_flag-- > 0) {
+                        put_bits(&s->pb, NOISE_PRE_BITS, diff + NOISE_PRE);
+                        continue;
+                    }
+                } else if (sce->band_type[w*16 + i] == INTENSITY_BT  ||
+                           sce->band_type[w*16 + i] == INTENSITY_BT2) {
+                    diff = sce->sf_idx[w*16 + i] - off_is;
+                    off_is = sce->sf_idx[w*16 + i];
+                } else {
+                    diff = sce->sf_idx[w*16 + i] - off_sf;
+                    off_sf = sce->sf_idx[w*16 + i];
+                }
+                diff += SCALE_DIFF_ZERO;
                av_assert0(diff >= 0 && diff <= 120);
-                off = sce->sf_idx[w*16 + i];
                put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
            }
        }
@@ -436,17 +378,40 @@ static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
                start += sce->ics.swb_sizes[i];
                continue;
            }
-            for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++)
-                s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,
-                                                   sce->ics.swb_sizes[i],
+            for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++) {
+                s->coder->quantize_and_encode_band(s, &s->pb,
+                                                   &sce->coeffs[start + w2*128],
+                                                   NULL, sce->ics.swb_sizes[i],
                                                   sce->sf_idx[w*16 + i],
                                                   sce->band_type[w*16 + i],
-                                                   s->lambda);
+                                                   s->lambda,
+                                                   sce->ics.window_clipping[w]);
+            }
            start += sce->ics.swb_sizes[i];
        }
    }
 }

+/**
+ * Downscale spectral coefficients for near-clipping windows to avoid artifacts
+ */
+static void avoid_clipping(AACEncContext *s, SingleChannelElement *sce)
+{
+    int start, i, j, w;
+
+    if (sce->ics.clip_avoidance_factor < 1.0f) {
+        for (w = 0; w < sce->ics.num_windows; w++) {
+            start = 0;
+            for (i = 0; i < sce->ics.max_sfb; i++) {
+                float *swb_coeffs = &sce->coeffs[start + w*128];
+                for (j = 0; j < sce->ics.swb_sizes[i]; j++)
+                    swb_coeffs[j] *= sce->ics.clip_avoidance_factor;
+                start += sce->ics.swb_sizes[i];
+            }
+        }
+    }
+}
+
 /**
 * Encode one channel of audio data.
 */
@@ -455,12 +420,17 @@ static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
                                     int common_window)
 {
    put_bits(&s->pb, 8, sce->sf_idx[0]);
-    if (!common_window)
+    if (!common_window) {
        put_ics_info(s, &sce->ics);
+        if (s->coder->encode_main_pred)
+            s->coder->encode_main_pred(s, sce);
+    }
    encode_band_info(s, sce);
    encode_scale_factors(avctx, s, sce);
    encode_pulses(s, &sce->pulse);
-    put_bits(&s->pb, 1, 0); //tns
+    put_bits(&s->pb, 1, !!sce->tns.present);
+    if (s->coder->encode_tns_info)
+        s->coder->encode_tns_info(s, sce);
    put_bits(&s->pb, 1, 0); //ssr
    encode_spectral_coeffs(s, sce);
    return 0;
@@ -518,9 +488,12 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
    AACEncContext *s = avctx->priv_data;
    float **samples = s->planar_samples, *samples2, *la, *overlap;
    ChannelElement *cpe;
-    int i, ch, w, g, chans, tag, start_ch, ret, ms_mode = 0;
+    SingleChannelElement *sce;
+    int i, ch, w, chans, tag, start_ch, ret;
+    int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0;
    int chan_el_counter[4];
    FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
+    int k;

