avformat/rpl: Fix check for negative values

Fixes: signed integer overflow: 10 * -1923267925333400000 cannot be represented in type 'int64_t' (aka 'long')
Fixes: 378891963/clusterfuzz-testcase-minimized-fuzzer_loadfile_direct-5714338935013376
Found-by: ossfuzz
Reported-by: Kacper Michajlow <kasper93@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit eab65379bf)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

.gitignore

@@ -28,6 +28,7 @@
 /ffserver
 /config.*
 /coverage.info
+/avversion.h
 /doc/*.1
 /doc/*.3
 /doc/*.html
@@ -36,7 +37,7 @@
 /doc/avoptions_codec.texi
 /doc/avoptions_format.texi
 /doc/doxy/html/
-/doc/examples/avio_list_dir
+/doc/examples/avio_dir_cmd
 /doc/examples/avio_reading
 /doc/examples/decoding_encoding
 /doc/examples/demuxing_decoding
@@ -62,6 +63,7 @@
 /libavutil/ffversion.h
 /tests/audiogen
 /tests/base64
+/tests/checkasm/checkasm
 /tests/data/
 /tests/pixfmts.mak
 /tests/rotozoom

CREDITS

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

INSTALL.md

@@ -15,3 +15,11 @@ NOTICE
 ------
 
  - Non system dependencies (e.g. libx264, libvpx) are disabled by default.
+
+NOTICE for Package Maintainers
+------------------------------
+
+ - It is recommended to build FFmpeg twice, first with minimal external dependencies so
+   that 3rd party packages, which depend on FFmpegs libavutil/libavfilter/libavcodec/libavformat
+   can then be built. And last build FFmpeg with full dependancies (which may in turn depend on
+   some of these 3rd party packages). This avoids circular dependencies during build.

LICENSE.md

@@ -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`

MAINTAINERS

@@ -14,7 +14,6 @@ patches and related discussions.
 Project Leader
 ==============
 
-Michael Niedermayer
   final design decisions
 
 
@@ -43,7 +42,7 @@ 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, Lou Logan
 presets                                 Robert Swain
@@ -57,7 +56,7 @@ 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, 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
 
@@ -576,9 +596,11 @@ Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
 Lou Logan                     7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
 Luca Barbato                  6677 4209 213C 8843 5B67 29E7 E84C 78C2 84E9 0E34
 Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
+                              DD1E C9E8 DE08 5C62 9B3E 1846 B18E 8928 B394 8D64
 Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
 Panagiotis Issaris            6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
 Peter Ross                    A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
+Philip Langdale               5DC5 8D66 5FBA 3A43 18EC 045E F8D6 B194 6A75 682E
 Reimar Doeffinger             C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
 Reinhard Tartler              9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
 Reynaldo H. Verdejo Pinochet  6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A

Makefile

@@ -31,7 +31,10 @@ $(foreach prog,$(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)
 
@@ -81,7 +85,7 @@ SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS      \
                ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS     \
                ALTIVEC-OBJS MMX-OBJS YASM-OBJS                           \
                MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MSA-OBJS       \
-               LOONGSON3-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS
+               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)

RELEASE

@@ -1 +1 @@
-2.6.git
+2.8.22

RELEASE_NOTES

@@ -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.libera.chat) or ask
+   on the mailing-lists.

arch.mak

@@ -8,7 +8,7 @@ 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_LOONGSON3) += $(LOONGSON3-OBJS)  $(LOONGSON3-OBJS-yes)
+OBJS-$(HAVE_MMI)   += $(MMI-OBJS)   $(MMI-OBJS-yes)
 
 OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
 OBJS-$(HAVE_VSX)     += $(VSX-OBJS) $(VSX-OBJS-yes)

cmdutils.c

@@ -60,10 +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;
 
@@ -73,20 +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);
@@ -113,6 +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))
@@ -529,7 +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;
@@ -565,14 +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
@@ -646,9 +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;
@@ -657,9 +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();
 
@@ -715,9 +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);
@@ -1324,12 +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");
@@ -1503,11 +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))
@@ -1581,10 +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)) ||
@@ -1875,64 +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)
@@ -2106,7 +2058,10 @@ double get_rotation(AVStream *st)
     theta -= 360*floor(theta/360 + 0.9/360);
 
     if (fabs(theta - 90*round(theta/90)) > 2)
-        av_log_ask_for_sample(NULL, "Odd rotation angle\n");
+        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;
 }
@@ -2120,7 +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");
@@ -2150,7 +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");

cmdutils.h

@@ -46,7 +46,7 @@ extern const int program_birth_year;
 
 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
-extern struct SwsContext *sws_opts;
+extern AVDictionary *sws_dict;
 extern AVDictionary *swr_opts;
 extern AVDictionary *format_opts, *codec_opts, *resample_opts;
 extern int hide_banner;
@@ -61,6 +61,11 @@ void register_exit(void (*cb)(int ret));
  */
 void exit_program(int ret) av_noreturn;
 
+/**
+ * Initialize dynamic library loading
+ */
+void init_dynload(void);
+
 /**
  * Initialize the cmdutils option system, in particular
  * allocate the *_opts contexts.
@@ -277,7 +282,7 @@ typedef struct OptionGroup {
     AVDictionary *codec_opts;
     AVDictionary *format_opts;
     AVDictionary *resample_opts;
-    struct SwsContext *sws_opts;
+    AVDictionary *sws_dict;
     AVDictionary *swr_opts;
 } OptionGroup;
 
@@ -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.
  *

common.mak

@@ -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))

compat/windows/mslink

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

doc/APIchanges

@@ -15,6 +15,42 @@ 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
@@ -196,7 +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.
 
@@ -213,7 +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
@@ -692,6 +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
 

doc/Doxyfile

@@ -31,7 +31,7 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.
 
-PROJECT_NUMBER         =
+PROJECT_NUMBER         = 2.8.22
 
 # With the PROJECT_LOGO tag one can specify a logo or icon that is included
 # in the documentation. The maximum height of the logo should not exceed 55

doc/Makefile

@@ -36,7 +36,7 @@ DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
 DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
 DOCS = $(DOCS-yes)
 
-DOC_EXAMPLES-$(CONFIG_AVIO_LIST_DIR_EXAMPLE)     += avio_list_dir
+DOC_EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE)      += avio_dir_cmd
 DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
 DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE)           += avcodec
 DOC_EXAMPLES-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding

doc/authors.texi

@@ -3,9 +3,9 @@
 The FFmpeg developers.
 
 For details about the authorship, see the Git history of the project
-(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
+(https://git.ffmpeg.org/ffmpeg), e.g. by typing the command
 @command{git log} in the FFmpeg source directory, or browsing the
-online repository at @url{http://source.ffmpeg.org}.
+online repository at @url{https://git.ffmpeg.org/ffmpeg}.
 
 Maintainers for the specific components are listed in the file
 @file{MAINTAINERS} in the source code tree.

doc/codecs.texi

@@ -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}

doc/decoders.texi

@@ -25,6 +25,13 @@ enabled decoders.
 A description of some of the currently available video decoders
 follows.
 
+@section hevc
+
+HEVC / H.265 decoder.
+
+Note: the @option{skip_loop_filter} option has effect only at level
+@code{all}.
+
 @section rawvideo
 
 Raw video decoder.
@@ -95,7 +102,7 @@ This decoder aims to implement the complete FLAC specification from Xiph.
 
 @item -use_buggy_lpc
 The lavc FLAC encoder used to produce buggy streams with high lpc values
-(like the default value). This option allows to decode such streams
+(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
@@ -188,6 +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

doc/demuxers.texi

@@ -18,6 +18,12 @@ enabled demuxers.
 
 The description of some of the currently available demuxers follows.
 
+@section aa
+
+Audible Format 2, 3, and 4 demuxer.
+
+This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
+
 @section applehttp
 
 Apple HTTP Live Streaming demuxer.
@@ -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
@@ -145,7 +192,9 @@ 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
@@ -234,6 +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.
@@ -369,17 +436,47 @@ ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
 @end example
 @end itemize
 
+@section mov/mp4/3gp/Quicktme
+
+Quicktime / MP4 demuxer.
+
+This demuxer accepts the following options:
+@table @option
+@item enable_drefs
+Enable loading of external tracks, disabled by default.
+Enabling this can theoretically leak information in some use cases.
+
+@item use_absolute_path
+Allows loading of external tracks via absolute paths, disabled by default.
+Enabling this poses a security risk. It should only be enabled if the source
+is known to be non malicious.
+
+@end table
+
 @section mpegts
 
 MPEG-2 transport stream demuxer.
 
+This demuxer accepts the following options:
 @table @option
+@item resync_size
+Set size limit for looking up a new synchronization. Default value is
+65536.
 
 @item fix_teletext_pts
-Overrides teletext packet PTS and DTS values with the timestamps calculated
+Override teletext packet PTS and DTS values with the timestamps calculated
 from the PCR of the first program which the teletext stream is part of and is
 not discarded. Default value is 1, set this option to 0 if you want your
 teletext packet PTS and DTS values untouched.
+
+@item ts_packetsize
+Output option carrying the raw packet size in bytes.
+Show the detected raw packet size, cannot be set by the user.
+
+@item scan_all_pmts
+Scan and combine all PMTs. The value is an integer with value from -1
+to 1 (-1 means automatic setting, 1 means enabled, 0 means
+disabled). Default value is -1.
 @end table
 
 @section rawvideo

doc/developer.texi

@@ -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
 
@@ -697,6 +734,25 @@ In case you need finer control over how valgrind is invoked, use the
 @code{--target-exec='valgrind <your_custom_valgrind_options>} option in
 your configure line instead.
 
+@anchor{Maintenance}
+@chapter Maintenance process
+
+@anchor{MAINTAINERS}
+@section MAINTAINERS
+
+The developers maintaining each part of the codebase are listed in @file{MAINTAINERS}.
+Being listed in @file{MAINTAINERS}, gives one the right to have git write access to
+the specific repository.
+
+@anchor{Becoming a maintainer}
+@section Becoming a maintainer
+
+People add themselves to @file{MAINTAINERS} by sending a patch like any other code
+change. These get reviewed by the community like any other patch. It is expected
+that, if someone has an objection to a new maintainer, she is willing to object
+in public with her full name and is willing to take over maintainership for the area.
+
+
 @anchor{Release process}
 @section Release process
 

doc/encoders.texi

@@ -1342,6 +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.
@@ -1416,113 +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
 
@@ -2260,6 +2324,30 @@ Setting a higher @option{bits_per_mb} limit will improve the speed.
 For the fastest encoding speed set the @option{qscale} parameter (4 is the
 recommended value) and do not set a size constraint.
 
+@section libkvazaar
+
+Kvazaar H.265/HEVC encoder.
+
+Requires the presence of the libkvazaar headers and library during
+configuration. You need to explicitly configure the build with
+@option{--enable-libkvazaar}.
+
+@subsection Options
+
+@table @option
+
+@item b
+Set target video bitrate in bit/s and enable rate control.
+
+@item threads
+Set number of encoding threads.
+
+@item kvazaar-params
+Set kvazaar parameters as a list of @var{name}=@var{value} pairs separated
+by commas (,). See kvazaar documentation for a list of options.
+
+@end table
+
 @c man end VIDEO ENCODERS
 
 @chapter Subtitles Encoders

doc/examples/Makefile

@@ -11,13 +11,14 @@ CFLAGS += -Wall -g
 CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
 LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
 
-EXAMPLES=       avio_list_dir                      \
+EXAMPLES=       avio_dir_cmd                       \
                 avio_reading                       \
                 decoding_encoding                  \
                 demuxing_decoding                  \
                 extract_mvs                        \
                 filtering_video                    \
                 filtering_audio                    \
+                http_multiclient                   \
                 metadata                           \
                 muxing                             \
                 remuxing                           \

doc/examples/avio_dir_cmd.c

@@ -54,28 +54,13 @@ static const char *type_string(int type)
     return "<UNKNOWN>";
 }
 
-int main(int argc, char *argv[])
+static int list_op(const char *input_dir)
 {
-    const char *input_dir = NULL;
     AVIODirEntry *entry = NULL;
     AVIODirContext *ctx = NULL;
     int cnt, ret;
     char filemode[4], uid_and_gid[20];
 
-    av_log_set_level(AV_LOG_DEBUG);
-
-    if (argc != 2) {
-        fprintf(stderr, "usage: %s input_dir\n"
-                "API example program to show how to list files in directory "
-                "accessed through AVIOContext.\n", argv[0]);
-        return 1;
-    }
-    input_dir = argv[1];
-
-    /* register codecs and formats and other lavf/lavc components*/
-    av_register_all();
-    avformat_network_init();
-
     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;
@@ -114,6 +99,81 @@ int main(int argc, char *argv[])
 
   fail:
     avio_close_dir(&ctx);
+    return ret;
+}
+
+static int del_op(const char *url)
+{
+    int ret = avpriv_io_delete(url);
+    if (ret < 0)
+        av_log(NULL, AV_LOG_ERROR, "Cannot delete '%s': %s.\n", url, av_err2str(ret));
+    return ret;
+}
+
+static int move_op(const char *src, const char *dst)
+{
+    int ret = avpriv_io_move(src, dst);
+    if (ret < 0)
+        av_log(NULL, AV_LOG_ERROR, "Cannot move '%s' into '%s': %s.\n", src, dst, av_err2str(ret));
+    return ret;
+}
+
+
+static void usage(const char *program_name)
+{
+    fprintf(stderr, "usage: %s OPERATION entry1 [entry2]\n"
+            "API example program to show how to manipulate resources "
+            "accessed through AVIOContext.\n"
+            "OPERATIONS:\n"
+            "list      list content of the directory\n"
+            "move      rename content in directory\n"
+            "del       delete content in directory\n",
+            program_name);
+}
+
+int main(int argc, char *argv[])
+{
+    const char *op = NULL;
+    int ret;
+
+    av_log_set_level(AV_LOG_DEBUG);
+
+    if (argc < 2) {
+        usage(argv[0]);
+        return 1;
+    }
+
+    /* register codecs and formats and other lavf/lavc components*/
+    av_register_all();
+    avformat_network_init();
+
+    op = argv[1];
+    if (strcmp(op, "list") == 0) {
+        if (argc < 3) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for list operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = list_op(argv[2]);
+        }
+    } else if (strcmp(op, "del") == 0) {
+        if (argc < 3) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for del operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = del_op(argv[2]);
+        }
+    } else if (strcmp(op, "move") == 0) {
+        if (argc < 4) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for move operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = move_op(argv[2], argv[3]);
+        }
+    } else {
+        av_log(NULL, AV_LOG_INFO, "Invalid operation %s\n", op);
+        ret = AVERROR(EINVAL);
+    }
+
     avformat_network_deinit();
 
     return ret < 0 ? 1 : 0;

doc/examples/decoding_encoding.c

@@ -245,7 +245,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
     AVCodecContext *c= NULL;
     int len;
     FILE *f, *outfile;
-    uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
+    uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
     AVPacket avpkt;
     AVFrame *decoded_frame = NULL;
 
@@ -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

doc/examples/demuxing_decoding.c

@@ -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,

doc/examples/filtering_video.c

@@ -38,7 +38,10 @@
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>
 
-const char *filter_descr = "scale=78:24";
+const char *filter_descr = "scale=78:24,transpose=cclock";
+/* other way:
+   scale=78:24 [scl]; [scl] transpose=cclock // assumes "[in]" and "[out]" to be input output pads respectively
+ */
 
 static AVFormatContext *fmt_ctx;
 static AVCodecContext *dec_ctx;

doc/examples/http_multiclient.c

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

doc/examples/muxing.c

@@ -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;

doc/examples/qsvdec.c

@@ -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;

doc/examples/remuxing.c

@@ -101,7 +101,7 @@ int main(int argc, char **argv)
         }
         out_stream->codec->codec_tag = 0;
         if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-            out_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
+            out_stream->codec->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
     }
     av_dump_format(ofmt_ctx, 0, out_filename, 1);
 

doc/examples/transcode_aac.c

@@ -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) {

doc/examples/transcoding.c

@@ -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) {

doc/ffmpeg.texi

@@ -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
@@ -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})
@@ -690,6 +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
@@ -1082,7 +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})
@@ -1193,9 +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.
@@ -1299,47 +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
 

doc/ffplay.texi

@@ -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

doc/ffprobe.texi

@@ -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

doc/fftools-common-opts.texi

@@ -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}]

doc/filter_design.txt

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

doc/formats.texi

@@ -205,6 +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

doc/general.texi

@@ -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
@@ -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
@@ -243,6 +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
@@ -273,6 +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
@@ -477,6 +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
@@ -663,6 +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
@@ -671,7 +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
@@ -836,7 +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
@@ -876,7 +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.
@@ -954,8 +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

doc/git-howto.texi

@@ -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
@@ -32,29 +32,28 @@ 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.
 
 
 @section Cloning the source tree
 
 @example
-git clone git://source.ffmpeg.org/ffmpeg <target>
+git clone https://git.ffmpeg.org/ffmpeg.git <target>
 @end example
 
 This will put the FFmpeg sources into the directory @var{<target>}.
@@ -108,7 +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.
 
@@ -128,8 +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
 
@@ -150,6 +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.
@@ -172,14 +172,21 @@ to make sure you don't have untracked files or deletions.
 git add [-i|-p|-A] <filenames/dirnames>
 @end example
 
-Make sure you have told git your name and email address
+Make sure you have told Git your name, email address and GPG key
 
 @example
 git config --global user.name "My Name"
 git config --global user.email my@@email.invalid
+git config --global user.signingkey ABCDEF0123245
 @end example
 
-Use @var{--global} to set the global configuration for all your git checkouts.
+Enable signing all commits or use -S
+
+@example
+git config --global commit.gpgsign true
+@end example
+
+Use @option{--global} to set the global configuration for all your Git checkouts.
 
 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
@@ -210,7 +217,7 @@ 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
 
@@ -326,10 +333,12 @@ 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 @ref{Updating the source tree to the latest revision}.
 
@@ -350,23 +359,24 @@ 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
@@ -375,7 +385,20 @@ git checkout -b svn_23456 $SHA1
 where @var{$SHA1} is the commit hash from the @command{git log} output.
 
 
-@chapter pre-push checklist
+@chapter gpg key generation
+
+If you have no gpg key yet, we recommend that you create a ed25519 based key as it
+is small, fast and secure. Especially it results in small signatures in git.
+
+@example
+gpg --default-new-key-algo "ed25519/cert,sign+cv25519/encr" --quick-generate-key "human@@server.com"
+@end example
+
+When generating a key, make sure the email specified matches the email used in git as some sites like
+github consider mismatches a reason to declare such commits unverified. After generating a key you
+can add it to the MAINTAINER file and upload it to a keyserver.
+
+@chapter Pre-push checklist
 
 Once you have a set of commits that you feel are ready for pushing,
 work through the following checklist to doublecheck everything is in
@@ -386,7 +409,7 @@ 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.
 
@@ -395,7 +418,7 @@ 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
 
@@ -413,5 +436,5 @@ recommended.
 
 @chapter Server Issues
 
-Contact the project admins @email{root@@ffmpeg.org} if you have technical
-problems with the GIT server.
+Contact the project admins at @email{root@@ffmpeg.org} if you have technical
+problems with the Git server.

doc/indevs.texi

@@ -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 +175,36 @@ $ 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
@@ -211,18 +266,6 @@ Capture video clip at 1080i50 10 bit:
 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
 
 @section dshow
@@ -275,11 +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
@@ -429,6 +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.
@@ -441,18 +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
 
@@ -638,6 +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.
@@ -678,6 +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
@@ -879,6 +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.
@@ -919,6 +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
@@ -954,6 +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.
@@ -971,6 +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.
@@ -1093,6 +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
@@ -1103,6 +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.

doc/mips.txt

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

doc/multithreading.txt

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

doc/muxers.texi

@@ -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

doc/platform.texi

@@ -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

doc/protocols.texi

@@ -19,6 +19,18 @@ supported protocols.
 
 A description of the currently available protocols follows.
 