    if (s->last_frame == 2)
        return 0;
@@ -547,6 +520,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
        for (ch = 0; ch < chans; ch++) {
            IndividualChannelStream *ics = &cpe->ch[ch].ics;
            int cur_channel = start_ch + ch;
+            float clip_avoidance_factor;
            overlap  = &samples[cur_channel][0];
            samples2 = overlap + 1024;
            la       = samples2 + (448+64);
@@ -574,25 +548,50 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
            ics->num_windows        = wi[ch].num_windows;
            ics->swb_sizes          = s->psy.bands    [ics->num_windows == 8];
            ics->num_swb            = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
+            ics->max_sfb            = FFMIN(ics->max_sfb, ics->num_swb);
+            ics->swb_offset         = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
+                                        ff_swb_offset_128 [s->samplerate_index]:
+                                        ff_swb_offset_1024[s->samplerate_index];
+            ics->tns_max_bands      = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
+                                        ff_tns_max_bands_128 [s->samplerate_index]:
+                                        ff_tns_max_bands_1024[s->samplerate_index];
+            clip_avoidance_factor = 0.0f;
            for (w = 0; w < ics->num_windows; w++)
                ics->group_len[w] = wi[ch].grouping[w];
+            for (w = 0; w < ics->num_windows; w++) {
+                if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) {
+                    ics->window_clipping[w] = 1;
+                    clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]);
+                } else {
+                    ics->window_clipping[w] = 0;
+                }
+            }
+            if (clip_avoidance_factor > CLIP_AVOIDANCE_FACTOR) {
+                ics->clip_avoidance_factor = CLIP_AVOIDANCE_FACTOR / clip_avoidance_factor;
+            } else {
+                ics->clip_avoidance_factor = 1.0f;
+            }

            apply_window_and_mdct(s, &cpe->ch[ch], overlap);
-            if (isnan(cpe->ch->coeffs[0])) {
-                av_log(avctx, AV_LOG_ERROR, "Input contains NaN\n");
-                return AVERROR(EINVAL);
+
+            for (k = 0; k < 1024; k++) {
+                if (!(fabs(cpe->ch[ch].coeffs[k]) < 1E16)) { // Ensure headroom for energy calculation
+                    av_log(avctx, AV_LOG_ERROR, "Input contains (near) NaN/+-Inf\n");
+                    return AVERROR(EINVAL);
+                }
            }
+            avoid_clipping(s, &cpe->ch[ch]);
        }
        start_ch += chans;
    }
-    if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels)) < 0)
+    if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels, 0)) < 0)
        return ret;
    do {
        int frame_bits;

        init_put_bits(&s->pb, avpkt->data, avpkt->size);