+@section async
+
+Asynchronous data filling wrapper for input stream.
+
+Fill data in a background thread, to decouple I/O operation from demux thread.
+
+@example
+async:@var{URL}
+async:http://host/resource
+async:cache:http://host/resource
+@end example
+
 @section bluray
 
 Read BluRay playlist.
@@ -292,6 +304,8 @@ autodetection in the future.
 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}

doc/scaler.texi

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

doc/texi2pod.pl

@@ -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;

doc/utils.texi

@@ -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}
@@ -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}.

doc/writing_filters.txt

@@ -420,4 +420,4 @@ done:
 
 When all of this is done, you can submit your patch to the ffmpeg-devel
 mailing-list for review.  If you need any help, feel free to come on our IRC
-channel, #ffmpeg-devel on irc.freenode.net.
+channel, #ffmpeg-devel on irc.libera.chat.

ffmpeg.c

@@ -49,6 +49,7 @@
 #include "libavutil/parseutils.h"
 #include "libavutil/samplefmt.h"
 #include "libavutil/fifo.h"
+#include "libavutil/internal.h"
 #include "libavutil/intreadwrite.h"
 #include "libavutil/dict.h"
 #include "libavutil/mathematics.h"
@@ -79,6 +80,10 @@
 #include <windows.h>
 #include <psapi.h>
 #endif
+#if HAVE_SETCONSOLECTRLHANDLER
+#include <windows.h>
+#endif
+
 
 #if HAVE_SYS_SELECT_H
 #include <sys/select.h>
@@ -132,8 +137,6 @@ AVIOContext *progress_avio = NULL;
 
 static uint8_t *subtitle_out;
 
-#define DEFAULT_PASS_LOGFILENAME_PREFIX "ffmpeg2pass"
-
 InputStream **input_streams = NULL;
 int        nb_input_streams = 0;
 InputFile   **input_files   = NULL;
@@ -169,8 +172,8 @@ static int sub2video_get_blank_frame(InputStream *ist)
     AVFrame *frame = ist->sub2video.frame;
 
     av_frame_unref(frame);
-    ist->sub2video.frame->width  = ist->sub2video.w;
-    ist->sub2video.frame->height = ist->sub2video.h;
+    ist->sub2video.frame->width  = ist->dec_ctx->width  ? ist->dec_ctx->width  : ist->sub2video.w;
+    ist->sub2video.frame->height = ist->dec_ctx->height ? ist->dec_ctx->height : ist->sub2video.h;
     ist->sub2video.frame->format = AV_PIX_FMT_RGB32;
     if ((ret = av_frame_get_buffer(frame, 32)) < 0)
         return ret;
@@ -190,7 +193,9 @@ static void sub2video_copy_rect(uint8_t *dst, int dst_linesize, int w, int h,
         return;
     }
     if (r->x < 0 || r->x + r->w > w || r->y < 0 || r->y + r->h > h) {
-        av_log(NULL, AV_LOG_WARNING, "sub2video: rectangle overflowing\n");
+        av_log(NULL, AV_LOG_WARNING, "sub2video: rectangle (%d %d %d %d) overflowing %d %d\n",
+            r->x, r->y, r->w, r->h, w, h
+        );
         return;
     }
 
@@ -222,7 +227,6 @@ static void sub2video_push_ref(InputStream *ist, int64_t pts)
 
 static void sub2video_update(InputStream *ist, AVSubtitle *sub)
 {
-    int w = ist->sub2video.w, h = ist->sub2video.h;
     AVFrame *frame = ist->sub2video.frame;
     int8_t *dst;
     int     dst_linesize;
@@ -250,7 +254,7 @@ static void sub2video_update(InputStream *ist, AVSubtitle *sub)
     dst          = frame->data    [0];
     dst_linesize = frame->linesize[0];
     for (i = 0; i < num_rects; i++)
-        sub2video_copy_rect(dst, dst_linesize, w, h, sub->rects[i]);
+        sub2video_copy_rect(dst, dst_linesize, frame->width, frame->height, sub->rects[i]);
     sub2video_push_ref(ist, pts);
     ist->sub2video.end_pts = end_pts;
 }
@@ -291,7 +295,7 @@ static void sub2video_flush(InputStream *ist)
     if (ist->sub2video.end_pts < INT64_MAX)
         sub2video_update(ist, NULL);
     for (i = 0; i < ist->nb_filters; i++)
-        av_buffersrc_add_ref(ist->filters[i]->filter, NULL, 0);
+        av_buffersrc_add_frame(ist->filters[i]->filter, NULL);
 }
 
 /* end of sub2video hack */
@@ -313,6 +317,7 @@ void term_exit(void)
 static volatile int received_sigterm = 0;
 static volatile int received_nb_signals = 0;
 static volatile int transcode_init_done = 0;
+static volatile int ffmpeg_exited = 0;
 static int main_return_code = 0;
 
 static void
@@ -321,10 +326,46 @@ sigterm_handler(int sig)
     received_sigterm = sig;
     received_nb_signals++;
     term_exit_sigsafe();
-    if(received_nb_signals > 3)
+    if(received_nb_signals > 3) {
+        write(2/*STDERR_FILENO*/, "Received > 3 system signals, hard exiting\n",
+                           strlen("Received > 3 system signals, hard exiting\n"));
+
         exit(123);
+    }
 }
 
+#if HAVE_SETCONSOLECTRLHANDLER
+static BOOL WINAPI CtrlHandler(DWORD fdwCtrlType)
+{
+    av_log(NULL, AV_LOG_DEBUG, "\nReceived windows signal %ld\n", fdwCtrlType);
+
+    switch (fdwCtrlType)
+    {
+    case CTRL_C_EVENT:
+    case CTRL_BREAK_EVENT:
+        sigterm_handler(SIGINT);
+        return TRUE;
+
+    case CTRL_CLOSE_EVENT:
+    case CTRL_LOGOFF_EVENT:
+    case CTRL_SHUTDOWN_EVENT:
+        sigterm_handler(SIGTERM);
+        /* Basically, with these 3 events, when we return from this method the
+           process is hard terminated, so stall as long as we need to
+           to try and let the main thread(s) clean up and gracefully terminate
+           (we have at most 5 seconds, but should be done far before that). */
+        while (!ffmpeg_exited) {
+            Sleep(0);
+        }
+        return TRUE;
+
+    default:
+        av_log(NULL, AV_LOG_ERROR, "Received unknown windows signal %ld\n", fdwCtrlType);
+        return FALSE;
+    }
+}
+#endif
+
 void term_init(void)
 {
 #if HAVE_TERMIOS_H
@@ -358,6 +399,9 @@ void term_init(void)
 #ifdef SIGXCPU
     signal(SIGXCPU, sigterm_handler);
 #endif
+#if HAVE_SETCONSOLECTRLHANDLER
+    SetConsoleCtrlHandler((PHANDLER_ROUTINE) CtrlHandler, TRUE);
+#endif
 }
 
 /* read a key without blocking */
@@ -391,10 +435,6 @@ static int read_key(void)
         is_pipe = !GetConsoleMode(input_handle, &dw);
     }
 
-    if (stdin->_cnt > 0) {
-        read(0, &ch, 1);
-        return ch;
-    }
     if (is_pipe) {
         /* When running under a GUI, you will end here. */
         if (!PeekNamedPipe(input_handle, NULL, 0, NULL, &nchars, NULL)) {
@@ -429,7 +469,7 @@ static void ffmpeg_cleanup(int ret)
 
     if (do_benchmark) {
         int maxrss = getmaxrss() / 1024;
-        printf("bench: maxrss=%ikB\n", maxrss);
+        av_log(NULL, AV_LOG_INFO, "bench: maxrss=%ikB\n", maxrss);
     }
 
     for (i = 0; i < nb_filtergraphs; i++) {
@@ -456,7 +496,10 @@ static void ffmpeg_cleanup(int ret)
     /* close files */
     for (i = 0; i < nb_output_files; i++) {
         OutputFile *of = output_files[i];
-        AVFormatContext *s = of->ctx;
+        AVFormatContext *s;
+        if (!of)
+            continue;
+        s = of->ctx;
         if (s && s->oformat && !(s->oformat->flags & AVFMT_NOFILE))
             avio_closep(&s->pb);
         avformat_free_context(s);
@@ -466,7 +509,12 @@ static void ffmpeg_cleanup(int ret)
     }
     for (i = 0; i < nb_output_streams; i++) {
         OutputStream *ost = output_streams[i];
-        AVBitStreamFilterContext *bsfc = ost->bitstream_filters;
+        AVBitStreamFilterContext *bsfc;
+
+        if (!ost)
+            continue;
+
+        bsfc = ost->bitstream_filters;
         while (bsfc) {
             AVBitStreamFilterContext *next = bsfc->next;
             av_bitstream_filter_close(bsfc);
@@ -527,12 +575,13 @@ static void ffmpeg_cleanup(int ret)
     avformat_network_deinit();
 
     if (received_sigterm) {
-        av_log(NULL, AV_LOG_INFO, "Received signal %d: terminating.\n",
+        av_log(NULL, AV_LOG_INFO, "Exiting normally, received signal %d.\n",
                (int) received_sigterm);
     } else if (ret && transcode_init_done) {
         av_log(NULL, AV_LOG_INFO, "Conversion failed!\n");
     }
     term_exit();
+    ffmpeg_exited = 1;
 }
 
 void remove_avoptions(AVDictionary **a, AVDictionary *b)
@@ -569,7 +618,7 @@ static void update_benchmark(const char *fmt, ...)
             va_start(va, fmt);
             vsnprintf(buf, sizeof(buf), fmt, va);
             va_end(va);
-            printf("bench: %8"PRIu64" %s \n", t - current_time, buf);
+            av_log(NULL, AV_LOG_INFO, "bench: %8"PRIu64" %s \n", t - current_time, buf);
         }
         current_time = t;
     }
@@ -591,7 +640,7 @@ static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
     int ret;
 
     if (!ost->st->codec->extradata_size && ost->enc_ctx->extradata_size) {
-        ost->st->codec->extradata = av_mallocz(ost->enc_ctx->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
+        ost->st->codec->extradata = av_mallocz(ost->enc_ctx->extradata_size + AV_INPUT_BUFFER_PADDING_SIZE);
         if (ost->st->codec->extradata) {
             memcpy(ost->st->codec->extradata, ost->enc_ctx->extradata, ost->enc_ctx->extradata_size);
             ost->st->codec->extradata_size = ost->enc_ctx->extradata_size;
@@ -616,6 +665,20 @@ static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
         }
         ost->frame_number++;
     }
+    if (avctx->codec_type == AVMEDIA_TYPE_VIDEO) {
+        int i;
+        uint8_t *sd = av_packet_get_side_data(pkt, AV_PKT_DATA_QUALITY_STATS,
+                                              NULL);
+        ost->quality = sd ? AV_RL32(sd) : -1;
+        ost->pict_type = sd ? sd[4] : AV_PICTURE_TYPE_NONE;
+
+        for (i = 0; i<FF_ARRAY_ELEMS(ost->error); i++) {
+            if (sd && i < sd[5])
+                ost->error[i] = AV_RL64(sd + 8 + 8*i);
+            else
+                ost->error[i] = -1;
+        }
+    }
 
     if (bsfc)
         av_packet_split_side_data(pkt);
@@ -630,11 +693,17 @@ static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
                                            &new_pkt.data, &new_pkt.size,
                                            pkt->data, pkt->size,
                                            pkt->flags & AV_PKT_FLAG_KEY);
-        if(a == 0 && new_pkt.data != pkt->data && new_pkt.destruct) {
-            uint8_t *t = av_malloc(new_pkt.size + FF_INPUT_BUFFER_PADDING_SIZE); //the new should be a subset of the old so cannot overflow
+FF_DISABLE_DEPRECATION_WARNINGS
+        if(a == 0 && new_pkt.data != pkt->data
+#if FF_API_DESTRUCT_PACKET
+           && new_pkt.destruct
+#endif
+           ) {
+FF_ENABLE_DEPRECATION_WARNINGS
+            uint8_t *t = av_malloc(new_pkt.size + AV_INPUT_BUFFER_PADDING_SIZE); //the new should be a subset of the old so cannot overflow
             if(t) {
                 memcpy(t, new_pkt.data, new_pkt.size);
-                memset(t + new_pkt.size, 0, FF_INPUT_BUFFER_PADDING_SIZE);
+                memset(t + new_pkt.size, 0, AV_INPUT_BUFFER_PADDING_SIZE);
                 new_pkt.data = t;
                 new_pkt.buf = NULL;
                 a = 1;
@@ -650,6 +719,7 @@ static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
             if (!new_pkt.buf)
                 exit_program(1);
         } else if (a < 0) {
+            new_pkt = *pkt;
             av_log(NULL, AV_LOG_ERROR, "Failed to open bitstream filter %s for stream %d with codec %s",
                    bsfc->filter->name, pkt->stream_index,
                    avctx->codec ? avctx->codec->name : "copy");
@@ -1061,7 +1131,7 @@ static void do_video_out(AVFormatContext *s,
         int got_packet, forced_keyframe = 0;
         double pts_time;
 
-        if (enc->flags & (CODEC_FLAG_INTERLACED_DCT|CODEC_FLAG_INTERLACED_ME) &&
+        if (enc->flags & (AV_CODEC_FLAG_INTERLACED_DCT | AV_CODEC_FLAG_INTERLACED_ME) &&
             ost->top_field_first >= 0)
             in_picture->top_field_first = !!ost->top_field_first;
 
@@ -1087,7 +1157,7 @@ static void do_video_out(AVFormatContext *s,
             ost->forced_keyframes_expr_const_values[FKF_T] = pts_time;
             res = av_expr_eval(ost->forced_keyframes_pexpr,
                                ost->forced_keyframes_expr_const_values, NULL);
-            av_dlog(NULL, "force_key_frame: n:%f n_forced:%f prev_forced_n:%f t:%f prev_forced_t:%f -> res:%f\n",
+            ff_dlog(NULL, "force_key_frame: n:%f n_forced:%f prev_forced_n:%f t:%f prev_forced_t:%f -> res:%f\n",
                     ost->forced_keyframes_expr_const_values[FKF_N],
                     ost->forced_keyframes_expr_const_values[FKF_N_FORCED],
                     ost->forced_keyframes_expr_const_values[FKF_PREV_FORCED_N],
@@ -1140,7 +1210,7 @@ static void do_video_out(AVFormatContext *s,
                        av_ts2str(pkt.dts), av_ts2timestr(pkt.dts, &enc->time_base));
             }
 
-            if (pkt.pts == AV_NOPTS_VALUE && !(enc->codec->capabilities & CODEC_CAP_DELAY))
+            if (pkt.pts == AV_NOPTS_VALUE && !(enc->codec->capabilities & AV_CODEC_CAP_DELAY))
                 pkt.pts = ost->sync_opts;
 
             av_packet_rescale_ts(&pkt, enc->time_base, ost->st->time_base);
@@ -1176,8 +1246,10 @@ static void do_video_out(AVFormatContext *s,
     if (!ost->last_frame)
         ost->last_frame = av_frame_alloc();
     av_frame_unref(ost->last_frame);
-    if (next_picture)
+    if (next_picture && ost->last_frame)
         av_frame_ref(ost->last_frame, next_picture);
+    else
+        av_frame_free(&ost->last_frame);
 }
 
 static double psnr(double d)
@@ -1203,9 +1275,11 @@ static void do_video_stats(OutputStream *ost, int frame_size)
     enc = ost->enc_ctx;
     if (enc->codec_type == AVMEDIA_TYPE_VIDEO) {
         frame_number = ost->st->nb_frames;
-        fprintf(vstats_file, "frame= %5d q= %2.1f ", frame_number, enc->coded_frame ? enc->coded_frame->quality / (float)FF_QP2LAMBDA : 0);
-        if (enc->coded_frame && (enc->flags&CODEC_FLAG_PSNR))
-            fprintf(vstats_file, "PSNR= %6.2f ", psnr(enc->coded_frame->error[0] / (enc->width * enc->height * 255.0 * 255.0)));
+        fprintf(vstats_file, "frame= %5d q= %2.1f ", frame_number,
+                ost->quality / (float)FF_QP2LAMBDA);
+
+        if (ost->error[0]>=0 && (enc->flags & AV_CODEC_FLAG_PSNR))
+            fprintf(vstats_file, "PSNR= %6.2f ", psnr(ost->error[0] / (enc->width * enc->height * 255.0 * 255.0)));
 
         fprintf(vstats_file,"f_size= %6d ", frame_size);
         /* compute pts value */
@@ -1217,7 +1291,7 @@ static void do_video_stats(OutputStream *ost, int frame_size)
         avg_bitrate = (double)(ost->data_size * 8) / ti1 / 1000.0;
         fprintf(vstats_file, "s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s ",
                (double)ost->data_size / 1024, ti1, bitrate, avg_bitrate);
-        fprintf(vstats_file, "type= %c\n", enc->coded_frame ? av_get_picture_type_char(enc->coded_frame->pict_type) : 'I');
+        fprintf(vstats_file, "type= %c\n", av_get_picture_type_char(ost->pict_type));
     }
 }
 
@@ -1315,7 +1389,7 @@ static int reap_filters(int flush)
                 do_video_out(of->ctx, ost, filtered_frame, float_pts);
                 break;
             case AVMEDIA_TYPE_AUDIO:
-                if (!(enc->codec->capabilities & CODEC_CAP_PARAM_CHANGE) &&
+                if (!(enc->codec->capabilities & AV_CODEC_CAP_PARAM_CHANGE) &&
                     enc->channels != av_frame_get_channels(filtered_frame)) {
                     av_log(NULL, AV_LOG_ERROR,
                            "Audio filter graph output is not normalized and encoder does not support parameter changes\n");
@@ -1354,8 +1428,8 @@ static void print_final_stats(int64_t total_size)
         }
         extra_size += ost->enc_ctx->extradata_size;
         data_size  += ost->data_size;
-        if (   (ost->enc_ctx->flags & (CODEC_FLAG_PASS1 | CODEC_FLAG_PASS2))
-            != CODEC_FLAG_PASS1)
+        if (   (ost->enc_ctx->flags & (AV_CODEC_FLAG_PASS1 | CODEC_FLAG_PASS2))
+            != AV_CODEC_FLAG_PASS1)
             pass1_used = 0;
     }
 
@@ -1494,8 +1568,9 @@ static void print_report(int is_last_report, int64_t timer_start, int64_t cur_ti
         float q = -1;
         ost = output_streams[i];
         enc = ost->enc_ctx;
-        if (!ost->stream_copy && enc->coded_frame)
-            q = enc->coded_frame->quality / (float)FF_QP2LAMBDA;
+        if (!ost->stream_copy)
+            q = ost->quality / (float) FF_QP2LAMBDA;
+
         if (vid && enc->codec_type == AVMEDIA_TYPE_VIDEO) {
             snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "q=%2.1f ", q);
             av_bprintf(&buf_script, "stream_%d_%d_q=%.1f\n",
@@ -1522,7 +1597,8 @@ static void print_report(int is_last_report, int64_t timer_start, int64_t cur_ti
                 for (j = 0; j < 32; j++)
                     snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "%X", (int)lrintf(log2(qp_histogram[j] + 1)));
             }
-            if ((enc->flags&CODEC_FLAG_PSNR) && (enc->coded_frame || is_last_report)) {
+
+            if ((enc->flags & AV_CODEC_FLAG_PSNR) && (ost->pict_type != AV_PICTURE_TYPE_NONE || is_last_report)) {
                 int j;
                 double error, error_sum = 0;
                 double scale, scale_sum = 0;
@@ -1534,7 +1610,7 @@ static void print_report(int is_last_report, int64_t timer_start, int64_t cur_ti
                         error = enc->error[j];
                         scale = enc->width * enc->height * 255.0 * 255.0 * frame_number;
                     } else {
-                        error = enc->coded_frame->error[j];
+                        error = ost->error[j];
                         scale = enc->width * enc->height * 255.0 * 255.0;
                     }
                     if (j)
@@ -1673,7 +1749,9 @@ static void flush_encoders(void)
                 ret = encode(enc, &pkt, NULL, &got_packet);
                 update_benchmark("flush %s %d.%d", desc, ost->file_index, ost->index);
                 if (ret < 0) {
-                    av_log(NULL, AV_LOG_FATAL, "%s encoding failed\n", desc);
+                    av_log(NULL, AV_LOG_FATAL, "%s encoding failed: %s\n",
+                           desc,
+                           av_err2str(ret));
                     exit_program(1);
                 }
                 if (ost->logfile && enc->stats_out) {
@@ -1789,17 +1867,22 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
 
     opkt.duration = av_rescale_q(pkt->duration, ist->st->time_base, ost->st->time_base);
     opkt.flags    = pkt->flags;
-
     // FIXME remove the following 2 lines they shall be replaced by the bitstream filters
-    if (  ost->enc_ctx->codec_id != AV_CODEC_ID_H264
-       && ost->enc_ctx->codec_id != AV_CODEC_ID_MPEG1VIDEO
-       && ost->enc_ctx->codec_id != AV_CODEC_ID_MPEG2VIDEO
-       && ost->enc_ctx->codec_id != AV_CODEC_ID_VC1
+    if (  ost->st->codec->codec_id != AV_CODEC_ID_H264
+       && ost->st->codec->codec_id != AV_CODEC_ID_MPEG1VIDEO
+       && ost->st->codec->codec_id != AV_CODEC_ID_MPEG2VIDEO
+       && ost->st->codec->codec_id != AV_CODEC_ID_VC1
        ) {
-        if (av_parser_change(ost->parser, ost->st->codec,
+        int ret = av_parser_change(ost->parser, ost->st->codec,
                              &opkt.data, &opkt.size,
                              pkt->data, pkt->size,
-                             pkt->flags & AV_PKT_FLAG_KEY)) {
+                             pkt->flags & AV_PKT_FLAG_KEY);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_FATAL, "av_parser_change failed: %s\n",
+                   av_err2str(ret));
+            exit_program(1);
+        }
+        if (ret) {
             opkt.buf = av_buffer_create(opkt.data, opkt.size, av_buffer_default_free, NULL, 0);
             if (!opkt.buf)
                 exit_program(1);
@@ -1810,9 +1893,16 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
     }
     av_copy_packet_side_data(&opkt, pkt);
 
-    if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && (of->ctx->oformat->flags & AVFMT_RAWPICTURE)) {
+    if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO &&
+        ost->st->codec->codec_id == AV_CODEC_ID_RAWVIDEO &&
+        (of->ctx->oformat->flags & AVFMT_RAWPICTURE)) {
         /* store AVPicture in AVPacket, as expected by the output format */
-        avpicture_fill(&pict, opkt.data, ost->st->codec->pix_fmt, ost->st->codec->width, ost->st->codec->height);
+        int ret = avpicture_fill(&pict, opkt.data, ost->st->codec->pix_fmt, ost->st->codec->width, ost->st->codec->height);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_FATAL, "avpicture_fill failed: %s\n",
+                   av_err2str(ret));
+            exit_program(1);
+        }
         opkt.data = (uint8_t *)&pict;
         opkt.size = sizeof(AVPicture);
         opkt.flags |= AV_PKT_FLAG_KEY;
@@ -1863,20 +1953,14 @@ static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
         ret = AVERROR_INVALIDDATA;
     }
 
-    if (*got_output || ret<0 || pkt->size)
+    if (*got_output || ret<0)
         decode_error_stat[ret<0] ++;
 