-        if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
+        if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & AV_CODEC_FLAG_BITEXACT))
            put_bitstream_info(s, LIBAVCODEC_IDENT);
        start_ch = 0;
        memset(chan_el_counter, 0, sizeof(chan_el_counter));
@@ -602,16 +601,26 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
            tag      = s->chan_map[i+1];
            chans    = tag == TYPE_CPE ? 2 : 1;
            cpe      = &s->cpe[i];
+            cpe->common_window = 0;
+            memset(cpe->is_mask, 0, sizeof(cpe->is_mask));
+            memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask));
            put_bits(&s->pb, 3, tag);
            put_bits(&s->pb, 4, chan_el_counter[tag]++);
-            for (ch = 0; ch < chans; ch++)
-                coeffs[ch] = cpe->ch[ch].coeffs;
+            for (ch = 0; ch < chans; ch++) {
+                sce = &cpe->ch[ch];
+                coeffs[ch] = sce->coeffs;
+                sce->ics.predictor_present = 0;
+                memset(&sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
+                memset(&sce->tns, 0, sizeof(TemporalNoiseShaping));
+                for (w = 0; w < 128; w++)
+                    if (sce->band_type[w] > RESERVED_BT)
+                        sce->band_type[w] = 0;
+            }
            s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
            for (ch = 0; ch < chans; ch++) {
                s->cur_channel = start_ch + ch;
                s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
            }
-            cpe->common_window = 0;
            if (chans > 1
                && wi[0].window_type[0] == wi[1].window_type[0]
                && wi[0].window_shape   == wi[1].window_shape) {
@@ -624,22 +633,59 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                    }
                }
            }
+            for (ch = 0; ch < chans; ch++) { /* TNS and PNS */
+                sce = &cpe->ch[ch];
+                s->cur_channel = start_ch + ch;
+                if (s->options.pns && s->coder->search_for_pns)
+                    s->coder->search_for_pns(s, avctx, sce);
+                if (s->options.tns && s->coder->search_for_tns)
+                    s->coder->search_for_tns(s, sce);
+                if (s->options.tns && s->coder->apply_tns_filt)
+                    s->coder->apply_tns_filt(s, sce);
+                if (sce->tns.present)
+                    tns_mode = 1;
+            }
            s->cur_channel = start_ch;
-            if (s->options.stereo_mode && cpe->common_window) {
-                if (s->options.stereo_mode > 0) {
-                    IndividualChannelStream *ics = &cpe->ch[0].ics;
-                    for (w = 0; w < ics->num_windows; w += ics->group_len[w])
-                        for (g = 0;  g < ics->num_swb; g++)
-                            cpe->ms_mask[w*16+g] = 1;
-                } else if (s->coder->search_for_ms) {
-                    s->coder->search_for_ms(s, cpe, s->lambda);
+            if (s->options.intensity_stereo) { /* Intensity Stereo */
+                if (s->coder->search_for_is)
+                    s->coder->search_for_is(s, avctx, cpe);
+                if (cpe->is_mode) is_mode = 1;
+                apply_intensity_stereo(cpe);
+            }
+            if (s->options.pred) { /* Prediction */
+                for (ch = 0; ch < chans; ch++) {
+                    sce = &cpe->ch[ch];
+                    s->cur_channel = start_ch + ch;
+                    if (s->options.pred && s->coder->search_for_pred)
+                        s->coder->search_for_pred(s, sce);
+                    if (cpe->ch[ch].ics.predictor_present) pred_mode = 1;
                }
+                if (s->coder->adjust_common_prediction)
+                    s->coder->adjust_common_prediction(s, cpe);
+                for (ch = 0; ch < chans; ch++) {
+                    sce = &cpe->ch[ch];
+                    s->cur_channel = start_ch + ch;
+                    if (s->options.pred && s->coder->apply_main_pred)
+                        s->coder->apply_main_pred(s, sce);
+                }
+                s->cur_channel = start_ch;
+            }
+            if (s->options.stereo_mode) { /* Mid/Side stereo */
+                if (s->options.stereo_mode == -1 && s->coder->search_for_ms)
+                    s->coder->search_for_ms(s, cpe);
+                else if (cpe->common_window)
+                    memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask));
+                for (w = 0; w < 128; w++)
+                    cpe->ms_mask[w] = cpe->is_mask[w] ? 0 : cpe->ms_mask[w];
+                apply_mid_side_stereo(cpe);
            }
            adjust_frame_information(cpe, chans);
            if (chans == 2) {
                put_bits(&s->pb, 1, cpe->common_window);
                if (cpe->common_window) {
                    put_ics_info(s, &cpe->ch[0].ics);
+                    if (s->coder->encode_main_pred)
+                        s->coder->encode_main_pred(s, &cpe->ch[0]);
                    encode_ms_info(&s->pb, cpe);
                    if (cpe->ms_mode) ms_mode = 1;
                }
@@ -656,7 +702,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
            s->psy.bitres.bits = frame_bits / s->channels;
            break;
        }
-        if (ms_mode) {
+        if (is_mode || ms_mode || tns_mode || pred_mode) {
            for (i = 0; i < s->chan_map[0]; i++) {
                // Must restore coeffs
                chans = tag == TYPE_CPE ? 2 : 1;
@@ -675,7 +721,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
    avctx->frame_bits = put_bits_count(&s->pb);

    // rate control stuff
-    if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
+    if (!(avctx->flags & AV_CODEC_FLAG_QSCALE)) {
        float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
        s->lambda *= ratio;
        s->lambda = FFMIN(s->lambda, 65536.f);
@@ -699,6 +745,7 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)
    ff_mdct_end(&s->mdct1024);
    ff_mdct_end(&s->mdct128);
    ff_psy_end(&s->psy);
+    ff_lpc_end(&s->lpc);
    if (s->psypp)
        ff_psy_preprocess_end(s->psypp);
    av_freep(&s->buffer.samples);
@@ -712,7 +759,7 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
 {
    int ret = 0;

-    s->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);
+    s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
    if (!s->fdsp)
        return AVERROR(ENOMEM);

@@ -722,9 +769,9 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
    ff_init_ff_sine_windows(10);
    ff_init_ff_sine_windows(7);

-    if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0))
+    if ((ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0)) < 0)
        return ret;
-    if (ret = ff_mdct_init(&s->mdct128,   8, 0, 32768.0))
+    if ((ret = ff_mdct_init(&s->mdct128,   8, 0, 32768.0)) < 0)
        return ret;

    return 0;
@@ -735,7 +782,7 @@ static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
    int ch;
    FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->buffer.samples, s->channels, 3 * 1024 * sizeof(s->buffer.samples[0]), alloc_fail);
    FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->cpe, s->chan_map[0], sizeof(ChannelElement), alloc_fail);
-    FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
+    FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + AV_INPUT_BUFFER_PADDING_SIZE, alloc_fail);

    for(ch = 0; ch < s->channels; ch++)
        s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
@@ -761,16 +808,29 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)

    s->channels = avctx->channels;

-    ERROR_IF(i == 16
-                || i >= (sizeof(swb_size_1024) / sizeof(*swb_size_1024))
-                || i >= (sizeof(swb_size_128) / sizeof(*swb_size_128)),
+    ERROR_IF(i == 16 || i >= ff_aac_swb_size_1024_len || i >= ff_aac_swb_size_128_len,
             "Unsupported sample rate %d\n", avctx->sample_rate);
    ERROR_IF(s->channels > AAC_MAX_CHANNELS,
             "Unsupported number of channels: %d\n", s->channels);
-    ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW,
-             "Unsupported profile %d\n", avctx->profile);
    WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
             "Too many bits per frame requested, clamping to max\n");
+    if (avctx->profile == FF_PROFILE_AAC_MAIN) {
+        s->options.pred = 1;
+    } else if ((avctx->profile == FF_PROFILE_AAC_LOW ||
+                avctx->profile == FF_PROFILE_UNKNOWN) && s->options.pred) {
+        s->profile = 0; /* Main */
+        WARN_IF(1, "Prediction requested, changing profile to AAC-Main\n");
+    } else if (avctx->profile == FF_PROFILE_AAC_LOW ||
+               avctx->profile == FF_PROFILE_UNKNOWN) {
+        s->profile = 1; /* Low */
+    } else {
+        ERROR_IF(1, "Unsupported profile %d\n", avctx->profile);
+    }
+
+    if (s->options.aac_coder != AAC_CODER_TWOLOOP) {
+        s->options.intensity_stereo = 0;
+        s->options.pns = 0;
+    }

    avctx->bit_rate = (int)FFMIN(
        6144 * s->channels / 1024.0 * avctx->sample_rate,
@@ -789,8 +849,8 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
    avctx->extradata_size = 5;
    put_audio_specific_config(avctx);

-    sizes[0]   = swb_size_1024[i];
-    sizes[1]   = swb_size_128[i];
+    sizes[0]   = ff_aac_swb_size_1024[i];
+    sizes[1]   = ff_aac_swb_size_128[i];
    lengths[0] = ff_aac_num_swb_1024[i];
    lengths[1] = ff_aac_num_swb_128[i];
    for (i = 0; i < s->chan_map[0]; i++)
@@ -800,6 +860,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
        goto fail;
    s->psypp = ff_psy_preprocess_init(avctx);
    s->coder = &ff_aac_coders[s->options.aac_coder];
+    ff_lpc_init(&s->lpc, 2*avctx->frame_size, TNS_MAX_ORDER, FF_LPC_TYPE_LEVINSON);

    if (HAVE_MIPSDSPR1)
        ff_aac_coder_init_mips(s);
@@ -808,9 +869,6 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)

    ff_aac_tableinit();

-    for (i = 0; i < 428; i++)
-        ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));
-
    avctx->initial_padding = 1024;
    ff_af_queue_init(avctx, &s->afq);