-    if (!*got_output || ret < 0) {
-        if (!pkt->size) {
-            for (i = 0; i < ist->nb_filters; i++)
-#if 1
-                av_buffersrc_add_ref(ist->filters[i]->filter, NULL, 0);
-#else
-                av_buffersrc_add_frame(ist->filters[i]->filter, NULL);
-#endif
-        }
+    if (ret < 0 && exit_on_error)
+        exit_program(1);
+
+    if (!*got_output || ret < 0)
         return ret;
-    }
 
     ist->samples_decoded += decoded_frame->nb_samples;
     ist->frames_decoded++;
@@ -2000,17 +2084,21 @@ static int decode_video(InputStream *ist, AVPacket *pkt, int *got_output)
         if (ist->dec_ctx->codec_id == AV_CODEC_ID_H264) {
             ist->st->codec->has_b_frames = ist->dec_ctx->has_b_frames;
         } else
-            av_log_ask_for_sample(
-                ist->dec_ctx,
-                "has_b_frames is larger in decoder than demuxer %d > %d ",
-                ist->dec_ctx->has_b_frames,
-                ist->st->codec->has_b_frames
-            );
+            av_log(ist->dec_ctx, AV_LOG_WARNING,
+                   "has_b_frames is larger in decoder than demuxer %d > %d.\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)",
+                   ist->dec_ctx->has_b_frames,
+                   ist->st->codec->has_b_frames);
     }
 
-    if (*got_output || ret<0 || pkt->size)
+    if (*got_output || ret<0)
         decode_error_stat[ret<0] ++;
 
+    if (ret < 0 && exit_on_error)
+        exit_program(1);
+
     if (*got_output && ret >= 0) {
         if (ist->dec_ctx->width  != decoded_frame->width ||
             ist->dec_ctx->height != decoded_frame->height ||
@@ -2025,17 +2113,8 @@ static int decode_video(InputStream *ist, AVPacket *pkt, int *got_output)
         }
     }
 
-    if (!*got_output || ret < 0) {
-        if (!pkt->size) {
-            for (i = 0; i < ist->nb_filters; i++)
-#if 1
-                av_buffersrc_add_ref(ist->filters[i]->filter, NULL, 0);
-#else
-                av_buffersrc_add_frame(ist->filters[i]->filter, NULL);
-#endif
-        }
+    if (!*got_output || ret < 0)
         return ret;
-    }
 
     if(ist->top_field_first>=0)
         decoded_frame->top_field_first = ist->top_field_first;
@@ -2126,9 +2205,12 @@ static int transcode_subtitles(InputStream *ist, AVPacket *pkt, int *got_output)
     int i, ret = avcodec_decode_subtitle2(ist->dec_ctx,
                                           &subtitle, got_output, pkt);
 
-    if (*got_output || ret<0 || pkt->size)
+    if (*got_output || ret<0)
         decode_error_stat[ret<0] ++;
 
+    if (ret < 0 && exit_on_error)
+        exit_program(1);
+
     if (ret < 0 || !*got_output) {
         if (!pkt->size)
             sub2video_flush(ist);
@@ -2180,6 +2262,17 @@ out:
     return ret;
 }
 
+static int send_filter_eof(InputStream *ist)
+{
+    int i, ret;
+    for (i = 0; i < ist->nb_filters; i++) {
+        ret = av_buffersrc_add_frame(ist->filters[i]->filter, NULL);
+        if (ret < 0)
+            return ret;
+    }
+    return 0;
+}
+
 /* pkt = NULL means EOF (needed to flush decoder buffers) */
 static int process_input_packet(InputStream *ist, const AVPacket *pkt)
 {
@@ -2227,7 +2320,7 @@ static int process_input_packet(InputStream *ist, const AVPacket *pkt)
         ist->dts = ist->next_dts;
 
         if (avpkt.size && avpkt.size != pkt->size &&
-            !(ist->dec->capabilities & CODEC_CAP_SUBFRAMES)) {
+            !(ist->dec->capabilities & AV_CODEC_CAP_SUBFRAMES)) {
             av_log(NULL, ist->showed_multi_packet_warning ? AV_LOG_VERBOSE : AV_LOG_WARNING,
                    "Multiple frames in a packet from stream %d\n", pkt->stream_index);
             ist->showed_multi_packet_warning = 1;
@@ -2264,8 +2357,13 @@ static int process_input_packet(InputStream *ist, const AVPacket *pkt)
             return -1;
         }
 
-        if (ret < 0)
-            return ret;
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Error while decoding stream #%d:%d: %s\n",
+                   ist->file_index, ist->st->index, av_err2str(ret));
+            if (exit_on_error)
+                exit_program(1);
+            break;
+        }
 
         avpkt.dts=
         avpkt.pts= AV_NOPTS_VALUE;
@@ -2284,13 +2382,26 @@ static int process_input_packet(InputStream *ist, const AVPacket *pkt)
             break;
     }
 
+    /* after flushing, send an EOF on all the filter inputs attached to the stream */
+    if (!pkt && ist->decoding_needed && !got_output) {
+        int ret = send_filter_eof(ist);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_FATAL, "Error marking filters as finished\n");
+            exit_program(1);
+        }
+    }
+
     /* handle stream copy */
     if (!ist->decoding_needed) {
         ist->dts = ist->next_dts;
         switch (ist->dec_ctx->codec_type) {
         case AVMEDIA_TYPE_AUDIO:
-            ist->next_dts += ((int64_t)AV_TIME_BASE * ist->dec_ctx->frame_size) /
-                             ist->dec_ctx->sample_rate;
+            if (ist->dec_ctx->sample_rate) {
+                ist->next_dts += ((int64_t)AV_TIME_BASE * ist->dec_ctx->frame_size) /
+                                  ist->dec_ctx->sample_rate;
+            } else {
+                ist->next_dts += av_rescale_q(pkt->duration, ist->st->time_base, AV_TIME_BASE_Q);
+            }
             break;
         case AVMEDIA_TYPE_VIDEO:
             if (ist->framerate.num) {
@@ -2340,6 +2451,9 @@ static void print_sdp(void)
         }
     }
 
+    if (!j)
+        goto fail;
+
     av_sdp_create(avc, j, sdp, sizeof(sdp));
 
     if (!sdp_filename) {
@@ -2355,6 +2469,7 @@ static void print_sdp(void)
         }
     }
 
+fail:
     av_freep(&avc);
 }
 
@@ -2475,6 +2590,76 @@ static int compare_int64(const void *a, const void *b)
     return va < vb ? -1 : va > vb ? +1 : 0;
 }
 
+static int init_output_stream(OutputStream *ost, char *error, int error_len)
+{
+    int ret = 0;
+
+    if (ost->encoding_needed) {
+        AVCodec      *codec = ost->enc;
+        AVCodecContext *dec = NULL;
+        InputStream *ist;
+
+        if ((ist = get_input_stream(ost)))
+            dec = ist->dec_ctx;
+        if (dec && dec->subtitle_header) {
+            /* ASS code assumes this buffer is null terminated so add extra byte. */
+            ost->enc_ctx->subtitle_header = av_mallocz(dec->subtitle_header_size + 1);
+            if (!ost->enc_ctx->subtitle_header)
+                return AVERROR(ENOMEM);
+            memcpy(ost->enc_ctx->subtitle_header, dec->subtitle_header, dec->subtitle_header_size);
+            ost->enc_ctx->subtitle_header_size = dec->subtitle_header_size;
+        }
+        if (!av_dict_get(ost->encoder_opts, "threads", NULL, 0))
+            av_dict_set(&ost->encoder_opts, "threads", "auto", 0);
+        av_dict_set(&ost->encoder_opts, "side_data_only_packets", "1", 0);
+        if (ost->enc->type == AVMEDIA_TYPE_AUDIO &&
+            !codec->defaults &&
+            !av_dict_get(ost->encoder_opts, "b", NULL, 0) &&
+            !av_dict_get(ost->encoder_opts, "ab", NULL, 0))
+            av_dict_set(&ost->encoder_opts, "b", "128000", 0);
+
+        if ((ret = avcodec_open2(ost->enc_ctx, codec, &ost->encoder_opts)) < 0) {
+            if (ret == AVERROR_EXPERIMENTAL)
+                abort_codec_experimental(codec, 1);
+            snprintf(error, error_len,
+                     "Error while opening encoder for output stream #%d:%d - "
+                     "maybe incorrect parameters such as bit_rate, rate, width or height",
+                    ost->file_index, ost->index);
+            return ret;
+        }
+        if (ost->enc->type == AVMEDIA_TYPE_AUDIO &&
+            !(ost->enc->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE))
+            av_buffersink_set_frame_size(ost->filter->filter,
+                                            ost->enc_ctx->frame_size);
+        assert_avoptions(ost->encoder_opts);
+        if (ost->enc_ctx->bit_rate && ost->enc_ctx->bit_rate < 1000)
+            av_log(NULL, AV_LOG_WARNING, "The bitrate parameter is set too low."
+                                         " It takes bits/s as argument, not kbits/s\n");
+
+        ret = avcodec_copy_context(ost->st->codec, ost->enc_ctx);
+        if (ret < 0) {
+            av_log(NULL, AV_LOG_FATAL,
+                   "Error initializing the output stream codec context.\n");
+            exit_program(1);
+        }
+
+        // copy timebase while removing common factors
+        ost->st->time_base = av_add_q(ost->enc_ctx->time_base, (AVRational){0, 1});
+        ost->st->codec->codec= ost->enc_ctx->codec;
+    } else {
+        ret = av_opt_set_dict(ost->enc_ctx, &ost->encoder_opts);
+        if (ret < 0) {
+           av_log(NULL, AV_LOG_FATAL,
+                  "Error setting up codec context options.\n");
+           return ret;
+        }
+        // copy timebase while removing common factors
+        ost->st->time_base = av_add_q(ost->st->codec->time_base, (AVRational){0, 1});
+    }
+
+    return ret;
+}
+
 static void parse_forced_key_frames(char *kf, OutputStream *ost,
                                     AVCodecContext *avctx)
 {
@@ -2585,7 +2770,7 @@ static void set_encoder_id(OutputFile *of, OutputStream *ost)
     if (!encoder_string)
         exit_program(1);
 
-    if (!(format_flags & AVFMT_FLAG_BITEXACT) && !(codec_flags & CODEC_FLAG_BITEXACT))
+    if (!(format_flags & AVFMT_FLAG_BITEXACT) && !(codec_flags & AV_CODEC_FLAG_BITEXACT))
         av_strlcpy(encoder_string, LIBAVCODEC_IDENT " ", encoder_string_len);
     else
         av_strlcpy(encoder_string, "Lavc ", encoder_string_len);
@@ -2626,21 +2811,6 @@ static int transcode_init(void)
                 input_streams[j + ifile->ist_index]->start = av_gettime_relative();
     }
 
-    /* output stream init */
-    for (i = 0; i < nb_output_files; i++) {
-        oc = output_files[i]->ctx;
-        if (!oc->nb_streams && !(oc->oformat->flags & AVFMT_NOSTREAMS)) {
-            av_dump_format(oc, i, oc->filename, 1);
-            av_log(NULL, AV_LOG_ERROR, "Output file #%d does not contain any stream\n", i);
-            return AVERROR(EINVAL);
-        }
-    }
-
-    /* init complex filtergraphs */
-    for (i = 0; i < nb_filtergraphs; i++)
-        if ((ret = avfilter_graph_config(filtergraphs[i]->graph, NULL)) < 0)
-            return ret;
-
     /* for each output stream, we compute the right encoding parameters */
     for (i = 0; i < nb_output_streams; i++) {
         AVCodecContext *enc_ctx;
@@ -2677,7 +2847,7 @@ static int transcode_init(void)
 
             av_assert0(ist && !ost->filter);
 
-            extra_size = (uint64_t)dec_ctx->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE;
+            extra_size = (uint64_t)dec_ctx->extradata_size + AV_INPUT_BUFFER_PADDING_SIZE;
 
             if (extra_size > INT_MAX) {
                 return AVERROR(EINVAL);
@@ -2716,7 +2886,8 @@ static int transcode_init(void)
              * overhead
              */
             if(!strcmp(oc->oformat->name, "avi")) {
-                if ( copy_tb<0 && av_q2d(ist->st->r_frame_rate) >= av_q2d(ist->st->avg_frame_rate)
+                if ( copy_tb<0 && ist->st->r_frame_rate.num
+                               && av_q2d(ist->st->r_frame_rate) >= av_q2d(ist->st->avg_frame_rate)
                                && 0.5/av_q2d(ist->st->r_frame_rate) > av_q2d(ist->st->time_base)
                                && 0.5/av_q2d(ist->st->r_frame_rate) > av_q2d(dec_ctx->time_base)
                                && av_q2d(ist->st->time_base) < 1.0/500 && av_q2d(dec_ctx->time_base) < 1.0/500
@@ -2752,7 +2923,7 @@ static int transcode_init(void)
                 enc_ctx->time_base = dec_ctx->time_base;
             }
 
-            if (ist && !ost->frame_rate.num)
+            if (!ost->frame_rate.num)
                 ost->frame_rate = ist->framerate;
             if(ost->frame_rate.num)
                 enc_ctx->time_base = av_inv_q(ost->frame_rate);
@@ -2849,10 +3020,6 @@ static int transcode_init(void)
                 goto dump_format;
             }
 
-            if (ist)
-                ist->decoding_needed |= DECODING_FOR_OST;
-            ost->encoding_needed = 1;
-
             set_encoder_id(output_files[ost->file_index], ost);
 
             if (!ost->filter &&
@@ -2981,39 +3148,6 @@ static int transcode_init(void)
                 abort();
                 break;
             }
-            /* two pass mode */
-            if (enc_ctx->flags & (CODEC_FLAG_PASS1 | CODEC_FLAG_PASS2)) {
-                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 (enc_ctx->flags & CODEC_FLAG_PASS2) {
-                        char  *logbuffer;
-                        size_t logbuffer_size;
-                        if (cmdutils_read_file(logfilename, &logbuffer, &logbuffer_size) < 0) {
-                            av_log(NULL, AV_LOG_FATAL, "Error reading log file '%s' for pass-2 encoding\n",
-                                   logfilename);
-                            exit_program(1);
-                        }
-                        enc_ctx->stats_in = logbuffer;
-                    }
-                    if (enc_ctx->flags & 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;
-                    }
-                }
-            }
         }
 
         if (ost->disposition) {
@@ -3050,63 +3184,9 @@ static int transcode_init(void)
 
     /* open each encoder */
     for (i = 0; i < nb_output_streams; i++) {
-        ost = output_streams[i];
-        if (ost->encoding_needed) {
-            AVCodec      *codec = ost->enc;
-            AVCodecContext *dec = NULL;
-
-            if ((ist = get_input_stream(ost)))
-                dec = ist->dec_ctx;
-            if (dec && dec->subtitle_header) {
-                /* ASS code assumes this buffer is null terminated so add extra byte. */
-                ost->enc_ctx->subtitle_header = av_mallocz(dec->subtitle_header_size + 1);
-                if (!ost->enc_ctx->subtitle_header) {
-                    ret = AVERROR(ENOMEM);
-                    goto dump_format;
-                }
-                memcpy(ost->enc_ctx->subtitle_header, dec->subtitle_header, dec->subtitle_header_size);
-                ost->enc_ctx->subtitle_header_size = dec->subtitle_header_size;
-            }
-            if (!av_dict_get(ost->encoder_opts, "threads", NULL, 0))
-                av_dict_set(&ost->encoder_opts, "threads", "auto", 0);
-            av_dict_set(&ost->encoder_opts, "side_data_only_packets", "1", 0);
-
-            if ((ret = avcodec_open2(ost->enc_ctx, codec, &ost->encoder_opts)) < 0) {
-                if (ret == AVERROR_EXPERIMENTAL)
-                    abort_codec_experimental(codec, 1);
-                snprintf(error, sizeof(error), "Error while opening encoder for output stream #%d:%d - maybe incorrect parameters such as bit_rate, rate, width or height",
-                        ost->file_index, ost->index);
-                goto dump_format;
-            }
-            if (ost->enc->type == AVMEDIA_TYPE_AUDIO &&
-                !(ost->enc->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE))
-                av_buffersink_set_frame_size(ost->filter->filter,
-                                             ost->enc_ctx->frame_size);
-            assert_avoptions(ost->encoder_opts);
-            if (ost->enc_ctx->bit_rate && ost->enc_ctx->bit_rate < 1000)
-                av_log(NULL, AV_LOG_WARNING, "The bitrate parameter is set too low."
-                                             " It takes bits/s as argument, not kbits/s\n");
-
-            ret = avcodec_copy_context(ost->st->codec, ost->enc_ctx);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_FATAL,
-                       "Error initializing the output stream codec context.\n");
-                exit_program(1);
-            }
-
-            // copy timebase while removing common factors
-            ost->st->time_base = av_add_q(ost->enc_ctx->time_base, (AVRational){0, 1});
-            ost->st->codec->codec= ost->enc_ctx->codec;
-        } else {
-            ret = av_opt_set_dict(ost->enc_ctx, &ost->encoder_opts);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_FATAL,
-                    "Error setting up codec context options.\n");
-                return ret;
-            }
-            // copy timebase while removing common factors
-            ost->st->time_base = av_add_q(ost->st->codec->time_base, (AVRational){0, 1});
-        }
+        ret = init_output_stream(output_streams[i], error, sizeof(error));
+        if (ret < 0)
+            goto dump_format;
     }
 
     /* init input streams */
@@ -3376,9 +3456,17 @@ static int check_keyboard_interaction(int64_t cur_time)
             if(!debug) debug = 1;
             while(debug & (FF_DEBUG_DCT_COEFF|FF_DEBUG_VIS_QP|FF_DEBUG_VIS_MB_TYPE)) //unsupported, would just crash
                 debug += debug;
-        }else
-            if(scanf("%d", &debug)!=1)
+        }else{
+            char buf[32];
+            int k = 0;
+            i = 0;
+            while ((k = read_key()) != '\n' && k != '\r' && i < sizeof(buf)-1)
+                if (k > 0)
+                    buf[i++] = k;
+            buf[i] = 0;
+            if (k <= 0 || sscanf(buf, "%d", &debug)!=1)
                 fprintf(stderr,"error parsing debug value\n");
+        }
         for(i=0;i<nb_input_streams;i++) {
             input_streams[i]->st->codec->debug = debug;
         }
@@ -3456,7 +3544,7 @@ static void free_input_threads(void)
         InputFile *f = input_files[i];
         AVPacket pkt;
 
-        if (!f->in_thread_queue)
+        if (!f || !f->in_thread_queue)
             continue;
         av_thread_message_queue_set_err_send(f->in_thread_queue, AVERROR_EOF);
         while (av_thread_message_queue_recv(f->in_thread_queue, &pkt, 0) >= 0)
@@ -3759,13 +3847,7 @@ static int process_input(int file_index)
 
     sub2video_heartbeat(ist, pkt.pts);
 
-    ret = process_input_packet(ist, &pkt);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Error while decoding stream #%d:%d: %s\n",
-               ist->file_index, ist->st->index, av_err2str(ret));
-        if (exit_on_error)
-            exit_program(1);
-    }
+    process_input_packet(ist, &pkt);
 
 discard_packet:
     av_free_packet(&pkt);
@@ -3984,6 +4066,7 @@ static int transcode(void)
                 av_freep(&ost->apad);
                 av_freep(&ost->disposition);
                 av_dict_free(&ost->encoder_opts);
+                av_dict_free(&ost->sws_dict);
                 av_dict_free(&ost->swr_opts);
                 av_dict_free(&ost->resample_opts);
                 av_dict_free(&ost->bsf_args);
@@ -4039,6 +4122,8 @@ int main(int argc, char **argv)
     int ret;
     int64_t ti;
 
+    init_dynload();
+
     register_exit(ffmpeg_cleanup);
 
     setvbuf(stderr,NULL,_IONBF,0); /* win32 runtime needs this */
@@ -4092,7 +4177,7 @@ int main(int argc, char **argv)
         exit_program(1);
     ti = getutime() - ti;
     if (do_benchmark) {
-        printf("bench: utime=%0.3fs\n", ti / 1000000.0);
+        av_log(NULL, AV_LOG_INFO, "bench: utime=%0.3fs\n", ti / 1000000.0);
     }
     av_log(NULL, AV_LOG_DEBUG, "%"PRIu64" frames successfully decoded, %"PRIu64" decoding errors\n",
            decode_error_stat[0], decode_error_stat[1]);

ffmpeg.h

@@ -63,6 +63,7 @@ enum HWAccelID {
     HWACCEL_VDPAU,
     HWACCEL_DXVA2,
     HWACCEL_VDA,
+    HWACCEL_VIDEOTOOLBOX,
 };
 
 typedef struct HWAccel {
@@ -92,6 +93,7 @@ typedef struct OptionsContext {
 
     /* input/output options */
     int64_t start_time;
+    int64_t start_time_eof;
     int seek_timestamp;
     const char *format;
 
@@ -229,6 +231,7 @@ typedef struct OutputFilter {
 
     /* temporary storage until stream maps are processed */
     AVFilterInOut       *out_tmp;
+    enum AVMediaType     type;
 } OutputFilter;
 
 typedef struct FilterGraph {
@@ -429,8 +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;
@@ -455,6 +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 {
@@ -509,6 +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;
 
@@ -536,11 +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 */

ffmpeg_filter.c

@@ -38,6 +38,28 @@
 #include "libavutil/imgutils.h"
 #include "libavutil/samplefmt.h"
 