@@ -826,11 +884,23 @@ static const AVOption aacenc_options[] = {
        {"auto",     "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
        {"ms_off",   "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 =  0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
        {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 =  1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
-    {"aac_coder", "", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "aac_coder"},
+    {"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "aac_coder"},
        {"faac",     "FAAC-inspired method",      0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAAC},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
        {"anmr",     "ANMR method",               0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
        {"twoloop",  "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
        {"fast",     "Constant quantizer",        0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
+    {"aac_pns", "Perceptual Noise Substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_INT, {.i64 = 1}, 0, 1, AACENC_FLAGS, "aac_pns"},
+        {"disable",  "Disable perceptual noise substitution", 0, AV_OPT_TYPE_CONST, {.i64 =  0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
+        {"enable",   "Enable perceptual noise substitution",  0, AV_OPT_TYPE_CONST, {.i64 =  1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
+    {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_INT, {.i64 = 1}, 0, 1, AACENC_FLAGS, "intensity_stereo"},
+        {"disable",  "Disable intensity stereo coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "intensity_stereo"},
+        {"enable",   "Enable intensity stereo coding", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "intensity_stereo"},
+    {"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "aac_tns"},
+        {"disable",  "Disable temporal noise shaping", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_tns"},
+        {"enable",   "Enable temporal noise shaping", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_tns"},
+    {"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "aac_pred"},
+        {"disable",  "Disable AAC-Main prediction", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pred"},
+        {"enable",   "Enable AAC-Main prediction", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pred"},
    {NULL}
 };

@@ -841,13 +911,6 @@ static const AVClass aacenc_class = {
    LIBAVUTIL_VERSION_INT,
 };

-/* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build
- * failures */
-static const int mpeg4audio_sample_rates[16] = {
-    96000, 88200, 64000, 48000, 44100, 32000,
-    24000, 22050, 16000, 12000, 11025, 8000, 7350
-};
-
 AVCodec ff_aac_encoder = {
    .name           = "aac",
    .long_name      = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
@@ -858,8 +921,8 @@ AVCodec ff_aac_encoder = {
    .encode2        = aac_encode_frame,
    .close          = aac_encode_end,
    .supported_samplerates = mpeg4audio_sample_rates,
-    .capabilities   = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |
-                      CODEC_CAP_EXPERIMENTAL,
+    .capabilities   = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY |
+                      AV_CODEC_CAP_EXPERIMENTAL,
    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
                                                     AV_SAMPLE_FMT_NONE },
    .priv_class     = &aacenc_class,
@@ -30,6 +30,8 @@
 #include "audio_frame_queue.h"
 #include "psymodel.h"

+#include "lpc.h"
+
 typedef enum AACCoder {
    AAC_CODER_FAAC = 0,
    AAC_CODER_ANMR,
@@ -42,6 +44,10 @@ typedef enum AACCoder {
 typedef struct AACEncOptions {
    int stereo_mode;
    int aac_coder;
+    int pns;
+    int tns;
+    int pred;
+    int intensity_stereo;
 } AACEncOptions;

 struct AACEncContext;
@@ -51,9 +57,19 @@ typedef struct AACCoefficientsEncoder {
                                  SingleChannelElement *sce, const float lambda);
    void (*encode_window_bands_info)(struct AACEncContext *s, SingleChannelElement *sce,
                                     int win, int group_len, const float lambda);
-    void (*quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, int size,
-                                     int scale_idx, int cb, const float lambda);
-    void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe, const float lambda);
+    void (*quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, float *out, int size,
+                                     int scale_idx, int cb, const float lambda, int rtz);
+    void (*encode_tns_info)(struct AACEncContext *s, SingleChannelElement *sce);
+    void (*encode_main_pred)(struct AACEncContext *s, SingleChannelElement *sce);
+    void (*adjust_common_prediction)(struct AACEncContext *s, ChannelElement *cpe);
+    void (*apply_main_pred)(struct AACEncContext *s, SingleChannelElement *sce);
+    void (*apply_tns_filt)(struct AACEncContext *s, SingleChannelElement *sce);
+    void (*set_special_band_scalefactors)(struct AACEncContext *s, SingleChannelElement *sce);
+    void (*search_for_pns)(struct AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce);
+    void (*search_for_tns)(struct AACEncContext *s, SingleChannelElement *sce);
+    void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe);
+    void (*search_for_is)(struct AACEncContext *s, AVCodecContext *avctx, ChannelElement *cpe);
+    void (*search_for_pred)(struct AACEncContext *s, SingleChannelElement *sce);
 } AACCoefficientsEncoder;

 extern AACCoefficientsEncoder ff_aac_coders[];
@@ -70,6 +86,8 @@ typedef struct AACEncContext {
    AVFloatDSPContext *fdsp;
    float *planar_samples[6];                    ///< saved preprocessed input

+    int profile;                                 ///< copied from avctx
+    LPCContext lpc;                              ///< used by TNS
    int samplerate_index;                        ///< MPEG-4 samplerate index
    int channels;                                ///< channel count
    const uint8_t *chan_map;                     ///< channel configuration map
@@ -90,8 +108,6 @@ typedef struct AACEncContext {
    } buffer;
 } AACEncContext;

-extern float ff_aac_pow34sf_tab[428];
-
 void ff_aac_coder_init_mips(AACEncContext *c);

 #endif /* AVCODEC_AACENC_H */
@@ -0,0 +1,136 @@
+/*
+ * AAC encoder intensity stereo
+ * Copyright (C) 2015 Rostislav Pehlivanov
+ *
+ * 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
+ * AAC encoder Intensity Stereo
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#include "aacenc.h"
+#include "aacenc_utils.h"
+#include "aacenc_is.h"
+#include "aacenc_quantization.h"
+
+struct AACISError ff_aac_is_encoding_err(AACEncContext *s, ChannelElement *cpe,
+                                         int start, int w, int g, float ener0,
+                                         float ener1, float ener01,
+                                         int use_pcoeffs, int phase)
+{
+    int i, w2;
+    SingleChannelElement *sce0 = &cpe->ch[0];
+    SingleChannelElement *sce1 = &cpe->ch[1];
+    float *L = use_pcoeffs ? sce0->pcoeffs : sce0->coeffs;
+    float *R = use_pcoeffs ? sce1->pcoeffs : sce1->coeffs;
+    float *L34 = &s->scoefs[256*0], *R34 = &s->scoefs[256*1];
+    float *IS  = &s->scoefs[256*2], *I34 = &s->scoefs[256*3];
+    float dist1 = 0.0f, dist2 = 0.0f;
+    struct AACISError is_error = {0};
+
+    for (w2 = 0; w2 < sce0->ics.group_len[w]; w2++) {
+        FFPsyBand *band0 = &s->psy.ch[s->cur_channel+0].psy_bands[(w+w2)*16+g];
+        FFPsyBand *band1 = &s->psy.ch[s->cur_channel+1].psy_bands[(w+w2)*16+g];
+        int is_band_type, is_sf_idx = FFMAX(1, sce0->sf_idx[(w+w2)*16+g]-4);
+        float e01_34 = phase*pow(ener1/ener0, 3.0/4.0);
+        float maxval, dist_spec_err = 0.0f;
+        float minthr = FFMIN(band0->threshold, band1->threshold);
+        for (i = 0; i < sce0->ics.swb_sizes[g]; i++)
+            IS[i] = (L[start+(w+w2)*128+i] + phase*R[start+(w+w2)*128+i])*sqrt(ener0/ener01);
+        abs_pow34_v(L34, &L[start+(w+w2)*128], sce0->ics.swb_sizes[g]);
+        abs_pow34_v(R34, &R[start+(w+w2)*128], sce0->ics.swb_sizes[g]);
+        abs_pow34_v(I34, IS,                   sce0->ics.swb_sizes[g]);
+        maxval = find_max_val(1, sce0->ics.swb_sizes[g], I34);
+        is_band_type = find_min_book(maxval, is_sf_idx);
+        dist1 += quantize_band_cost(s, &L[start + (w+w2)*128], L34,
+                                    sce0->ics.swb_sizes[g],
+                                    sce0->sf_idx[(w+w2)*16+g],
+                                    sce0->band_type[(w+w2)*16+g],
+                                    s->lambda / band0->threshold, INFINITY, NULL, 0);
+        dist1 += quantize_band_cost(s, &R[start + (w+w2)*128], R34,