+
+static const enum AVPixelFormat *get_compliance_unofficial_pix_fmts(enum AVCodecID codec_id, const enum AVPixelFormat default_formats[])
+{
+    static const enum AVPixelFormat mjpeg_formats[] =
+        { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P,
+          AV_PIX_FMT_YUV420P,  AV_PIX_FMT_YUV422P,
+          AV_PIX_FMT_NONE };
+    static const enum AVPixelFormat ljpeg_formats[] =
+        { AV_PIX_FMT_BGRA    ,
+          AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ444P, AV_PIX_FMT_YUVJ422P,
+          AV_PIX_FMT_YUV420P , AV_PIX_FMT_YUV444P , AV_PIX_FMT_YUV422P,
+          AV_PIX_FMT_NONE};
+
+    if (codec_id == AV_CODEC_ID_MJPEG) {
+        return mjpeg_formats;
+    } else if (codec_id == AV_CODEC_ID_LJPEG) {
+        return ljpeg_formats;
+    } else {
+        return default_formats;
+    }
+}
+
 enum AVPixelFormat choose_pixel_fmt(AVStream *st, AVCodecContext *enc_ctx, AVCodec *codec, enum AVPixelFormat target)
 {
     if (codec && codec->pix_fmts) {
@@ -45,18 +67,9 @@ enum AVPixelFormat choose_pixel_fmt(AVStream *st, AVCodecContext *enc_ctx, AVCod
         const AVPixFmtDescriptor *desc = av_pix_fmt_desc_get(target);
         int has_alpha = desc ? desc->nb_components % 2 == 0 : 0;
         enum AVPixelFormat best= AV_PIX_FMT_NONE;
-        static const enum AVPixelFormat mjpeg_formats[] =
-            { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUV420P, AV_PIX_FMT_YUV422P, AV_PIX_FMT_NONE };
-        static const enum AVPixelFormat ljpeg_formats[] =
-            { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUVJ444P, AV_PIX_FMT_YUV420P,
-              AV_PIX_FMT_YUV422P, AV_PIX_FMT_YUV444P, AV_PIX_FMT_BGRA, AV_PIX_FMT_NONE };
 
         if (enc_ctx->strict_std_compliance <= FF_COMPLIANCE_UNOFFICIAL) {
-            if (enc_ctx->codec_id == AV_CODEC_ID_MJPEG) {
-                p = mjpeg_formats;
-            } else if (enc_ctx->codec_id == AV_CODEC_ID_LJPEG) {
-                p =ljpeg_formats;
-            }
+            p = get_compliance_unofficial_pix_fmts(enc_ctx->codec_id, p);
         }
         for (; *p != AV_PIX_FMT_NONE; p++) {
             best= avcodec_find_best_pix_fmt_of_2(best, *p, target, has_alpha, NULL);
@@ -85,7 +98,7 @@ 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,
@@ -126,12 +139,7 @@ static char *choose_pix_fmts(OutputStream *ost)
 
         p = ost->enc->pix_fmts;
         if (ost->enc_ctx->strict_std_compliance <= FF_COMPLIANCE_UNOFFICIAL) {
-            if (ost->enc_ctx->codec_id == AV_CODEC_ID_MJPEG) {
-                p = (const enum AVPixelFormat[]) { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUV420P, AV_PIX_FMT_YUV422P, AV_PIX_FMT_NONE };
-            } else if (ost->enc_ctx->codec_id == AV_CODEC_ID_LJPEG) {
-                p = (const enum AVPixelFormat[]) { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUVJ444P, AV_PIX_FMT_YUV420P,
-                                                    AV_PIX_FMT_YUV422P, AV_PIX_FMT_YUV444P, AV_PIX_FMT_BGRA, AV_PIX_FMT_NONE };
-            }
+            p = get_compliance_unofficial_pix_fmts(ost->enc_ctx->codec_id, p);
         }
 
         for (; *p != AV_PIX_FMT_NONE; p++) {
@@ -289,6 +297,45 @@ 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)
@@ -384,11 +431,17 @@ 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"),
@@ -499,7 +552,7 @@ 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);
@@ -602,6 +655,11 @@ 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);
@@ -632,8 +690,8 @@ 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 */
@@ -688,7 +746,7 @@ 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,
@@ -904,7 +962,7 @@ 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;
 
@@ -917,7 +975,13 @@ 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;
@@ -947,14 +1011,30 @@ 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);
@@ -963,27 +1043,12 @@ 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;
 
@@ -991,7 +1056,7 @@ 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);
     }

ffmpeg_opt.c

@@ -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 },
 };
@@ -153,12 +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)
 {
@@ -230,6 +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;
@@ -274,6 +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);
@@ -311,8 +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);
@@ -647,9 +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;
@@ -922,6 +951,12 @@ 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 (!o->seek_timestamp && ic->start_time != AV_NOPTS_VALUE)
@@ -1203,12 +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;
     }
 
@@ -1216,9 +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)
@@ -1438,17 +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);
             }
         }
@@ -1458,6 +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);
@@ -1736,8 +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:
@@ -1770,13 +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;
@@ -1799,10 +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;
@@ -1857,8 +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;
@@ -1917,7 +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 &&
@@ -1934,12 +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;
                 }
             }
@@ -2026,6 +2100,7 @@ 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;
@@ -2049,6 +2124,9 @@ loop_end:
                         exit_program(1);
                     }
                 }
+                if (ost)
+                    ost->sync_ist = input_streams[  input_files[map->sync_file_index]->ist_index
+                                                  + map->sync_stream_index];
             }
         }
     }
@@ -2078,7 +2156,7 @@ 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->st->codec->extradata      = attachment;
@@ -2100,6 +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++) {
@@ -2142,6 +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)) {
@@ -2237,6 +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) {
@@ -2246,6 +2342,17 @@ 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++) {
@@ -2438,6 +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);
@@ -2771,8 +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,
@@ -2791,6 +2903,7 @@ static int open_files(OptionGroupList *l, const char *inout,
         if (ret < 0) {
             av_log(NULL, AV_LOG_ERROR, "Error parsing options for %s file "
                    "%s.\n", inout, g->arg);
+            uninit_options(&o);
             return ret;
         }
 
@@ -2838,6 +2951,13 @@ 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) {
@@ -2845,6 +2965,13 @@ 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) {
@@ -2919,6 +3046,9 @@ const OptionDef options[] = {
     { "ss",             HAS_ARG | OPT_TIME | OPT_OFFSET |
                         OPT_INPUT | OPT_OUTPUT,                      { .off = OFFSET(start_time) },
         "set the start time offset", "time_off" },
+    { "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" },
@@ -2956,8 +3086,8 @@ 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 },
@@ -3119,10 +3249,15 @@ 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" },

ffmpeg_vda.c

@@ -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,
-                  (const uint8_t **)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;
-}

ffmpeg_vdpau.c

@@ -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);

ffmpeg_videotoolbox.c

@@ -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;
+}

ffplay.c

@@ -31,7 +31,6 @@
 #include <stdint.h>
 
 #include "libavutil/avstring.h"
-#include "libavutil/colorspace.h"
 #include "libavutil/eval.h"
 #include "libavutil/mathematics.h"
 #include "libavutil/pixdesc.h"
@@ -66,7 +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
@@ -102,7 +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;
@@ -223,6 +224,9 @@ typedef struct VideoState {
     Decoder viddec;
     Decoder subdec;
 
+    int viddec_width;
+    int viddec_height;
+
     int audio_stream;
 
     int av_sync_type;
@@ -278,6 +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;
 
@@ -829,229 +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;
     }
 }
 
@@ -1306,6 +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);
 }
 
@@ -1475,11 +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;
@@ -1851,7 +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);
@@ -1893,6 +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);
@@ -1958,15 +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),
@@ -2328,8 +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)))
@@ -2348,14 +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 */
@@ -2448,6 +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);
@@ -2695,10 +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))
@@ -2771,6 +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;
@@ -2979,7 +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;
     }
@@ -3794,6 +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);
 

ffprobe.c

@@ -1732,6 +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];
@@ -2174,28 +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];
@@ -2831,6 +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);
@@ -3139,6 +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);
 

ffserver.c

@@ -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;
@@ -1166,8 +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;
@@ -1271,17 +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];
@@ -1399,7 +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);
 
@@ -1738,8 +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;
 }
@@ -1754,6 +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;
@@ -1804,8 +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);
@@ -1894,7 +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)
             {
@@ -1903,8 +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) {
@@ -1924,42 +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;
     }
 
@@ -2111,8 +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;
@@ -2120,11 +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;
 }
 
 
@@ -2151,7 +2156,8 @@ static int http_prepare_data(HTTPContext *c)
             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];
@@ -2216,23 +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;
@@ -2303,9 +2309,9 @@ static int http_prepare_data(HTTPContext *c)
                             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 {
+                    } else
                         ret = avio_open_dyn_buf(&ctx->pb);
-                    }
+
                     if (ret < 0) {
                         /* XXX: potential leak */
                         return -1;
@@ -2368,7 +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;
@@ -2449,8 +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
@@ -2534,9 +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),
@@ -2574,16 +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) {
@@ -2598,6 +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;
@@ -2616,7 +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? */
@@ -2822,10 +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;
@@ -2844,9 +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);
@@ -2855,9 +2863,8 @@ 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");
-    }
 
     avc->streams = av_malloc_array(avc->nb_streams, sizeof(*avc->streams));
     if (!avc->streams)
@@ -2887,7 +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",
@@ -3054,7 +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;
@@ -3115,8 +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)
 {
@@ -3139,10 +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] == '/' &&
@@ -3220,7 +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;
 
@@ -3275,9 +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)
@@ -3355,10 +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)
@@ -3395,12 +3406,12 @@ static AVStream *add_av_stream1(FFServerStream *stream,
             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);
@@ -3502,7 +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;
-                    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;
@@ -3532,7 +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);
             }
 
@@ -3554,7 +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);
@@ -3581,16 +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]);
         }
     }
 
@@ -3757,24 +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);
         }
     }
 
@@ -3808,6 +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);

ffserver_config.c

@@ -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);
@@ -923,7 +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")) {

libavcodec/012v.c

@@ -131,8 +131,8 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
             u = x/2 + (uint16_t *)(pic->data[1] + line * pic->linesize[1]);
             v = x/2 + (uint16_t *)(pic->data[2] + line * pic->linesize[2]);
             memcpy(y, y_temp, sizeof(*y) * (width - x));
-            memcpy(u, u_temp, sizeof(*u) * (width - x + 1) / 2);
-            memcpy(v, v_temp, sizeof(*v) * (width - x + 1) / 2);
+            memcpy(u, u_temp, sizeof(*u) * ((width - x + 1) / 2));
+            memcpy(v, v_temp, sizeof(*v) * ((width - x + 1) / 2));
         }
 
         line_end += stride;
@@ -151,5 +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,
 };

libavcodec/4xm.c

@@ -158,7 +158,7 @@ typedef struct FourXContext {
 #define FIX_1_847759065 121095
 #define FIX_2_613125930 171254
 
-#define MULTIPLY(var, const) (((var) * (const)) >> 16)
+#define MULTIPLY(var, const) ((int)((var) * (unsigned)(const)) >> 16)
 
 static void idct(int16_t block[64])
 {
@@ -351,6 +351,8 @@ static int decode_p_block(FourXContext *f, uint16_t *dst, const uint16_t *src,
     index = size2index[log2h][log2w];
     av_assert0(index >= 0);
 
+    if (get_bits_left(&f->gb) < 1)
+        return AVERROR_INVALIDDATA;
     h     = 1 << log2h;
     code  = get_vlc2(&f->gb, block_type_vlc[1 - (f->version > 1)][index].table,
                      BLOCK_TYPE_VLC_BITS, 1);
@@ -496,9 +498,9 @@ static int decode_i_block(FourXContext *f, int16_t *block)
 {
     int code, i, j, level, val;
 
-    if (get_bits_left(&f->gb) < 2){
-        av_log(f->avctx, AV_LOG_ERROR, "%d bits left before decode_i_block()\n", get_bits_left(&f->gb));
-        return -1;
+    if (get_bits_left(&f->pre_gb) < 2) {
+        av_log(f->avctx, AV_LOG_ERROR, "%d bits left before decode_i_block()\n", get_bits_left(&f->pre_gb));
+        return AVERROR_INVALIDDATA;
     }
 
     /* DC coef */
@@ -523,6 +525,10 @@ static int decode_i_block(FourXContext *f, int16_t *block)
             break;
         if (code == 0xf0) {
             i += 16;
+            if (i >= 64) {
+                av_log(f->avctx, AV_LOG_ERROR, "run %d overflow\n", i);
+                return 0;
+            }
         } else {
             if (code & 0xf) {
                 level = get_xbits(&f->gb, code & 0xf);
@@ -559,7 +565,7 @@ 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]);
     }
@@ -732,7 +738,7 @@ static int decode_i2_frame(FourXContext *f, const uint8_t *buf, int length)
         for (x = 0; x < width; x += 16) {
             unsigned int color[4] = { 0 }, bits;
             if (buf_end - buf < 8)
-                return -1;
+                return AVERROR_INVALIDDATA;
             // warning following is purely guessed ...
             color[0] = bytestream2_get_le16u(&g3);
             color[1] = bytestream2_get_le16u(&g3);
@@ -878,16 +884,18 @@ static int decode_frame(AVCodecContext *avctx, void *data,
         }
 
         if (i >= CFRAME_BUFFER_COUNT) {
+            if (free_index < 0)
+                return AVERROR_INVALIDDATA;
             i             = free_index;
             f->cfrm[i].id = id;
         }
         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 +1034,5 @@ AVCodec ff_fourxm_decoder = {
     .init           = decode_init,
     .close          = decode_end,
     .decode         = decode_frame,
-    .capabilities   = CODEC_CAP_DR1,
+    .capabilities   = AV_CODEC_CAP_DR1,
 };

libavcodec/8bps.c

@@ -70,6 +70,9 @@ static int decode_frame(AVCodecContext *avctx, void *data,
     unsigned char *planemap = c->planemap;
     int ret;
 
+    if (buf_size < planes * height *2)
+        return AVERROR_INVALIDDATA;
+
     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
         return ret;
 
@@ -120,12 +123,15 @@ 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 +190,5 @@ AVCodec ff_eightbps_decoder = {
     .priv_data_size = sizeof(EightBpsContext),
     .init           = decode_init,
     .decode         = decode_frame,
-    .capabilities   = CODEC_CAP_DR1,
+    .capabilities   = AV_CODEC_CAP_DR1,
 };

libavcodec/8svx.c

@@ -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 },
 };

libavcodec/Makefile

@@ -13,6 +13,7 @@ HEADERS = avcodec.h                                                     \
           vda.h                                                         \
           vdpau.h                                                       \
           version.h                                                     \
+          videotoolbox.h                                                \
           vorbis_parser.h                                               \
           xvmc.h                                                        \
 
@@ -56,6 +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-$(CONFIG_H263DSP)                 += h263dsp.o
@@ -71,6 +73,7 @@ OBJS-$(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-$(CONFIG_LLAUDDSP)                += lossless_audiodsp.o
@@ -88,10 +91,11 @@ OBJS-$(CONFIG_MPEGAUDIODSP)            += mpegaudiodsp.o                \
                                           mpegaudiodsp_float.o
 OBJS-$(CONFIG_MPEGVIDEO)               += mpegvideo.o mpegvideodsp.o rl.o \
                                           mpegvideo_motion.o mpegutils.o \
-                                          mpegvideodata.o
+                                          mpegvideodata.o mpegpicture.o
 OBJS-$(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
@@ -101,23 +105,36 @@ 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
@@ -202,6 +219,7 @@ OBJS-$(CONFIG_DCA_DECODER)             += dcadec.o dca.o dcadsp.o      \
                                           dcadata.o dca_exss.o         \
                                           dca_xll.o synth_filter.o
 OBJS-$(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-$(CONFIG_DFA_DECODER)             += dfa.o
@@ -243,8 +261,8 @@ OBJS-$(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
@@ -253,7 +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-$(CONFIG_G2M_DECODER)             += g2meet.o
+OBJS-$(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
@@ -266,20 +284,24 @@ 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)        += qsvdec_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-$(CONFIG_HQ_HQA_DECODER)          += hq_hqa.o hq_hqadata.o hq_hqadsp.o \
                                           canopus.o
@@ -293,8 +315,8 @@ OBJS-$(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.o ivi_dsp.o
-OBJS-$(CONFIG_INDEO5_DECODER)          += indeo5.o ivi.o ivi_dsp.o
+OBJS-$(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
@@ -346,6 +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-$(CONFIG_MSMPEG4V1_DECODER)       += msmpeg4dec.o msmpeg4.o msmpeg4data.o
@@ -354,13 +378,13 @@ OBJS-$(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-$(CONFIG_MXPEG_DECODER)           += mxpegdec.o
@@ -424,8 +448,8 @@ OBJS-$(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
@@ -482,7 +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-$(CONFIG_UTVIDEO_ENCODER)         += utvideoenc.o utvideo.o
@@ -502,6 +526,7 @@ OBJS-$(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
@@ -511,12 +536,11 @@ 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_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-$(CONFIG_VPLAYER_DECODER)         += textdec.o ass.o
@@ -538,9 +562,9 @@ OBJS-$(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
@@ -676,27 +700,35 @@ OBJS-$(CONFIG_VIMA_DECODER)               += vima.o adpcm_data.o
 OBJS-$(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
@@ -757,6 +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
@@ -792,6 +825,7 @@ OBJS-$(CONFIG_LIBVPX_VP9_ENCODER)         += libvpxenc.o libvpx.o
 OBJS-$(CONFIG_LIBWAVPACK_ENCODER)         += libwavpackenc.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
@@ -817,16 +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-$(CONFIG_MPEGAUDIO_PARSER)        += mpegaudio_parser.o \
                                           mpegaudiodecheader.o mpegaudiodata.o
@@ -849,6 +884,7 @@ OBJS-$(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
 OBJS-$(CONFIG_MJPEGA_DUMP_HEADER_BSF)     += mjpega_dump_header_bsf.o
@@ -873,7 +909,6 @@ SKIPHEADERS                            += %_tablegen.h                  \
                                           %_tables.h                    \
                                           aac_tablegen_decl.h           \
                                           fft-internal.h                \
-                                          libutvideo.h                  \
                                           old_codec_ids.h               \
                                           tableprint.h                  \
                                           tableprint_vlc.h              \
@@ -883,21 +918,22 @@ SKIPHEADERS-$(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 += api-flac
-
 TESTPROGS-$(CONFIG_CABAC)                 += cabac
 TESTPROGS-$(CONFIG_FFT)                   += fft fft-fixed fft-fixed32
 TESTPROGS-$(CONFIG_IDCTDSP)               += dct
@@ -913,9 +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                                                 \
@@ -924,6 +963,7 @@ HOSTPROGS = aac_tablegen                                                \
             pcm_tablegen                                                \
             qdm2_tablegen                                               \
             sinewin_tablegen                                            \
+            sinewin_fixed_tablegen                                      \
 
 CLEANFILES = *_tables.c *_tables.h *_tablegen$(HOSTEXESUF)
 
@@ -942,8 +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))
 
@@ -952,13 +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

libavcodec/a64multienc.c

@@ -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

libavcodec/aac.h

@@ -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
@@ -151,7 +161,7 @@ typedef struct PredictorState {
 typedef struct LongTermPrediction {
     int8_t present;
     int16_t lag;
-    float coef;
+    INTFLOAT coef;
     int8_t used[MAX_LTP_LONG_SFB];
 } LongTermPrediction;
 
@@ -173,7 +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;
 
 /**
@@ -185,7 +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;
 
 /**
@@ -222,7 +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;
 
 /**
@@ -233,17 +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;
 
 /**
@@ -254,7 +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
@@ -288,7 +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];
     /** @} */
 
     /**
@@ -299,8 +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;
     /** @} */
 
@@ -320,7 +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;
@@ -328,11 +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);
 
 };
 

libavcodec/aac_adtstoasc_bsf.c

@@ -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);

libavcodec/aac_defines.h

@@ -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 */

libavcodec/aac_parser.c

@@ -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);

libavcodec/aac_tablegen.c

@@ -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;
 }

libavcodec/aac_tablegen.h

@@ -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 */
 

libavcodec/aac_tablegen_decl.h

@@ -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 */

libavcodec/aacdec_fixed.c

@@ -0,0 +1,463 @@
+/*
+ * 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) & 8191];
+        else
+            coef =  (int)cbrt_tab[  coef  & 8191];
+        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 s = -scale;
+    unsigned int round;
+    int i, out, c = exp2tab[s & 3];
+    int nlz = 0;
+
+    av_assert0(s >= 0);
+    while (band_energy > 0x7fff) {
+        band_energy >>= 1;
+        nlz++;
+    }
+    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);
+        }
+    }
+    else {
+        s = s + 32;
+        if (s > 0) {
+            round = 1 << (s-1);
+            for (i=0; i<len; i++) {
+                out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
+                coefs[i] = -out;
+            }
+        } else {
+            for (i=0; i<len; i++)
+                coefs[i] = -(int64_t)coefs[i] * c * (1 << -s);
+        }
+    }
+}
+
+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,
+};

libavcodec/aacdectab.h

@@ -38,46 +38,11 @@
 /* @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 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, 4, 5, 0, 5, 0 };
 
 static const uint8_t aac_channel_layout_map[16][5][3] = {

libavcodec/aacenc.c

@@ -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);
 }
@@ -389,7 +316,7 @@ static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
                                  SingleChannelElement *sce)
 {
     int diff, off_sf = sce->sf_idx[0], off_pns = sce->sf_idx[0] - NOISE_OFFSET;
-    int noise_flag = 1;
+    int off_is = 0, noise_flag = 1;
     int i, w;
 
     for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
@@ -402,6 +329,10 @@ static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
                         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];
@@ -447,17 +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.
  */
@@ -466,12 +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;
@@ -529,9 +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;
@@ -558,6 +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);
@@ -585,25 +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));
@@ -613,16 +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) {
@@ -635,22 +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;
                 }
@@ -667,7 +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;
@@ -686,7 +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);
@@ -710,6 +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);
@@ -723,7 +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);
 
@@ -746,7 +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;
@@ -772,16 +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,
@@ -800,8 +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++)
@@ -811,6 +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);
@@ -819,9 +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);
 
@@ -837,14 +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 = 0}, 0, 1, AACENC_FLAGS, "aac_pns"},
-        {"disable",  "Disable PNS", 0, AV_OPT_TYPE_CONST, {.i64 =  0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
-        {"enable",   "Enable PNS (Proof of concept)",  0, AV_OPT_TYPE_CONST, {.i64 =  1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
+    {"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}
 };
 
@@ -855,13 +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)"),
@@ -872,8 +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,

libavcodec/aacenc.h

@@ -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,
@@ -43,6 +45,9 @@ typedef struct AACEncOptions {
     int stereo_mode;
     int aac_coder;
     int pns;
+    int tns;
+    int pred;
+    int intensity_stereo;
 } AACEncOptions;
 
 struct AACEncContext;
@@ -52,9 +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[];
@@ -71,6 +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
@@ -91,8 +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 */

libavcodec/aacenc_is.c

@@ -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;
+}

libavcodec/aacenc_is.h

@@ -0,0 +1,50 @@
+/*
+ * 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 )
+ */
+
+#ifndef AVCODEC_AACENC_IS_H
+#define AVCODEC_AACENC_IS_H
+
+#include "aacenc.h"
+
+/** Frequency in Hz for lower limit of intensity stereo **/
+#define INT_STEREO_LOW_LIMIT 6100
+
+struct AACISError {
+    int pass;    /* 1 if dist2 <= dist1  */
+    int phase;   /* -1 or +1             */
+    float error; /* fabs(dist1 - dist2)  */
+    float dist1; /* From original coeffs */
+    float dist2; /* From IS'd coeffs     */
+};
+
+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);
+void ff_aac_search_for_is(AACEncContext *s, AVCodecContext *avctx, ChannelElement *cpe);
+
+#endif /* AVCODEC_AACENC_IS_H */

libavcodec/aacenc_pred.c

@@ -0,0 +1,342 @@
+/*
+ * AAC encoder main-type prediction
+ * 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 "aactab.h"
+#include "aacenc_pred.h"
+#include "aacenc_utils.h"
+#include "aacenc_is.h"            /* <- Needed for common window distortions */
+#include "aacenc_quantization.h"
+
+#define RESTORE_PRED(sce, sfb) \
+        if (sce->ics.prediction_used[sfb]) {\
+            sce->ics.prediction_used[sfb] = 0;\
+            sce->band_type[sfb] = sce->band_alt[sfb];\
+        }
+
+static inline float flt16_round(float pf)
+{
+    union av_intfloat32 tmp;
+    tmp.f = pf;
+    tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
+    return tmp.f;
+}
+
+static inline float flt16_even(float pf)
+{
+    union av_intfloat32 tmp;
+    tmp.f = pf;
+    tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
+    return tmp.f;
+}
+
+static inline float flt16_trunc(float pf)
+{
+    union av_intfloat32 pun;
+    pun.f = pf;
+    pun.i &= 0xFFFF0000U;
+    return pun.f;
+}
+
+static inline void predict(PredictorState *ps, float *coef, float *rcoef, int set)
+{
+    float k2;
+    const float a     = 0.953125; // 61.0 / 64
+    const float alpha = 0.90625;  // 29.0 / 32
+    const float   k1 = ps->k1;
+    const float   r0 = ps->r0,     r1 = ps->r1;
+    const float cor0 = ps->cor0, cor1 = ps->cor1;
+    const float var0 = ps->var0, var1 = ps->var1;
+    const float e0 = *coef - ps->x_est;
+    const float e1 = e0 - k1 * r0;
+
+    if (set)
+        *coef = e0;
+
+    ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
+    ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
+    ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
+    ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
+    ps->r1   = flt16_trunc(a * (r0 - k1 * e0));
+    ps->r0   = flt16_trunc(a * e0);
+
+    /* Prediction for next frame */
+    ps->k1   = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0;
+    k2       = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0;
+    *rcoef   = ps->x_est = flt16_round(ps->k1*ps->r0 + k2*ps->r1);
+}
+
+static inline void reset_predict_state(PredictorState *ps)
+{
+    ps->r0    = 0.0f;
+    ps->r1    = 0.0f;
+    ps->k1    = 0.0f;
+    ps->cor0  = 0.0f;
+    ps->cor1  = 0.0f;
+    ps->var0  = 1.0f;
+    ps->var1  = 1.0f;
+    ps->x_est = 0.0f;
+}
+
+static inline void reset_all_predictors(PredictorState *ps)
+{
+    int i;
+    for (i = 0; i < MAX_PREDICTORS; i++)
+        reset_predict_state(&ps[i]);
+}
+
+static inline void reset_predictor_group(SingleChannelElement *sce, int group_num)
+{
+    int i;
+    PredictorState *ps = sce->predictor_state;
+    for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
+        reset_predict_state(&ps[i]);
+}
+
+void ff_aac_apply_main_pred(AACEncContext *s, SingleChannelElement *sce)
+{
+    int sfb, k;
+    const int pmax = FFMIN(sce->ics.max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
+
+    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
+        for (sfb = 0; sfb < pmax; sfb++) {
+            for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
+                predict(&sce->predictor_state[k], &sce->coeffs[k], &sce->prcoeffs[k],
+                        sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
+            }
+        }
+        if (sce->ics.predictor_reset_group) {
+            reset_predictor_group(sce, sce->ics.predictor_reset_group);
+        }
+    } else {
+        reset_all_predictors(sce->predictor_state);
+    }
+}
+
+/* If inc = 0 you can check if this returns 0 to see if you can reset freely */
+static inline int update_counters(IndividualChannelStream *ics, int inc)
+{
+    int i;
+    for (i = 1; i < 31; i++) {
+        ics->predictor_reset_count[i] += inc;
+        if (ics->predictor_reset_count[i] > PRED_RESET_FRAME_MIN)
+            return i; /* Reset this immediately */
+    }
+    return 0;
+}
+
+void ff_aac_adjust_common_prediction(AACEncContext *s, ChannelElement *cpe)
+{
+    int start, w, w2, g, i, count = 0;
+    SingleChannelElement *sce0 = &cpe->ch[0];
+    SingleChannelElement *sce1 = &cpe->ch[1];
+    const int pmax0 = FFMIN(sce0->ics.max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
+    const int pmax1 = FFMIN(sce1->ics.max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
+    const int pmax  = FFMIN(pmax0, pmax1);
+
+    if (!cpe->common_window ||
+        sce0->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE ||
+        sce1->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE)
+        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++) {
+            int sfb = w*16+g;
+            int sum = sce0->ics.prediction_used[sfb] + sce1->ics.prediction_used[sfb];
+            float ener0 = 0.0f, ener1 = 0.0f, ener01 = 0.0f;
+            struct AACISError ph_err1, ph_err2, *erf;
+            if (sfb < PRED_SFB_START || sfb > pmax || sum != 2) {
+                RESTORE_PRED(sce0, sfb);
+                RESTORE_PRED(sce1, sfb);
+                start += sce0->ics.swb_sizes[g];
+                continue;
+            }
+            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, 1, -1);
+            ph_err2 = ff_aac_is_encoding_err(s, cpe, start, w, g,
+                                             ener0, ener1, ener01, 1, +1);
+            erf = ph_err1.error < ph_err2.error ? &ph_err1 : &ph_err2;
+            if (erf->pass) {
+                sce0->ics.prediction_used[sfb] = 1;
+                sce1->ics.prediction_used[sfb] = 1;
+                count++;
+            } else {
+                RESTORE_PRED(sce0, sfb);
+                RESTORE_PRED(sce1, sfb);
+            }
+            start += sce0->ics.swb_sizes[g];
+        }
+    }
+
+    sce1->ics.predictor_present = sce0->ics.predictor_present = !!count;
+}
+
+static void update_pred_resets(SingleChannelElement *sce)
+{
+    int i, max_group_id_c, max_frame = 0;
+    float avg_frame = 0.0f;
+    IndividualChannelStream *ics = &sce->ics;
+
+    /* Update the counters and immediately update any frame behind schedule */
+    if ((ics->predictor_reset_group = update_counters(&sce->ics, 1)))
+        return;
+
+    for (i = 1; i < 31; i++) {
+        /* Count-based */
+        if (ics->predictor_reset_count[i] > max_frame) {
+            max_group_id_c = i;
+            max_frame = ics->predictor_reset_count[i];
+        }
+        avg_frame = (ics->predictor_reset_count[i] + avg_frame)/2;
+    }
+
+    if (max_frame > PRED_RESET_MIN) {
+        ics->predictor_reset_group = max_group_id_c;
+    } else {
+        ics->predictor_reset_group = 0;
+    }
+}
+
+void ff_aac_search_for_pred(AACEncContext *s, SingleChannelElement *sce)
+{
+    int sfb, i, count = 0, cost_coeffs = 0, cost_pred = 0;
+    const int pmax = FFMIN(sce->ics.max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
+    float *O34  = &s->scoefs[128*0], *P34 = &s->scoefs[128*1];
+    float *SENT = &s->scoefs[128*2], *S34 = &s->scoefs[128*3];
+    float *QERR = &s->scoefs[128*4];
+
+    if (sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+        sce->ics.predictor_present = 0;
+        return;
+    }
+
+    if (!sce->ics.predictor_initialized) {
+        reset_all_predictors(sce->predictor_state);
+        sce->ics.predictor_initialized = 1;
+        memcpy(sce->prcoeffs, sce->coeffs, 1024*sizeof(float));
+        for (i = 1; i < 31; i++)
+            sce->ics.predictor_reset_count[i] = i;
+    }
+
+    update_pred_resets(sce);
+    memcpy(sce->band_alt, sce->band_type, sizeof(sce->band_type));
+
+    for (sfb = PRED_SFB_START; sfb < pmax; sfb++) {
+        int cost1, cost2, cb_p;
+        float dist1, dist2, dist_spec_err = 0.0f;
+        const int cb_n = sce->band_type[sfb];
+        const int start_coef = sce->ics.swb_offset[sfb];
+        const int num_coeffs = sce->ics.swb_offset[sfb + 1] - start_coef;
+        const FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[sfb];
+
+        if (start_coef + num_coeffs > MAX_PREDICTORS)
+            continue;
+
+        /* Normal coefficients */
+        abs_pow34_v(O34, &sce->coeffs[start_coef], num_coeffs);
+        dist1 = quantize_and_encode_band_cost(s, NULL, &sce->coeffs[start_coef], NULL,
+                                              O34, num_coeffs, sce->sf_idx[sfb],
+                                              cb_n, s->lambda / band->threshold, INFINITY, &cost1, 0);
+        cost_coeffs += cost1;
+
+        /* Encoded coefficients - needed for #bits, band type and quant. error */
+        for (i = 0; i < num_coeffs; i++)
+            SENT[i] = sce->coeffs[start_coef + i] - sce->prcoeffs[start_coef + i];
+        abs_pow34_v(S34, SENT, num_coeffs);
+        if (cb_n < RESERVED_BT)
+            cb_p = find_min_book(find_max_val(1, num_coeffs, S34), sce->sf_idx[sfb]);
+        else
+            cb_p = cb_n;
+        quantize_and_encode_band_cost(s, NULL, SENT, QERR, S34, num_coeffs,
+                                      sce->sf_idx[sfb], cb_p, s->lambda / band->threshold, INFINITY,
+                                      &cost2, 0);
+
+        /* Reconstructed coefficients - needed for distortion measurements */
+        for (i = 0; i < num_coeffs; i++)
+            sce->prcoeffs[start_coef + i] += QERR[i] != 0.0f ? (sce->prcoeffs[start_coef + i] - QERR[i]) : 0.0f;
+        abs_pow34_v(P34, &sce->prcoeffs[start_coef], num_coeffs);
+        if (cb_n < RESERVED_BT)
+            cb_p = find_min_book(find_max_val(1, num_coeffs, P34), sce->sf_idx[sfb]);
+        else
+            cb_p = cb_n;
+        dist2 = quantize_and_encode_band_cost(s, NULL, &sce->prcoeffs[start_coef], NULL,
+                                              P34, num_coeffs, sce->sf_idx[sfb],
+                                              cb_p, s->lambda / band->threshold, INFINITY, NULL, 0);
+        for (i = 0; i < num_coeffs; i++)
+            dist_spec_err += (O34[i] - P34[i])*(O34[i] - P34[i]);
+        dist_spec_err *= s->lambda / band->threshold;
+        dist2 += dist_spec_err;
+
+        if (dist2 <= dist1 && cb_p <= cb_n) {
+            cost_pred += cost2;
+            sce->ics.prediction_used[sfb] = 1;
+            sce->band_alt[sfb]  = cb_n;
+            sce->band_type[sfb] = cb_p;
+            count++;
+        } else {
+            cost_pred += cost1;
+            sce->band_alt[sfb] = cb_p;
+        }
+    }
+
+    if (count && cost_coeffs < cost_pred) {
+        count = 0;
+        for (sfb = PRED_SFB_START; sfb < pmax; sfb++)
+            RESTORE_PRED(sce, sfb);
+        memset(&sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
+    }
+
+    sce->ics.predictor_present = !!count;
+}
+
+/**
+ * Encoder predictors data.
+ */
+void ff_aac_encode_main_pred(AACEncContext *s, SingleChannelElement *sce)
+{
+    int sfb;
+    IndividualChannelStream *ics = &sce->ics;
+    const int pmax = FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
+
+    if (!ics->predictor_present)
+        return;
+
+    put_bits(&s->pb, 1, !!ics->predictor_reset_group);
+    if (ics->predictor_reset_group)
+        put_bits(&s->pb, 5, ics->predictor_reset_group);
+    for (sfb = 0; sfb < pmax; sfb++)
+        put_bits(&s->pb, 1, ics->prediction_used[sfb]);
+}

libavcodec/aacenc_pred.h

@@ -0,0 +1,47 @@
+/*
+ * AAC encoder main-type prediction
+ * 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 main prediction
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#ifndef AVCODEC_AACENC_PRED_H
+#define AVCODEC_AACENC_PRED_H
+
+#include "aacenc.h"
+
+/* Every predictor group needs to get reset at least once in this many frames */
+#define PRED_RESET_FRAME_MIN 240
+
+/* Any frame with less than this amount of frames since last reset is ok */
+#define PRED_RESET_MIN 64
+
+/* Raise to filter any low frequency artifacts due to prediction */
+#define PRED_SFB_START 10
+
+void ff_aac_apply_main_pred(AACEncContext *s, SingleChannelElement *sce);
+void ff_aac_adjust_common_prediction(AACEncContext *s, ChannelElement *cpe);
+void ff_aac_search_for_pred(AACEncContext *s, SingleChannelElement *sce);
+void ff_aac_encode_main_pred(AACEncContext *s, SingleChannelElement *sce);
+
+#endif /* AVCODEC_AACENC_PRED_H */

libavcodec/aacenc_quantization.h

@@ -0,0 +1,260 @@
+/*
+ * 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 quantizer
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#ifndef AVCODEC_AACENC_QUANTIZATION_H
+#define AVCODEC_AACENC_QUANTIZATION_H
+
+#include "aactab.h"
+#include "aacenc.h"
+#include "aacenctab.h"
+#include "aacenc_utils.h"
+
+/**
+ * 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, float *out,
+                                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, int BT_NOISE, int BT_STEREO,
+                                const float ROUNDING)
+{
+    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;
+    int off;
+
+    if (BT_ZERO || BT_NOISE || BT_STEREO) {
+        for (i = 0; i < size; i++)
+            cost += in[i]*in[i];
+        if (bits)
+            *bits = 0;
+        if (out) {
+            for (i = 0; i < size; i += dim)
+                for (j = 0; j < dim; j++)
+                    out[i+j] = 0.0f;
+        }
+        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, aac_cb_maxval[cb], ROUNDING);
+    if (BT_UNSIGNED) {
+        off = 0;
+    } else {
+        off = aac_cb_maxval[cb];
+    }
+    for (i = 0; i < size; i += dim) {
+        const float *vec;
+        int *quants = s->qcoefs + i;
+        int curidx = 0;
+        int curbits;
+        float quantized, rd = 0.0f;
+        for (j = 0; j < dim; j++) {
+            curidx *= aac_cb_range[cb];
+            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) {
+                        quantized = CLIPPED_ESCAPE;
+                        curbits += 21;
+                    } else {
+                        int c = av_clip_uintp2(quant(t, Q, ROUNDING), 13);
+                        quantized = c*cbrtf(c)*IQ;
+                        curbits += av_log2(c)*2 - 4 + 1;
+                    }
+                } else {
+                    quantized = vec[j]*IQ;
+                }
+                di = t - quantized;
+                if (out)
+                    out[i+j] = in[i+j] >= 0 ? quantized : -quantized;
+                if (vec[j] != 0.0f)
+                    curbits++;
+                rd += di*di;
+            }
+        } else {
+            for (j = 0; j < dim; j++) {
+                quantized = vec[j]*IQ;
+                if (out)
+                    out[i+j] = quantized;
+                rd += (in[i+j] - quantized)*(in[i+j] - quantized);
+            }
+        }
+        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, ROUNDING), 13);
+                        int len = av_log2(coef);
+
+                        put_bits(pb, len - 4 + 1, (1 << (len - 4 + 1)) - 2);
+                        put_sbits(pb, len, coef);
+                    }
+                }
+            }
+        }
+    }
+
+    if (bits)
+        *bits = resbits;
+    return cost;
+}
+
+static inline float quantize_and_encode_band_cost_NONE(struct AACEncContext *s, PutBitContext *pb,
+                                                const float *in, float *quant, const float *scaled,
+                                                int size, int scale_idx, int cb,
+                                                const float lambda, const float uplim,
+                                                int *bits) {
+    av_assert0(0);
+    return 0.0f;
+}
+
+#define QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NAME, BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC, BT_NOISE, BT_STEREO, ROUNDING) \
+static float quantize_and_encode_band_cost_ ## NAME(                                         \
+                                struct AACEncContext *s,                                     \
+                                PutBitContext *pb, const float *in, float *quant,            \
+                                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, quant, scaled, size, scale_idx,                   \
+                                BT_ESC ? ESC_BT : cb, lambda, uplim, bits,                   \
+                                BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC, BT_NOISE, BT_STEREO,  \
+                                ROUNDING);                                                   \
+}
+
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ZERO,  1, 0, 0, 0, 0, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SQUAD, 0, 0, 0, 0, 0, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UQUAD, 0, 1, 0, 0, 0, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SPAIR, 0, 0, 1, 0, 0, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UPAIR, 0, 1, 1, 0, 0, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ESC,   0, 1, 1, 1, 0, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ESC_RTZ, 0, 1, 1, 1, 0, 0, ROUND_TO_ZERO)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NOISE, 0, 0, 0, 0, 1, 0, ROUND_STANDARD)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(STEREO,0, 0, 0, 0, 0, 1, ROUND_STANDARD)
+
+static float (*const quantize_and_encode_band_cost_arr[])(
+                                struct AACEncContext *s,
+                                PutBitContext *pb, const float *in, float *quant,
+                                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,
+    quantize_and_encode_band_cost_NONE,     /* CB 12 doesn't exist */
+    quantize_and_encode_band_cost_NOISE,
+    quantize_and_encode_band_cost_STEREO,
+    quantize_and_encode_band_cost_STEREO,
+};
+
+static float (*const quantize_and_encode_band_cost_rtz_arr[])(
+                                struct AACEncContext *s,
+                                PutBitContext *pb, const float *in, float *quant,
+                                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_RTZ,
+    quantize_and_encode_band_cost_NONE,     /* CB 12 doesn't exist */
+    quantize_and_encode_band_cost_NOISE,
+    quantize_and_encode_band_cost_STEREO,
+    quantize_and_encode_band_cost_STEREO,
+};
+
+#define quantize_and_encode_band_cost(                                  \
+                                s, pb, in, quant, scaled, size, scale_idx, cb, \
+                                lambda, uplim, bits, rtz)               \
+    ((rtz) ? quantize_and_encode_band_cost_rtz_arr : quantize_and_encode_band_cost_arr)[cb]( \
+                                s, pb, in, quant, scaled, size, scale_idx, cb, \
+                                lambda, uplim, bits)
+
+static inline 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, int rtz)
+{
+    return quantize_and_encode_band_cost(s, NULL, in, NULL, scaled, size, scale_idx,
+                                         cb, lambda, uplim, bits, rtz);
+}
+
+static inline 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)
+{
+    quantize_and_encode_band_cost(s, pb, in, out, NULL, size, scale_idx, cb, lambda,
+                                  INFINITY, NULL, rtz);
+}
+
+#endif /* AVCODEC_AACENC_QUANTIZATION_H */