+                                    sce1->ics.swb_sizes[g],
+                                    sce1->sf_idx[(w+w2)*16+g],
+                                    sce1->band_type[(w+w2)*16+g],
+                                    s->lambda / band1->threshold, INFINITY, NULL, 0);
+        dist2 += quantize_band_cost(s, IS, I34, sce0->ics.swb_sizes[g],
+                                    is_sf_idx, is_band_type,
+                                    s->lambda / minthr, INFINITY, NULL, 0);
+        for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
+            dist_spec_err += (L34[i] - I34[i])*(L34[i] - I34[i]);
+            dist_spec_err += (R34[i] - I34[i]*e01_34)*(R34[i] - I34[i]*e01_34);
+        }
+        dist_spec_err *= s->lambda / minthr;
+        dist2 += dist_spec_err;
+    }
+
+    is_error.pass = dist2 <= dist1;
+    is_error.phase = phase;
+    is_error.error = fabsf(dist1 - dist2);
+    is_error.dist1 = dist1;
+    is_error.dist2 = dist2;
+
+    return is_error;
+}
+
+void ff_aac_search_for_is(AACEncContext *s, AVCodecContext *avctx, ChannelElement *cpe)
+{
+    SingleChannelElement *sce0 = &cpe->ch[0];
+    SingleChannelElement *sce1 = &cpe->ch[1];
+    int start = 0, count = 0, w, w2, g, i;
+    const float freq_mult = avctx->sample_rate/(1024.0f/sce0->ics.num_windows)/2.0f;
+
+    if (!cpe->common_window)
+        return;
+
+    for (w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]) {
+        start = 0;
+        for (g = 0;  g < sce0->ics.num_swb; g++) {
+            if (start*freq_mult > INT_STEREO_LOW_LIMIT*(s->lambda/170.0f) &&
+                cpe->ch[0].band_type[w*16+g] != NOISE_BT && !cpe->ch[0].zeroes[w*16+g] &&
+                cpe->ch[1].band_type[w*16+g] != NOISE_BT && !cpe->ch[1].zeroes[w*16+g]) {
+                float ener0 = 0.0f, ener1 = 0.0f, ener01 = 0.0f;
+                struct AACISError ph_err1, ph_err2, *erf;
+                for (w2 = 0; w2 < sce0->ics.group_len[w]; w2++) {
+                    for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
+                        float coef0 = sce0->pcoeffs[start+(w+w2)*128+i];
+                        float coef1 = sce1->pcoeffs[start+(w+w2)*128+i];
+                        ener0  += coef0*coef0;
+                        ener1  += coef1*coef1;
+                        ener01 += (coef0 + coef1)*(coef0 + coef1);
+                    }
+                }
+                ph_err1 = ff_aac_is_encoding_err(s, cpe, start, w, g,
+                                                 ener0, ener1, ener01, 0, -1);
+                ph_err2 = ff_aac_is_encoding_err(s, cpe, start, w, g,
+                                                 ener0, ener1, ener01, 0, +1);
+                erf = ph_err1.error < ph_err2.error ? &ph_err1 : &ph_err2;
+                if (erf->pass) {
+                    cpe->is_mask[w*16+g] = 1;
+                    cpe->ch[0].is_ener[w*16+g] = sqrt(ener0/ener01);
+                    cpe->ch[1].is_ener[w*16+g] = ener0/ener1;
+                    cpe->ch[1].band_type[w*16+g] = erf->phase ? INTENSITY_BT : INTENSITY_BT2;
+                    count++;
+                }
+            }
+            start += sce0->ics.swb_sizes[g];
+        }
+    }
+    cpe->is_mode = !!count;
+}
--- a/Show More
+++ b/Show More
@@ -1 +1 @@
 .5.git
 .8.15