libavcodec/aacenc_tns.c

@@ -0,0 +1,194 @@
+/*
+ * AAC encoder TNS
+ * 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 temporal noise shaping
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#include "aacenc.h"
+#include "aacenc_tns.h"
+#include "aactab.h"
+#include "aacenc_utils.h"
+#include "aacenc_quantization.h"
+
+/**
+ * Encode TNS data.
+ * Coefficient compression saves a single bit per coefficient.
+ */
+void ff_aac_encode_tns_info(AACEncContext *s, SingleChannelElement *sce)
+{
+    uint8_t u_coef;
+    const uint8_t coef_res = TNS_Q_BITS == 4;
+    int i, w, filt, coef_len, coef_compress = 0;
+    const int is8 = sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE;
+    TemporalNoiseShaping *tns = &sce->tns;
+
+    if (!sce->tns.present)
+        return;
+
+    for (i = 0; i < sce->ics.num_windows; i++) {
+        put_bits(&s->pb, 2 - is8, sce->tns.n_filt[i]);
+        if (tns->n_filt[i]) {
+            put_bits(&s->pb, 1, coef_res);
+            for (filt = 0; filt < tns->n_filt[i]; filt++) {
+                put_bits(&s->pb, 6 - 2 * is8, tns->length[i][filt]);
+                put_bits(&s->pb, 5 - 2 * is8, tns->order[i][filt]);
+                if (tns->order[i][filt]) {
+                    put_bits(&s->pb, 1, !!tns->direction[i][filt]);
+                    put_bits(&s->pb, 1, !!coef_compress);
+                    coef_len = coef_res + 3 - coef_compress;
+                    for (w = 0; w < tns->order[i][filt]; w++) {
+                        u_coef = (tns->coef_idx[i][filt][w])&(~(~0<<coef_len));
+                        put_bits(&s->pb, coef_len, u_coef);
+                    }
+                }
+            }
+        }
+    }
+}
+
+static inline void quantize_coefs(double *coef, int *idx, float *lpc, int order)
+{
+    int i;
+    uint8_t u_coef;
+    const float *quant_arr = tns_tmp2_map[TNS_Q_BITS == 4];
+    const double iqfac_p = ((1 << (TNS_Q_BITS-1)) - 0.5)/(M_PI/2.0);
+    const double iqfac_m = ((1 << (TNS_Q_BITS-1)) + 0.5)/(M_PI/2.0);
+    for (i = 0; i < order; i++) {
+        idx[i] = ceilf(asin(coef[i])*((coef[i] >= 0) ? iqfac_p : iqfac_m));
+        u_coef = (idx[i])&(~(~0<<TNS_Q_BITS));
+        lpc[i] = quant_arr[u_coef];
+    }
+}
+
+/* Apply TNS filter */
+void ff_aac_apply_tns(AACEncContext *s, SingleChannelElement *sce)
+{
+    TemporalNoiseShaping *tns = &sce->tns;
+    IndividualChannelStream *ics = &sce->ics;
+    int w, filt, m, i, top, order, bottom, start, end, size, inc;
+    const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
+    float lpc[TNS_MAX_ORDER];
+
+    for (w = 0; w < ics->num_windows; w++) {
+        bottom = ics->num_swb;
+        for (filt = 0; filt < tns->n_filt[w]; filt++) {
+            top    = bottom;
+            bottom = FFMAX(0, top - tns->length[w][filt]);
+            order  = tns->order[w][filt];
+            if (order == 0)
+                continue;
+
+            // tns_decode_coef
+            compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
+
+            start = ics->swb_offset[FFMIN(bottom, mmm)];
+            end   = ics->swb_offset[FFMIN(   top, mmm)];
+            if ((size = end - start) <= 0)
+                continue;
+            if (tns->direction[w][filt]) {
+                inc = -1;
+                start = end - 1;
+            } else {
+                inc = 1;
+            }
+            start += w * 128;
+
+            // ar filter
+            for (m = 0; m < size; m++, start += inc)
+                for (i = 1; i <= FFMIN(m, order); i++)
+                    sce->coeffs[start] += lpc[i-1]*sce->pcoeffs[start - i*inc];
+        }
+    }
+}
+
+void ff_aac_search_for_tns(AACEncContext *s, SingleChannelElement *sce)
+{
+    TemporalNoiseShaping *tns = &sce->tns;
+    int w, w2, g, count = 0;
+    const int mmm = FFMIN(sce->ics.tns_max_bands, sce->ics.max_sfb);
+    const int is8 = sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE;
+    const int order = is8 ? 7 : s->profile == FF_PROFILE_AAC_LOW ? 12 : TNS_MAX_ORDER;
+
+    int sfb_start = av_clip(tns_min_sfb[is8][s->samplerate_index], 0, mmm);
+    int sfb_end   = av_clip(sce->ics.num_swb, 0, mmm);
+
+    for (w = 0; w < sce->ics.num_windows; w++) {
+        float e_ratio = 0.0f, threshold = 0.0f, spread = 0.0f, en[2] = {0.0, 0.0f};
+        double gain = 0.0f, coefs[MAX_LPC_ORDER] = {0};
+        int coef_start = w*sce->ics.num_swb + sce->ics.swb_offset[sfb_start];
+        int coef_len = sce->ics.swb_offset[sfb_end] - sce->ics.swb_offset[sfb_start];
+
+        for (g = 0;  g < sce->ics.num_swb; g++) {
+            if (w*16+g < sfb_start || w*16+g > sfb_end)
+                continue;
+            for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
+                FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
+                if ((w+w2)*16+g > sfb_start + ((sfb_end - sfb_start)/2))
+                    en[1] += band->energy;
+                else
+                    en[0] += band->energy;
+                threshold += band->threshold;
+                spread += band->spread;
+            }
+        }
+
+        if (coef_len <= 0 || (sfb_end - sfb_start) <= 0)
+            continue;
+        else
+            e_ratio = en[0]/en[1];
+
+        /* LPC */
+        gain = ff_lpc_calc_ref_coefs_f(&s->lpc, &sce->coeffs[coef_start],
+                                       coef_len, order, coefs);
+
+        if (gain > TNS_GAIN_THRESHOLD_LOW && gain < TNS_GAIN_THRESHOLD_HIGH &&
+            (en[0]+en[1]) > TNS_GAIN_THRESHOLD_LOW*threshold &&
+            spread < TNS_SPREAD_THRESHOLD && order) {
+            if (is8 || order < 2 || (e_ratio > TNS_E_RATIO_LOW && e_ratio < TNS_E_RATIO_HIGH)) {
+                tns->n_filt[w] = 1;
+                for (g = 0; g < tns->n_filt[w]; g++) {
+                    tns->length[w][g] = sfb_end - sfb_start;
+                    tns->direction[w][g] = en[0] < en[1];
+                    tns->order[w][g] = order;
+                    quantize_coefs(coefs, tns->coef_idx[w][g], tns->coef[w][g],
+                                   order);
+                }
+            } else {  /* 2 filters due to energy disbalance */
+                tns->n_filt[w] = 2;
+                for (g = 0; g < tns->n_filt[w]; g++) {
+                    tns->direction[w][g] = en[g] < en[!g];
+                    tns->order[w][g] = !g ? order/2 : order - tns->order[w][g-1];
+                    tns->length[w][g] = !g ? (sfb_end - sfb_start)/2 : \
+                                    (sfb_end - sfb_start) - tns->length[w][g-1];
+                    quantize_coefs(&coefs[!g ? 0 : order - tns->order[w][g-1]],
+                                   tns->coef_idx[w][g], tns->coef[w][g],
+                                   tns->order[w][g]);
+                }
+            }
+            count++;
+        }
+    }
+
+    sce->tns.present = !!count;
+}

libavcodec/aacenc_tns.h

@@ -0,0 +1,52 @@
+/*
+ * AAC encoder TNS
+ * 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 temporal noise shaping
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#ifndef AVCODEC_AACENC_TNS_H
+#define AVCODEC_AACENC_TNS_H
+
+#include "aacenc.h"
+
+/* Could be set to 3 to save an additional bit at the cost of little quality */
+#define TNS_Q_BITS 4
+
+/* TNS will only be used if the LPC gain is within these margins */
+#define TNS_GAIN_THRESHOLD_LOW  1.395f
+#define TNS_GAIN_THRESHOLD_HIGH 11.19f
+
+/* If the energy ratio between the low SFBs vs the high SFBs is not between
+ * those two values, use 2 filters instead */
+#define TNS_E_RATIO_LOW  0.77
+#define TNS_E_RATIO_HIGH 1.23
+
+/* Do not use TNS if the psy band spread is below this value */
+#define TNS_SPREAD_THRESHOLD 37.081512f
+
+void ff_aac_encode_tns_info(AACEncContext *s, SingleChannelElement *sce);
+void ff_aac_apply_tns(AACEncContext *s, SingleChannelElement *sce);
+void ff_aac_search_for_tns(AACEncContext *s, SingleChannelElement *sce);
+
+#endif /* AVCODEC_AACENC_TNS_H */

libavcodec/aacenc_utils.h

@@ -0,0 +1,143 @@
+/*
+ * AAC encoder utilities
+ * 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 utilities
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#ifndef AVCODEC_AACENC_UTILS_H
+#define AVCODEC_AACENC_UTILS_H
+
+#include "aac.h"
+#include "aac_tablegen_decl.h"
+#include "aacenctab.h"
+
+#define ROUND_STANDARD 0.4054f
+#define ROUND_TO_ZERO 0.1054f
+#define C_QUANT 0.4054f
+
+static inline void abs_pow34_v(float *out, const float *in, const int size)
+{
+    int i;
+    for (i = 0; i < size; i++) {
+        float a = fabsf(in[i]);
+        out[i] = sqrtf(a * sqrtf(a));
+    }
+}
+
+/**
+ * Quantize one coefficient.
+ * @return absolute value of the quantized coefficient
+ * @see 3GPP TS26.403 5.6.2 "Scalefactor determination"
+ */
+static inline int quant(float coef, const float Q, const float rounding)
+{
+    float a = coef * Q;
+    return sqrtf(a * sqrtf(a)) + rounding;
+}
+
+static inline void quantize_bands(int *out, const float *in, const float *scaled,
+                                  int size, float Q34, int is_signed, int maxval,
+                                  const float rounding)
+{
+    int i;
+    double qc;
+    for (i = 0; i < size; i++) {
+        qc = scaled[i] * Q34;
+        out[i] = (int)FFMIN(qc + rounding, (double)maxval);
+        if (is_signed && in[i] < 0.0f) {
+            out[i] = -out[i];
+        }
+    }
+}
+
+static inline 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 inline 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 + C_QUANT;
+    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;
+}
+
+/** Return the minimum scalefactor where the quantized coef does not clip. */
+static 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 inline uint8_t coef2maxsf(float coef)
+{
+    return av_clip_uint8(log2f(coef)*4 +  6 + SCALE_ONE_POS - SCALE_DIV_512);
+}
+
+/*
+ * Returns the closest possible index to an array of float values, given a value.
+ */
+static inline int quant_array_idx(const float val, const float *arr, const int num)
+{
+    int i, index = 0;
+    float quant_min_err = INFINITY;
+    for (i = 0; i < num; i++) {
+        float error = (val - arr[i])*(val - arr[i]);
+        if (error < quant_min_err) {
+            quant_min_err = error;
+            index = i;
+        }
+    }
+    return index;
+}
+
+#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__); \
+    }
+
+
+#endif /* AVCODEC_AACENC_UTILS_H */

libavcodec/aacenctab.c

@@ -0,0 +1,108 @@
+/*
+ * AAC encoder data
+ * Copyright (c) 2015 Rostislav Pehlivanov ( atomnuker gmail com )
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "aacenctab.h"
+
+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_64[] = {
+    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_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
+};
+
+const uint8_t *ff_aac_swb_size_128[] = {
+    swb_size_128_96, swb_size_128_96, swb_size_128_64,
+    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
+};
+
+const uint8_t *ff_aac_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
+};
+
+const int ff_aac_swb_size_128_len  = FF_ARRAY_ELEMS(ff_aac_swb_size_128);
+const int ff_aac_swb_size_1024_len = FF_ARRAY_ELEMS(ff_aac_swb_size_1024);

libavcodec/aacenctab.h

@@ -0,0 +1,113 @@
+/*
+ * AAC encoder data
+ * Copyright (c) 2015 Rostislav Pehlivanov ( atomnuker gmail com )
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC encoder data
+ * @author Rostislav Pehlivanov ( atomnuker gmail com )
+ */
+
+#ifndef AVCODEC_AACENCTAB_H
+#define AVCODEC_AACENCTAB_H
+
+#include "aac.h"
+
+/** Total number of usable codebooks **/
+#define CB_TOT 12
+
+/** Total number of codebooks, including special ones **/
+#define CB_TOT_ALL 15
+
+#define AAC_MAX_CHANNELS 6
+
+extern const uint8_t *ff_aac_swb_size_1024[];
+extern const int      ff_aac_swb_size_1024_len;
+extern const uint8_t *ff_aac_swb_size_128[];
+extern const int      ff_aac_swb_size_128_len;
+
+/** 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 },
+};
+
+/* 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
+};
+
+/** 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
+};
+
+/** 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
+};
+
+/* TNS starting SFBs for long and short windows */
+static const uint8_t tns_min_sfb_short[16] = {
+    2, 2, 2, 3, 3, 4, 6, 6, 8, 10, 10, 12, 12, 12, 12, 12
+};
+
+static const uint8_t tns_min_sfb_long[16] = {
+    12, 13, 15, 16, 17, 20, 25, 26, 24, 28, 30, 31, 31, 31, 31, 31
+};
+
+static const uint8_t * const tns_min_sfb[2] = {
+    tns_min_sfb_long, tns_min_sfb_short
+};
+
+static const uint8_t * const run_value_bits[2] = {
+    run_value_bits_long, run_value_bits_short
+};
+
+/** Map to convert values from BandCodingPath index to a codebook index **/
+static const uint8_t aac_cb_out_map[CB_TOT_ALL]  = {0,1,2,3,4,5,6,7,8,9,10,11,13,14,15};
+/** Inverse map to convert from codebooks to BandCodingPath indices **/
+static const uint8_t aac_cb_in_map[CB_TOT_ALL+1] = {0,1,2,3,4,5,6,7,8,9,10,11,0,12,13,14};
+
+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};
+
+#endif /* AVCODEC_AACENCTAB_H */

libavcodec/aacps.c

@@ -17,16 +17,23 @@
  * 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
+ *
+ * Note: Rounding-to-nearest used unless otherwise stated
+ *
  */
 
 #include <stdint.h>
 #include "libavutil/common.h"
-#include "libavutil/internal.h"
 #include "libavutil/mathematics.h"
 #include "avcodec.h"
 #include "get_bits.h"
 #include "aacps.h"
+#if USE_FIXED
+#include "aacps_fixed_tablegen.h"
+#else
+#include "libavutil/internal.h"
 #include "aacps_tablegen.h"
+#endif /* USE_FIXED */
 #include "aacpsdata.c"
 
 #define PS_BASELINE 0  ///< Operate in Baseline PS mode
@@ -148,7 +155,7 @@ static void ipdopd_reset(int8_t *ipd_hist, int8_t *opd_hist)
     }
 }
 
-int ff_ps_read_data(AVCodecContext *avctx, GetBitContext *gb_host, PSContext *ps, int bits_left)
+int AAC_RENAME(ff_ps_read_data)(AVCodecContext *avctx, GetBitContext *gb_host, PSContext *ps, int bits_left)
 {
     int e;
     int bit_count_start = get_bits_count(gb_host);
@@ -302,35 +309,41 @@ err:
 
 /** Split one subband into 2 subsubbands with a symmetric real filter.
  * The filter must have its non-center even coefficients equal to zero. */
-static void hybrid2_re(float (*in)[2], float (*out)[32][2], const float filter[8], int len, int reverse)
+static void hybrid2_re(INTFLOAT (*in)[2], INTFLOAT (*out)[32][2], const INTFLOAT filter[8], int len, int reverse)
 {
     int i, j;
     for (i = 0; i < len; i++, in++) {
-        float re_in = filter[6] * in[6][0];          //real inphase
-        float re_op = 0.0f;                          //real out of phase
-        float im_in = filter[6] * in[6][1];          //imag inphase
-        float im_op = 0.0f;                          //imag out of phase
+        INT64FLOAT re_in = AAC_MUL31(filter[6], in[6][0]); //real inphase
+        INT64FLOAT re_op = 0.0f;                          //real out of phase
+        INT64FLOAT im_in = AAC_MUL31(filter[6], in[6][1]); //imag inphase
+        INT64FLOAT im_op = 0.0f;                          //imag out of phase
         for (j = 0; j < 6; j += 2) {
-            re_op += filter[j+1] * (in[j+1][0] + in[12-j-1][0]);
-            im_op += filter[j+1] * (in[j+1][1] + in[12-j-1][1]);
+            re_op += (INT64FLOAT)filter[j+1] * (in[j+1][0] + in[12-j-1][0]);
+            im_op += (INT64FLOAT)filter[j+1] * (in[j+1][1] + in[12-j-1][1]);
         }
-        out[ reverse][i][0] = re_in + re_op;
-        out[ reverse][i][1] = im_in + im_op;
-        out[!reverse][i][0] = re_in - re_op;
-        out[!reverse][i][1] = im_in - im_op;
+
+#if USE_FIXED
+        re_op = (re_op + 0x40000000) >> 31;
+        im_op = (im_op + 0x40000000) >> 31;
+#endif /* USE_FIXED */
+
+        out[ reverse][i][0] = (INTFLOAT)(re_in + re_op);
+        out[ reverse][i][1] = (INTFLOAT)(im_in + im_op);
+        out[!reverse][i][0] = (INTFLOAT)(re_in - re_op);
+        out[!reverse][i][1] = (INTFLOAT)(im_in - im_op);
     }
 }
 
 /** Split one subband into 6 subsubbands with a complex filter */
-static void hybrid6_cx(PSDSPContext *dsp, float (*in)[2], float (*out)[32][2],
-                       TABLE_CONST float (*filter)[8][2], int len)
+static void hybrid6_cx(PSDSPContext *dsp, INTFLOAT (*in)[2], INTFLOAT (*out)[32][2],
+                       TABLE_CONST INTFLOAT (*filter)[8][2], int len)
 {
     int i;
     int N = 8;
-    LOCAL_ALIGNED_16(float, temp, [8], [2]);
+    LOCAL_ALIGNED_16(INTFLOAT, temp, [8], [2]);
 
     for (i = 0; i < len; i++, in++) {
-        dsp->hybrid_analysis(temp, in, (const float (*)[8][2]) filter, 1, N);
+        dsp->hybrid_analysis(temp, in, (const INTFLOAT (*)[8][2]) filter, 1, N);
         out[0][i][0] = temp[6][0];
         out[0][i][1] = temp[6][1];
         out[1][i][0] = temp[7][0];
@@ -347,18 +360,18 @@ static void hybrid6_cx(PSDSPContext *dsp, float (*in)[2], float (*out)[32][2],
 }
 
 static void hybrid4_8_12_cx(PSDSPContext *dsp,
-                            float (*in)[2], float (*out)[32][2],
-                            TABLE_CONST float (*filter)[8][2], int N, int len)
+                            INTFLOAT (*in)[2], INTFLOAT (*out)[32][2],
+                            TABLE_CONST INTFLOAT (*filter)[8][2], int N, int len)
 {
     int i;
 
     for (i = 0; i < len; i++, in++) {
-        dsp->hybrid_analysis(out[0] + i, in, (const float (*)[8][2]) filter, 32, N);
+        dsp->hybrid_analysis(out[0] + i, in, (const INTFLOAT (*)[8][2]) filter, 32, N);
     }
 }
 
-static void hybrid_analysis(PSDSPContext *dsp, float out[91][32][2],
-                            float in[5][44][2], float L[2][38][64],
+static void hybrid_analysis(PSDSPContext *dsp, INTFLOAT out[91][32][2],
+                            INTFLOAT in[5][44][2], INTFLOAT L[2][38][64],
                             int is34, int len)
 {
     int i, j;
@@ -387,8 +400,8 @@ static void hybrid_analysis(PSDSPContext *dsp, float out[91][32][2],
     }
 }
 
-static void hybrid_synthesis(PSDSPContext *dsp, float out[2][38][64],
-                             float in[91][32][2], int is34, int len)
+static void hybrid_synthesis(PSDSPContext *dsp, INTFLOAT out[2][38][64],
+                             INTFLOAT in[91][32][2], int is34, int len)
 {
     int i, n;
     if (is34) {
@@ -429,7 +442,7 @@ static void hybrid_synthesis(PSDSPContext *dsp, float out[2][38][64],
 }
 
 /// All-pass filter decay slope
-#define DECAY_SLOPE      0.05f
+#define DECAY_SLOPE      Q30(0.05f)
 /// Number of frequency bands that can be addressed by the parameter index, b(k)
 static const int   NR_PAR_BANDS[]      = { 20, 34 };
 static const int   NR_IPDOPD_BANDS[]   = { 11, 17 };
@@ -483,28 +496,43 @@ static void map_idx_34_to_20(int8_t *par_mapped, const int8_t *par, int full)
     }
 }
 
-static void map_val_34_to_20(float par[PS_MAX_NR_IIDICC])
+static void map_val_34_to_20(INTFLOAT par[PS_MAX_NR_IIDICC])
 {
+#if USE_FIXED
+    par[ 0] = (int)(((int64_t)(par[ 0] + (unsigned)(par[ 1]>>1)) * 1431655765 + \
+                      0x40000000) >> 31);
+    par[ 1] = (int)(((int64_t)((par[ 1]>>1) + (unsigned)par[ 2]) * 1431655765 + \
+                      0x40000000) >> 31);
+    par[ 2] = (int)(((int64_t)(par[ 3] + (unsigned)(par[ 4]>>1)) * 1431655765 + \
+                      0x40000000) >> 31);
+    par[ 3] = (int)(((int64_t)((par[ 4]>>1) + (unsigned)par[ 5]) * 1431655765 + \
+                      0x40000000) >> 31);
+#else
     par[ 0] = (2*par[ 0] +   par[ 1]) * 0.33333333f;
     par[ 1] = (  par[ 1] + 2*par[ 2]) * 0.33333333f;
     par[ 2] = (2*par[ 3] +   par[ 4]) * 0.33333333f;
     par[ 3] = (  par[ 4] + 2*par[ 5]) * 0.33333333f;
-    par[ 4] = (  par[ 6] +   par[ 7]) * 0.5f;
-    par[ 5] = (  par[ 8] +   par[ 9]) * 0.5f;
+#endif /* USE_FIXED */
+    par[ 4] = AAC_HALF_SUM(par[ 6], par[ 7]);
+    par[ 5] = AAC_HALF_SUM(par[ 8], par[ 9]);
     par[ 6] =    par[10];
     par[ 7] =    par[11];
-    par[ 8] = (  par[12] +   par[13]) * 0.5f;
-    par[ 9] = (  par[14] +   par[15]) * 0.5f;
+    par[ 8] = AAC_HALF_SUM(par[12], par[13]);
+    par[ 9] = AAC_HALF_SUM(par[14], par[15]);
     par[10] =    par[16];
     par[11] =    par[17];
     par[12] =    par[18];
     par[13] =    par[19];
-    par[14] = (  par[20] +   par[21]) * 0.5f;
-    par[15] = (  par[22] +   par[23]) * 0.5f;
-    par[16] = (  par[24] +   par[25]) * 0.5f;
-    par[17] = (  par[26] +   par[27]) * 0.5f;
+    par[14] = AAC_HALF_SUM(par[20], par[21]);
+    par[15] = AAC_HALF_SUM(par[22], par[23]);
+    par[16] = AAC_HALF_SUM(par[24], par[25]);
+    par[17] = AAC_HALF_SUM(par[26], par[27]);
+#if USE_FIXED
+    par[18] = (((par[28]+2)>>2) + ((par[29]+2)>>2) + ((par[30]+2)>>2) + ((par[31]+2)>>2));
+#else
     par[18] = (  par[28] +   par[29] +   par[30] +   par[31]) * 0.25f;
-    par[19] = (  par[32] +   par[33]) * 0.5f;
+#endif /* USE_FIXED */
+    par[19] = AAC_HALF_SUM(par[32], par[33]);
 }
 
 static void map_idx_10_to_34(int8_t *par_mapped, const int8_t *par, int full)
@@ -589,7 +617,7 @@ static void map_idx_20_to_34(int8_t *par_mapped, const int8_t *par, int full)
     par_mapped[ 0] =  par[ 0];
 }
 
-static void map_val_20_to_34(float par[PS_MAX_NR_IIDICC])
+static void map_val_20_to_34(INTFLOAT par[PS_MAX_NR_IIDICC])
 {
     par[33] =  par[19];
     par[32] =  par[19];
@@ -620,27 +648,29 @@ static void map_val_20_to_34(float par[PS_MAX_NR_IIDICC])
     par[ 7] =  par[ 4];
     par[ 6] =  par[ 4];
     par[ 5] =  par[ 3];
-    par[ 4] = (par[ 2] + par[ 3]) * 0.5f;
+    par[ 4] = AAC_HALF_SUM(par[ 2], par[ 3]);
     par[ 3] =  par[ 2];
     par[ 2] =  par[ 1];
-    par[ 1] = (par[ 0] + par[ 1]) * 0.5f;
+    par[ 1] = AAC_HALF_SUM(par[ 0], par[ 1]);
 }
 
-static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[32][2], int is34)
+static void decorrelation(PSContext *ps, INTFLOAT (*out)[32][2], const INTFLOAT (*s)[32][2], int is34)
 {
-    LOCAL_ALIGNED_16(float, power, [34], [PS_QMF_TIME_SLOTS]);
-    LOCAL_ALIGNED_16(float, transient_gain, [34], [PS_QMF_TIME_SLOTS]);
-    float *peak_decay_nrg = ps->peak_decay_nrg;
-    float *power_smooth = ps->power_smooth;
-    float *peak_decay_diff_smooth = ps->peak_decay_diff_smooth;
-    float (*delay)[PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2] = ps->delay;
-    float (*ap_delay)[PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2] = ps->ap_delay;
-    const int8_t *k_to_i = is34 ? k_to_i_34 : k_to_i_20;
-    const float peak_decay_factor = 0.76592833836465f;
+    LOCAL_ALIGNED_16(INTFLOAT, power, [34], [PS_QMF_TIME_SLOTS]);
+    LOCAL_ALIGNED_16(INTFLOAT, transient_gain, [34], [PS_QMF_TIME_SLOTS]);
+    INTFLOAT *peak_decay_nrg = ps->peak_decay_nrg;
+    INTFLOAT *power_smooth = ps->power_smooth;
+    INTFLOAT *peak_decay_diff_smooth = ps->peak_decay_diff_smooth;
+    INTFLOAT (*delay)[PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2] = ps->delay;
+    INTFLOAT (*ap_delay)[PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2] = ps->ap_delay;
+#if !USE_FIXED
     const float transient_impact  = 1.5f;
     const float a_smooth          = 0.25f; ///< Smoothing coefficient
+#endif /* USE_FIXED */
+    const int8_t *k_to_i = is34 ? k_to_i_34 : k_to_i_20;
     int i, k, m, n;
     int n0 = 0, nL = 32;
+    const INTFLOAT peak_decay_factor = Q31(0.76592833836465f);
 
     memset(power, 0, 34 * sizeof(*power));
 
@@ -658,6 +688,24 @@ static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[3
     }
 
     //Transient detection
+#if USE_FIXED
+    for (i = 0; i < NR_PAR_BANDS[is34]; i++) {
+        for (n = n0; n < nL; n++) {
+            int decayed_peak;
+            decayed_peak = (int)(((int64_t)peak_decay_factor * \
+                                           peak_decay_nrg[i] + 0x40000000) >> 31);
+            peak_decay_nrg[i] = FFMAX(decayed_peak, power[i][n]);
+            power_smooth[i] += (power[i][n] + 2LL - power_smooth[i]) >> 2;
+            peak_decay_diff_smooth[i] += (peak_decay_nrg[i] + 2LL - power[i][n] - \
+                                          peak_decay_diff_smooth[i]) >> 2;
+
+            if (peak_decay_diff_smooth[i]) {
+                transient_gain[i][n] = FFMIN(power_smooth[i]*43691LL / peak_decay_diff_smooth[i], 1<<16);
+            } else
+                transient_gain[i][n] = 1 << 16;
+        }
+    }
+#else
     for (i = 0; i < NR_PAR_BANDS[is34]; i++) {
         for (n = n0; n < nL; n++) {
             float decayed_peak = peak_decay_factor * peak_decay_nrg[i];
@@ -671,6 +719,7 @@ static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[3
         }
     }
 
+#endif /* USE_FIXED */
     //Decorrelation and transient reduction
     //                         PS_AP_LINKS - 1
     //                               -----
@@ -681,8 +730,22 @@ static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[3
     //d[k][z] (out) = transient_gain_mapped[k][z] * H[k][z] * s[k][z]
     for (k = 0; k < NR_ALLPASS_BANDS[is34]; k++) {
         int b = k_to_i[k];
+#if USE_FIXED
+        int g_decay_slope;
+
+        if (k - DECAY_CUTOFF[is34] <= 0) {
+          g_decay_slope = 1 << 30;
+        }
+        else if (k - DECAY_CUTOFF[is34] >= 20) {
+          g_decay_slope = 0;
+        }
+        else {
+          g_decay_slope = (1 << 30) - DECAY_SLOPE * (k - DECAY_CUTOFF[is34]);
+        }
+#else
         float g_decay_slope = 1.f - DECAY_SLOPE * (k - DECAY_CUTOFF[is34]);
         g_decay_slope = av_clipf(g_decay_slope, 0.f, 1.f);
+#endif /* USE_FIXED */
         memcpy(delay[k], delay[k]+nL, PS_MAX_DELAY*sizeof(delay[k][0]));
         memcpy(delay[k]+PS_MAX_DELAY, s[k], numQMFSlots*sizeof(delay[k][0]));
         for (m = 0; m < PS_AP_LINKS; m++) {
@@ -690,7 +753,7 @@ static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[3
         }
         ps->dsp.decorrelate(out[k], delay[k] + PS_MAX_DELAY - 2, ap_delay[k],
                             phi_fract[is34][k],
-                            (const float (*)[2]) Q_fract_allpass[is34][k],
+                            (const INTFLOAT (*)[2]) Q_fract_allpass[is34][k],
                             transient_gain[b], g_decay_slope, nL - n0);
     }
     for (; k < SHORT_DELAY_BAND[is34]; k++) {
@@ -749,14 +812,14 @@ static void remap20(int8_t (**p_par_mapped)[PS_MAX_NR_IIDICC],
     }
 }
 
-static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2], int is34)
+static void stereo_processing(PSContext *ps, INTFLOAT (*l)[32][2], INTFLOAT (*r)[32][2], int is34)
 {
     int e, b, k;
 
-    float (*H11)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H11;
-    float (*H12)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H12;
-    float (*H21)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H21;
-    float (*H22)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H22;
+    INTFLOAT (*H11)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H11;
+    INTFLOAT (*H12)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H12;
+    INTFLOAT (*H21)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H21;
+    INTFLOAT (*H22)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H22;
     int8_t *opd_hist = ps->opd_hist;
     int8_t *ipd_hist = ps->ipd_hist;
     int8_t iid_mapped_buf[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC];
@@ -768,7 +831,7 @@ static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2
     int8_t (*ipd_mapped)[PS_MAX_NR_IIDICC] = ipd_mapped_buf;
     int8_t (*opd_mapped)[PS_MAX_NR_IIDICC] = opd_mapped_buf;
     const int8_t *k_to_i = is34 ? k_to_i_34 : k_to_i_20;
-    TABLE_CONST float (*H_LUT)[8][4] = (PS_BASELINE || ps->icc_mode < 3) ? HA : HB;
+    TABLE_CONST INTFLOAT (*H_LUT)[8][4] = (PS_BASELINE || ps->icc_mode < 3) ? HA : HB;
 
     //Remapping
     if (ps->num_env_old) {
@@ -823,7 +886,7 @@ static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2
     //Mixing
     for (e = 0; e < ps->num_env; e++) {
         for (b = 0; b < NR_PAR_BANDS[is34]; b++) {
-            float h11, h12, h21, h22;
+            INTFLOAT h11, h12, h21, h22;
             h11 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][0];
             h12 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][1];
             h21 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][2];
@@ -832,27 +895,27 @@ static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2
             if (!PS_BASELINE && ps->enable_ipdopd && b < NR_IPDOPD_BANDS[is34]) {
                 //The spec say says to only run this smoother when enable_ipdopd
                 //is set but the reference decoder appears to run it constantly
-                float h11i, h12i, h21i, h22i;
-                float ipd_adj_re, ipd_adj_im;
+                INTFLOAT h11i, h12i, h21i, h22i;
+                INTFLOAT ipd_adj_re, ipd_adj_im;
                 int opd_idx = opd_hist[b] * 8 + opd_mapped[e][b];
                 int ipd_idx = ipd_hist[b] * 8 + ipd_mapped[e][b];
-                float opd_re = pd_re_smooth[opd_idx];
-                float opd_im = pd_im_smooth[opd_idx];
-                float ipd_re = pd_re_smooth[ipd_idx];
-                float ipd_im = pd_im_smooth[ipd_idx];
+                INTFLOAT opd_re = pd_re_smooth[opd_idx];
+                INTFLOAT opd_im = pd_im_smooth[opd_idx];
+                INTFLOAT ipd_re = pd_re_smooth[ipd_idx];
+                INTFLOAT ipd_im = pd_im_smooth[ipd_idx];
                 opd_hist[b] = opd_idx & 0x3F;
                 ipd_hist[b] = ipd_idx & 0x3F;
 
-                ipd_adj_re = opd_re*ipd_re + opd_im*ipd_im;
-                ipd_adj_im = opd_im*ipd_re - opd_re*ipd_im;
-                h11i = h11 * opd_im;
-                h11  = h11 * opd_re;
-                h12i = h12 * ipd_adj_im;
-                h12  = h12 * ipd_adj_re;
-                h21i = h21 * opd_im;
-                h21  = h21 * opd_re;
-                h22i = h22 * ipd_adj_im;
-                h22  = h22 * ipd_adj_re;
+                ipd_adj_re = AAC_MADD30(opd_re, ipd_re, opd_im, ipd_im);
+                ipd_adj_im = AAC_MSUB30(opd_im, ipd_re, opd_re, ipd_im);
+                h11i = AAC_MUL30(h11,  opd_im);
+                h11  = AAC_MUL30(h11,  opd_re);
+                h12i = AAC_MUL30(h12,  ipd_adj_im);
+                h12  = AAC_MUL30(h12,  ipd_adj_re);
+                h21i = AAC_MUL30(h21,  opd_im);
+                h21  = AAC_MUL30(h21,  opd_re);
+                h22i = AAC_MUL30(h22,  ipd_adj_im);
+                h22  = AAC_MUL30(h22,  ipd_adj_re);
                 H11[1][e+1][b] = h11i;
                 H12[1][e+1][b] = h12i;
                 H21[1][e+1][b] = h21i;
@@ -864,11 +927,14 @@ static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2
             H22[0][e+1][b] = h22;
         }
         for (k = 0; k < NR_BANDS[is34]; k++) {
-            float h[2][4];
-            float h_step[2][4];
+            LOCAL_ALIGNED_16(INTFLOAT, h, [2], [4]);
+            LOCAL_ALIGNED_16(INTFLOAT, h_step, [2], [4]);
             int start = ps->border_position[e];
             int stop  = ps->border_position[e+1];
-            float width = 1.f / (stop - start);
+            INTFLOAT width = Q30(1.f) / ((stop - start) ? (stop - start) : 1);
+#if USE_FIXED
+            width = FFMIN(2U*width, INT_MAX);
+#endif
             b = k_to_i[k];
             h[0][0] = H11[0][e][b];
             h[0][1] = H12[0][e][b];
@@ -889,27 +955,27 @@ static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2
             }
             }
             //Interpolation
-            h_step[0][0] = (H11[0][e+1][b] - h[0][0]) * width;
-            h_step[0][1] = (H12[0][e+1][b] - h[0][1]) * width;
-            h_step[0][2] = (H21[0][e+1][b] - h[0][2]) * width;
-            h_step[0][3] = (H22[0][e+1][b] - h[0][3]) * width;
+            h_step[0][0] = AAC_MSUB31_V3(H11[0][e+1][b], h[0][0], width);
+            h_step[0][1] = AAC_MSUB31_V3(H12[0][e+1][b], h[0][1], width);
+            h_step[0][2] = AAC_MSUB31_V3(H21[0][e+1][b], h[0][2], width);
+            h_step[0][3] = AAC_MSUB31_V3(H22[0][e+1][b], h[0][3], width);
             if (!PS_BASELINE && ps->enable_ipdopd) {
-                h_step[1][0] = (H11[1][e+1][b] - h[1][0]) * width;
-                h_step[1][1] = (H12[1][e+1][b] - h[1][1]) * width;
-                h_step[1][2] = (H21[1][e+1][b] - h[1][2]) * width;
-                h_step[1][3] = (H22[1][e+1][b] - h[1][3]) * width;
+                h_step[1][0] = AAC_MSUB31_V3(H11[1][e+1][b], h[1][0], width);
+                h_step[1][1] = AAC_MSUB31_V3(H12[1][e+1][b], h[1][1], width);
+                h_step[1][2] = AAC_MSUB31_V3(H21[1][e+1][b], h[1][2], width);
+                h_step[1][3] = AAC_MSUB31_V3(H22[1][e+1][b], h[1][3], width);
             }
             ps->dsp.stereo_interpolate[!PS_BASELINE && ps->enable_ipdopd](
-                l[k] + start + 1, r[k] + start + 1,
+                l[k] + 1 + start, r[k] + 1 + start,
                 h, h_step, stop - start);
         }
     }
 }
 
-int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float R[2][38][64], int top)
+int AAC_RENAME(ff_ps_apply)(AVCodecContext *avctx, PSContext *ps, INTFLOAT L[2][38][64], INTFLOAT R[2][38][64], int top)
 {
-    float (*Lbuf)[32][2] = ps->Lbuf;
-    float (*Rbuf)[32][2] = ps->Rbuf;
+    INTFLOAT (*Lbuf)[32][2] = ps->Lbuf;
+    INTFLOAT (*Rbuf)[32][2] = ps->Rbuf;
     const int len = 32;
     int is34 = ps->is34bands;
 
@@ -919,7 +985,7 @@ int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float
         memset(ps->ap_delay + top, 0, (NR_ALLPASS_BANDS[is34] - top)*sizeof(ps->ap_delay[0]));
 
     hybrid_analysis(&ps->dsp, Lbuf, ps->in_buf, L, is34, len);
-    decorrelation(ps, Rbuf, (const float (*)[32][2]) Lbuf, is34);
+    decorrelation(ps, Rbuf, (const INTFLOAT (*)[32][2]) Lbuf, is34);
     stereo_processing(ps, Lbuf, Rbuf, is34);
     hybrid_synthesis(&ps->dsp, L, Lbuf, is34, len);
     hybrid_synthesis(&ps->dsp, R, Rbuf, is34, len);
@@ -936,7 +1002,7 @@ int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float
 #define PS_VLC_ROW(name) \
     { name ## _codes, name ## _bits, sizeof(name ## _codes), sizeof(name ## _codes[0]) }
 
-av_cold void ff_ps_init(void) {
+av_cold void AAC_RENAME(ff_ps_init)(void) {
     // Syntax initialization
     static const struct {
         const void *ps_codes, *ps_bits;
@@ -968,7 +1034,7 @@ av_cold void ff_ps_init(void) {
     ps_tableinit();
 }
 
-av_cold void ff_ps_ctx_init(PSContext *ps)
+av_cold void AAC_RENAME(ff_ps_ctx_init)(PSContext *ps)
 {
-    ff_psdsp_init(&ps->dsp);
+    AAC_RENAME(ff_psdsp_init)(&ps->dsp);
 }

libavcodec/aacps.h

@@ -61,26 +61,26 @@ typedef struct PSContext {
     int    is34bands;
     int    is34bands_old;
 
-    DECLARE_ALIGNED(16, float, in_buf)[5][44][2];
-    DECLARE_ALIGNED(16, float, delay)[PS_MAX_SSB][PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2];
-    DECLARE_ALIGNED(16, float, ap_delay)[PS_MAX_AP_BANDS][PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2];
-    DECLARE_ALIGNED(16, float, peak_decay_nrg)[34];
-    DECLARE_ALIGNED(16, float, power_smooth)[34];
-    DECLARE_ALIGNED(16, float, peak_decay_diff_smooth)[34];
-    DECLARE_ALIGNED(16, float, H11)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
-    DECLARE_ALIGNED(16, float, H12)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
-    DECLARE_ALIGNED(16, float, H21)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
-    DECLARE_ALIGNED(16, float, H22)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
-    DECLARE_ALIGNED(16, float, Lbuf)[91][32][2];
-    DECLARE_ALIGNED(16, float, Rbuf)[91][32][2];
+    DECLARE_ALIGNED(16, INTFLOAT, in_buf)[5][44][2];
+    DECLARE_ALIGNED(16, INTFLOAT, delay)[PS_MAX_SSB][PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2];
+    DECLARE_ALIGNED(16, INTFLOAT, ap_delay)[PS_MAX_AP_BANDS][PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2];
+    DECLARE_ALIGNED(16, INTFLOAT, peak_decay_nrg)[34];
+    DECLARE_ALIGNED(16, INTFLOAT, power_smooth)[34];
+    DECLARE_ALIGNED(16, INTFLOAT, peak_decay_diff_smooth)[34];
+    DECLARE_ALIGNED(16, INTFLOAT, H11)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    DECLARE_ALIGNED(16, INTFLOAT, H12)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    DECLARE_ALIGNED(16, INTFLOAT, H21)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    DECLARE_ALIGNED(16, INTFLOAT, H22)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    DECLARE_ALIGNED(16, INTFLOAT, Lbuf)[91][32][2];
+    DECLARE_ALIGNED(16, INTFLOAT, Rbuf)[91][32][2];
     int8_t opd_hist[PS_MAX_NR_IIDICC];
     int8_t ipd_hist[PS_MAX_NR_IIDICC];
     PSDSPContext dsp;
 } PSContext;
 
-void ff_ps_init(void);
-void ff_ps_ctx_init(PSContext *ps);
-int ff_ps_read_data(AVCodecContext *avctx, GetBitContext *gb, PSContext *ps, int bits_left);
-int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float R[2][38][64], int top);
+void AAC_RENAME(ff_ps_init)(void);
+void AAC_RENAME(ff_ps_ctx_init)(PSContext *ps);
+int AAC_RENAME(ff_ps_read_data)(AVCodecContext *avctx, GetBitContext *gb, PSContext *ps, int bits_left);
+int AAC_RENAME(ff_ps_apply)(AVCodecContext *avctx, PSContext *ps, INTFLOAT L[2][38][64], INTFLOAT R[2][38][64], int top);
 
 #endif /* AVCODEC_PS_H */

libavcodec/aacps_fixed.c

@@ -0,0 +1,24 @@
+/*
+ * MPEG-4 Parametric Stereo decoding functions
+ * Copyright (c) 2010 Alex Converse <alex.converse@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
+ */
+
+#define USE_FIXED 1
+
+#include "aacps.c"

libavcodec/aacps_fixed_tablegen.c

@@ -0,0 +1,24 @@
+/*
+ * Generate a header file for hardcoded Parametric Stereo tables
+ *
+ * Copyright (c) 2010 Alex Converse <alex.converse@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
+ */
+
+#define USE_FIXED 1
+#include "aacps_tablegen_template.c"

libavcodec/aacps_fixed_tablegen.h

@@ -0,0 +1,403 @@
+/*
+ * Header file for hardcoded Parametric Stereo tables
+ *
+ * Copyright (c) 2010 Alex Converse <alex.converse@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
+ *
+ * Note: Rounding-to-nearest used unless otherwise stated
+ *
+ */
+
+#ifndef AACPS_FIXED_TABLEGEN_H
+#define AACPS_FIXED_TABLEGEN_H
+
+#include <math.h>
+#include <stdint.h>
+
+#if CONFIG_HARDCODED_TABLES
+#define ps_tableinit()
+#define TABLE_CONST const
+#include "libavcodec/aacps_fixed_tables.h"
+#else
+#include "libavutil/common.h"
+#include "libavutil/mathematics.h"
+#include "libavutil/mem.h"
+
+#include "aac_defines.h"
+#include "libavutil/softfloat.h"
+#define NR_ALLPASS_BANDS20 30
+#define NR_ALLPASS_BANDS34 50
+#define PS_AP_LINKS 3
+#define TABLE_CONST
+static int pd_re_smooth[8*8*8];
+static int pd_im_smooth[8*8*8];
+static int HA[46][8][4];
+static int HB[46][8][4];
+static DECLARE_ALIGNED(16, int, f20_0_8) [ 8][8][2];
+static DECLARE_ALIGNED(16, int, f34_0_12)[12][8][2];
+static DECLARE_ALIGNED(16, int, f34_1_8) [ 8][8][2];
+static DECLARE_ALIGNED(16, int, f34_2_4) [ 4][8][2];
+static TABLE_CONST DECLARE_ALIGNED(16, int, Q_fract_allpass)[2][50][3][2];
+static DECLARE_ALIGNED(16, int, phi_fract)[2][50][2];
+
+static const int g0_Q8[] = {
+    Q31(0.00746082949812f), Q31(0.02270420949825f), Q31(0.04546865930473f), Q31(0.07266113929591f),
+    Q31(0.09885108575264f), Q31(0.11793710567217f), Q31(0.125f)
+};
+
+static const int g0_Q12[] = {
+    Q31(0.04081179924692f), Q31(0.03812810994926f), Q31(0.05144908135699f), Q31(0.06399831151592f),
+    Q31(0.07428313801106f), Q31(0.08100347892914f), Q31(0.08333333333333f)
+};
+
+static const int g1_Q8[] = {
+    Q31(0.01565675600122f), Q31(0.03752716391991f), Q31(0.05417891378782f), Q31(0.08417044116767f),
+    Q31(0.10307344158036f), Q31(0.12222452249753f), Q31(0.125f)
+};
+
+static const int g2_Q4[] = {
+    Q31(-0.05908211155639f), Q31(-0.04871498374946f), Q31(0.0f),   Q31(0.07778723915851f),
+    Q31( 0.16486303567403f), Q31( 0.23279856662996f), Q31(0.25f)
+};
+
+static const int sintbl_4[4]   = {           0,  1073741824,           0, -1073741824 };
+static const int costbl_4[4]   = {  1073741824,           0, -1073741824,           0 };
+static const int sintbl_8[8]   = {           0,   759250125,  1073741824,   759250125,
+                                             0,  -759250125, -1073741824,  -759250125 };
+static const int costbl_8[8]   = {  1073741824,   759250125,           0,  -759250125,
+                                   -1073741824,  -759250125,           0,   759250125 };
+static const int sintbl_12[12] = {           0,   536870912,   929887697,  1073741824,
+                                     929887697,   536870912,           0,  -536870912,
+                                    -929887697, -1073741824,  -929887697,  -536870912 };
+static const int costbl_12[12] = {  1073741824,   929887697,   536870912,           0,
+                                    -536870912,  -929887697, -1073741824,  -929887697,
+                                    -536870912,           0,   536870912,   929887697 };
+
+static void make_filters_from_proto(int (*filter)[8][2], const int *proto, int bands)
+{
+
+    const int *sinptr, *cosptr;
+    int s, c, sinhalf, coshalf;
+    int q, n;
+
+    if (bands == 4) {
+        sinptr = sintbl_4;
+        cosptr = costbl_4;
+        sinhalf = 759250125;
+        coshalf = 759250125;
+    } else if (bands == 8) {
+        sinptr = sintbl_8;
+        cosptr = costbl_8;
+        sinhalf = 410903207;
+        coshalf = 992008094;
+    } else {
+        sinptr = sintbl_12;
+        cosptr = costbl_12;
+        sinhalf = 277904834;
+        coshalf = 1037154959;
+    }
+
+    for (q = 0; q < bands; q++) {
+        for (n = 0; n < 7; n++) {
+            int theta = (q*(n-6) + (n>>1) - 3) % bands;
+
+            if (theta < 0)
+                theta += bands;
+            s = sinptr[theta];
+            c = cosptr[theta];
+
+            if (n & 1) {
+                theta = (int)(((int64_t)c * coshalf - (int64_t)s * sinhalf + 0x20000000) >> 30);
+                s = (int)(((int64_t)s * coshalf + (int64_t)c * sinhalf + 0x20000000) >> 30);
+                c = theta;
+            }
+            filter[q][n][0] = (int)(((int64_t)proto[n] * c + 0x20000000) >> 30);
+            filter[q][n][1] = -(int)(((int64_t)proto[n] * s + 0x20000000) >> 30);
+        }
+    }
+}
+
+static void ps_tableinit(void)
+{
+    static const int ipdopd_sin[] = { Q30(0), Q30(M_SQRT1_2), Q30(1), Q30( M_SQRT1_2), Q30( 0), Q30(-M_SQRT1_2), Q30(-1), Q30(-M_SQRT1_2) };
+    static const int ipdopd_cos[] = { Q30(1), Q30(M_SQRT1_2), Q30(0), Q30(-M_SQRT1_2), Q30(-1), Q30(-M_SQRT1_2), Q30( 0), Q30( M_SQRT1_2) };
+    int pd0, pd1, pd2;
+    int idx;
+
+    static const int alpha_tab[] =
+    {
+      Q30(1.5146213770f/M_PI), Q30(1.5181334019f/M_PI), Q30(1.5234849453f/M_PI), Q30(1.5369486809f/M_PI), Q30(1.5500687361f/M_PI), Q30(1.5679757595f/M_PI),
+      Q30(1.4455626011f/M_PI), Q30(1.4531552792f/M_PI), Q30(1.4648091793f/M_PI), Q30(1.4945238829f/M_PI), Q30(1.5239057541f/M_PI), Q30(1.5644006729f/M_PI),
+      Q30(1.3738563061f/M_PI), Q30(1.3851221800f/M_PI), Q30(1.4026404619f/M_PI), Q30(1.4484288692f/M_PI), Q30(1.4949874878f/M_PI), Q30(1.5604078770f/M_PI),
+      Q30(1.2645189762f/M_PI), Q30(1.2796478271f/M_PI), Q30(1.3038636446f/M_PI), Q30(1.3710125685f/M_PI), Q30(1.4443849325f/M_PI), Q30(1.5532352924f/M_PI),
+      Q30(1.1507037878f/M_PI), Q30(1.1669205427f/M_PI), Q30(1.1938756704f/M_PI), Q30(1.2754167318f/M_PI), Q30(1.3761177063f/M_PI), Q30(1.5429240465f/M_PI),
+      Q30(1.0079245567f/M_PI), Q30(1.0208238363f/M_PI), Q30(1.0433073044f/M_PI), Q30(1.1208510399f/M_PI), Q30(1.2424604893f/M_PI), Q30(1.5185726881f/M_PI),
+      Q30(0.8995233774f/M_PI), Q30(0.9069069624f/M_PI), Q30(0.9201194048f/M_PI), Q30(0.9698365927f/M_PI), Q30(1.0671583414f/M_PI), Q30(1.4647934437f/M_PI),
+      Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI),
+      Q30(0.6712729335f/M_PI), Q30(0.6638893485f/M_PI), Q30(0.6506769061f/M_PI), Q30(0.6009597182f/M_PI), Q30(0.5036380291f/M_PI), Q30(0.1060028747f/M_PI),
+      Q30(0.5628717542f/M_PI), Q30(0.5499725342f/M_PI), Q30(0.5274890065f/M_PI), Q30(0.4499453008f/M_PI), Q30(0.3283358216f/M_PI), Q30(0.0522236861f/M_PI),
+      Q30(0.4200925827f/M_PI), Q30(0.4038758278f/M_PI), Q30(0.3769206405f/M_PI), Q30(0.2953795493f/M_PI), Q30(0.1946786791f/M_PI), Q30(0.0278722942f/M_PI),
+      Q30(0.3062773645f/M_PI), Q30(0.2911485136f/M_PI), Q30(0.2669326365f/M_PI), Q30(0.1997837722f/M_PI), Q30(0.1264114529f/M_PI), Q30(0.0175609849f/M_PI),
+      Q30(0.1969399750f/M_PI), Q30(0.1856741160f/M_PI), Q30(0.1681558639f/M_PI), Q30(0.1223674342f/M_PI), Q30(0.0758088827f/M_PI), Q30(0.0103884479f/M_PI),
+      Q30(0.1252337098f/M_PI), Q30(0.1176410317f/M_PI), Q30(0.1059871912f/M_PI), Q30(0.0762724727f/M_PI), Q30(0.0468905345f/M_PI), Q30(0.0063956482f/M_PI),
+      Q30(0.0561749674f/M_PI), Q30(0.0526629239f/M_PI), Q30(0.0473113805f/M_PI), Q30(0.0338476151f/M_PI), Q30(0.0207276177f/M_PI), Q30(0.0028205961f/M_PI),
+      Q30(1.5676341057f/M_PI), Q30(1.5678333044f/M_PI), Q30(1.5681363344f/M_PI), Q30(1.5688960552f/M_PI), Q30(1.5696337223f/M_PI), Q30(1.5706381798f/M_PI),
+      Q30(1.5651730299f/M_PI), Q30(1.5655272007f/M_PI), Q30(1.5660660267f/M_PI), Q30(1.5674170256f/M_PI), Q30(1.5687289238f/M_PI), Q30(1.5705151558f/M_PI),
+      Q30(1.5607966185f/M_PI), Q30(1.5614265203f/M_PI), Q30(1.5623844862f/M_PI), Q30(1.5647867918f/M_PI), Q30(1.5671195984f/M_PI), Q30(1.5702962875f/M_PI),
+      Q30(1.5530153513f/M_PI), Q30(1.5541347265f/M_PI), Q30(1.5558375120f/M_PI), Q30(1.5601085424f/M_PI), Q30(1.5642569065f/M_PI), Q30(1.5699069500f/M_PI),
+      Q30(1.5391840935f/M_PI), Q30(1.5411708355f/M_PI), Q30(1.5441943407f/M_PI), Q30(1.5517836809f/M_PI), Q30(1.5591609478f/M_PI), Q30(1.5692136288f/M_PI),
+      Q30(1.5146213770f/M_PI), Q30(1.5181334019f/M_PI), Q30(1.5234849453f/M_PI), Q30(1.5369486809f/M_PI), Q30(1.5500687361f/M_PI), Q30(1.5679757595f/M_PI),
+      Q30(1.4915299416f/M_PI), Q30(1.4964480400f/M_PI), Q30(1.5039558411f/M_PI), Q30(1.5229074955f/M_PI), Q30(1.5414420366f/M_PI), Q30(1.5667995214f/M_PI),
+      Q30(1.4590617418f/M_PI), Q30(1.4658898115f/M_PI), Q30(1.4763505459f/M_PI), Q30(1.5029321909f/M_PI), Q30(1.5291173458f/M_PI), Q30(1.5651149750f/M_PI),
+      Q30(1.4136143923f/M_PI), Q30(1.4229322672f/M_PI), Q30(1.4373078346f/M_PI), Q30(1.4743183851f/M_PI), Q30(1.5113102198f/M_PI), Q30(1.5626684427f/M_PI),
+      Q30(1.3505556583f/M_PI), Q30(1.3628427982f/M_PI), Q30(1.3820509911f/M_PI), Q30(1.4327841997f/M_PI), Q30(1.4850014448f/M_PI), Q30(1.5590143204f/M_PI),
+      Q30(1.2645189762f/M_PI), Q30(1.2796478271f/M_PI), Q30(1.3038636446f/M_PI), Q30(1.3710125685f/M_PI), Q30(1.4443849325f/M_PI), Q30(1.5532352924f/M_PI),
+      Q30(1.1919227839f/M_PI), Q30(1.2081253529f/M_PI), Q30(1.2346779108f/M_PI), Q30(1.3123005629f/M_PI), Q30(1.4034168720f/M_PI), Q30(1.5471596718f/M_PI),
+      Q30(1.1061993837f/M_PI), Q30(1.1219338179f/M_PI), Q30(1.1484941244f/M_PI), Q30(1.2320860624f/M_PI), Q30(1.3421301842f/M_PI), Q30(1.5373806953f/M_PI),
+      Q30(1.0079245567f/M_PI), Q30(1.0208238363f/M_PI), Q30(1.0433073044f/M_PI), Q30(1.1208510399f/M_PI), Q30(1.2424604893f/M_PI), Q30(1.5185726881f/M_PI),
+      Q30(0.8995233774f/M_PI), Q30(0.9069069624f/M_PI), Q30(0.9201194048f/M_PI), Q30(0.9698365927f/M_PI), Q30(1.0671583414f/M_PI), Q30(1.4647934437f/M_PI),
+      Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI),
+      Q30(0.6712729335f/M_PI), Q30(0.6638893485f/M_PI), Q30(0.6506769061f/M_PI), Q30(0.6009597182f/M_PI), Q30(0.5036380291f/M_PI), Q30(0.1060028747f/M_PI),
+      Q30(0.5628717542f/M_PI), Q30(0.5499725342f/M_PI), Q30(0.5274890065f/M_PI), Q30(0.4499453008f/M_PI), Q30(0.3283358216f/M_PI), Q30(0.0522236861f/M_PI),
+      Q30(0.4645969570f/M_PI), Q30(0.4488625824f/M_PI), Q30(0.4223022461f/M_PI), Q30(0.3387103081f/M_PI), Q30(0.2286661267f/M_PI), Q30(0.0334156826f/M_PI),
+      Q30(0.3788735867f/M_PI), Q30(0.3626709878f/M_PI), Q30(0.3361184299f/M_PI), Q30(0.2584958076f/M_PI), Q30(0.1673794836f/M_PI), Q30(0.0236366931f/M_PI),
+      Q30(0.3062773645f/M_PI), Q30(0.2911485136f/M_PI), Q30(0.2669326365f/M_PI), Q30(0.1997837722f/M_PI), Q30(0.1264114529f/M_PI), Q30(0.0175609849f/M_PI),
+      Q30(0.2202406377f/M_PI), Q30(0.2079535723f/M_PI), Q30(0.1887452900f/M_PI), Q30(0.1380121708f/M_PI), Q30(0.0857949182f/M_PI), Q30(0.0117820343f/M_PI),
+      Q30(0.1571819335f/M_PI), Q30(0.1478640437f/M_PI), Q30(0.1334884763f/M_PI), Q30(0.0964778885f/M_PI), Q30(0.0594860613f/M_PI), Q30(0.0081279324f/M_PI),
+      Q30(0.1117345318f/M_PI), Q30(0.1049065739f/M_PI), Q30(0.0944457650f/M_PI), Q30(0.0678641573f/M_PI), Q30(0.0416790098f/M_PI), Q30(0.0056813755f/M_PI),
+      Q30(0.0792663917f/M_PI), Q30(0.0743482932f/M_PI), Q30(0.0668405443f/M_PI), Q30(0.0478888862f/M_PI), Q30(0.0293543357f/M_PI), Q30(0.0039967746f/M_PI),
+      Q30(0.0561749674f/M_PI), Q30(0.0526629239f/M_PI), Q30(0.0473113805f/M_PI), Q30(0.0338476151f/M_PI), Q30(0.0207276177f/M_PI), Q30(0.0028205961f/M_PI),
+      Q30(0.0316122435f/M_PI), Q30(0.0296254847f/M_PI), Q30(0.0266019460f/M_PI), Q30(0.0190126132f/M_PI), Q30(0.0116353342f/M_PI), Q30(0.0015827164f/M_PI),
+      Q30(0.0177809205f/M_PI), Q30(0.0166615788f/M_PI), Q30(0.0149587989f/M_PI), Q30(0.0106877899f/M_PI), Q30(0.0065393616f/M_PI), Q30(0.0008894200f/M_PI),
+      Q30(0.0099996664f/M_PI), Q30(0.0093698399f/M_PI), Q30(0.0084118480f/M_PI), Q30(0.0060095116f/M_PI), Q30(0.0036767013f/M_PI), Q30(0.0005000498f/M_PI),
+      Q30(0.0056233541f/M_PI), Q30(0.0052691097f/M_PI), Q30(0.0047303112f/M_PI), Q30(0.0033792770f/M_PI), Q30(0.0020674451f/M_PI), Q30(0.0002811795f/M_PI),
+      Q30(0.0031622672f/M_PI), Q30(0.0029630491f/M_PI), Q30(0.0026600463f/M_PI), Q30(0.0019002859f/M_PI), Q30(0.0011625893f/M_PI), Q30(0.0001581155f/M_PI)
+    };
+
+    static const int gamma_tab[] =
+    {
+      Q30(0.0000000000f/M_PI), Q30(0.0195873566f/M_PI), Q30(0.0303316917f/M_PI), Q30(0.0448668823f/M_PI), Q30(0.0522258915f/M_PI), Q30(0.0561044961f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0433459543f/M_PI), Q30(0.0672172382f/M_PI), Q30(0.0997167900f/M_PI), Q30(0.1162951663f/M_PI), Q30(0.1250736862f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0672341362f/M_PI), Q30(0.1045235619f/M_PI), Q30(0.1558904350f/M_PI), Q30(0.1824723780f/M_PI), Q30(0.1966800541f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1011129096f/M_PI), Q30(0.1580764502f/M_PI), Q30(0.2387557179f/M_PI), Q30(0.2820728719f/M_PI), Q30(0.3058380187f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1315985769f/M_PI), Q30(0.2072522491f/M_PI), Q30(0.3188187480f/M_PI), Q30(0.3825501204f/M_PI), Q30(0.4193951190f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1603866369f/M_PI), Q30(0.2549437582f/M_PI), Q30(0.4029446840f/M_PI), Q30(0.4980689585f/M_PI), Q30(0.5615641475f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1736015975f/M_PI), Q30(0.2773745656f/M_PI), Q30(0.4461984038f/M_PI), Q30(0.5666890144f/M_PI), Q30(0.6686112881f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1784276664f/M_PI), Q30(0.2856673002f/M_PI), Q30(0.4630723596f/M_PI), Q30(0.5971632004f/M_PI), Q30(0.7603877187f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1736015975f/M_PI), Q30(0.2773745656f/M_PI), Q30(0.4461984038f/M_PI), Q30(0.5666890144f/M_PI), Q30(0.6686112881f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1603866369f/M_PI), Q30(0.2549437582f/M_PI), Q30(0.4029446840f/M_PI), Q30(0.4980689585f/M_PI), Q30(0.5615641475f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1315985769f/M_PI), Q30(0.2072522491f/M_PI), Q30(0.3188187480f/M_PI), Q30(0.3825501204f/M_PI), Q30(0.4193951190f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1011129096f/M_PI), Q30(0.1580764502f/M_PI), Q30(0.2387557179f/M_PI), Q30(0.2820728719f/M_PI), Q30(0.3058380187f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0672341362f/M_PI), Q30(0.1045235619f/M_PI), Q30(0.1558904350f/M_PI), Q30(0.1824723780f/M_PI), Q30(0.1966800541f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0433459543f/M_PI), Q30(0.0672172382f/M_PI), Q30(0.0997167900f/M_PI), Q30(0.1162951663f/M_PI), Q30(0.1250736862f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0195873566f/M_PI), Q30(0.0303316917f/M_PI), Q30(0.0448668823f/M_PI), Q30(0.0522258915f/M_PI), Q30(0.0561044961f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0011053939f/M_PI), Q30(0.0017089852f/M_PI), Q30(0.0025254129f/M_PI), Q30(0.0029398468f/M_PI), Q30(0.0031597170f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0019607407f/M_PI), Q30(0.0030395309f/M_PI), Q30(0.0044951206f/M_PI), Q30(0.0052305623f/M_PI), Q30(0.0056152637f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0034913034f/M_PI), Q30(0.0054070661f/M_PI), Q30(0.0079917293f/M_PI), Q30(0.0092999367f/M_PI), Q30(0.0099875759f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0062100487f/M_PI), Q30(0.0096135242f/M_PI), Q30(0.0142110568f/M_PI), Q30(0.0165348612f/M_PI), Q30(0.0177587029f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0110366223f/M_PI), Q30(0.0170863140f/M_PI), Q30(0.0252620988f/M_PI), Q30(0.0293955617f/M_PI), Q30(0.0315726399f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0195873566f/M_PI), Q30(0.0303316917f/M_PI), Q30(0.0448668823f/M_PI), Q30(0.0522258915f/M_PI), Q30(0.0561044961f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0275881495f/M_PI), Q30(0.0427365713f/M_PI), Q30(0.0632618815f/M_PI), Q30(0.0736731067f/M_PI), Q30(0.0791663304f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0387469754f/M_PI), Q30(0.0600636788f/M_PI), Q30(0.0890387669f/M_PI), Q30(0.1037906483f/M_PI), Q30(0.1115923747f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0541138873f/M_PI), Q30(0.0839984417f/M_PI), Q30(0.1248718798f/M_PI), Q30(0.1458375156f/M_PI), Q30(0.1569785923f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0747506917f/M_PI), Q30(0.1163287833f/M_PI), Q30(0.1738867164f/M_PI), Q30(0.2038587779f/M_PI), Q30(0.2199459076f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1011129096f/M_PI), Q30(0.1580764502f/M_PI), Q30(0.2387557179f/M_PI), Q30(0.2820728719f/M_PI), Q30(0.3058380187f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1212290376f/M_PI), Q30(0.1903949380f/M_PI), Q30(0.2907958031f/M_PI), Q30(0.3466993868f/M_PI), Q30(0.3782821596f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1418247074f/M_PI), Q30(0.2240308374f/M_PI), Q30(0.3474813402f/M_PI), Q30(0.4202919006f/M_PI), Q30(0.4637607038f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1603866369f/M_PI), Q30(0.2549437582f/M_PI), Q30(0.4029446840f/M_PI), Q30(0.4980689585f/M_PI), Q30(0.5615641475f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1736015975f/M_PI), Q30(0.2773745656f/M_PI), Q30(0.4461984038f/M_PI), Q30(0.5666890144f/M_PI), Q30(0.6686112881f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1784276664f/M_PI), Q30(0.2856673002f/M_PI), Q30(0.4630723596f/M_PI), Q30(0.5971632004f/M_PI), Q30(0.7603877187f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1736015975f/M_PI), Q30(0.2773745656f/M_PI), Q30(0.4461984038f/M_PI), Q30(0.5666890144f/M_PI), Q30(0.6686112881f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1603866369f/M_PI), Q30(0.2549437582f/M_PI), Q30(0.4029446840f/M_PI), Q30(0.4980689585f/M_PI), Q30(0.5615641475f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1418247074f/M_PI), Q30(0.2240308374f/M_PI), Q30(0.3474813402f/M_PI), Q30(0.4202919006f/M_PI), Q30(0.4637607038f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1212290376f/M_PI), Q30(0.1903949380f/M_PI), Q30(0.2907958031f/M_PI), Q30(0.3466993868f/M_PI), Q30(0.3782821596f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.1011129096f/M_PI), Q30(0.1580764502f/M_PI), Q30(0.2387557179f/M_PI), Q30(0.2820728719f/M_PI), Q30(0.3058380187f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0747506917f/M_PI), Q30(0.1163287833f/M_PI), Q30(0.1738867164f/M_PI), Q30(0.2038587779f/M_PI), Q30(0.2199459076f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0541138873f/M_PI), Q30(0.0839984417f/M_PI), Q30(0.1248718798f/M_PI), Q30(0.1458375156f/M_PI), Q30(0.1569785923f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0387469754f/M_PI), Q30(0.0600636788f/M_PI), Q30(0.0890387669f/M_PI), Q30(0.1037906483f/M_PI), Q30(0.1115923747f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0275881495f/M_PI), Q30(0.0427365713f/M_PI), Q30(0.0632618815f/M_PI), Q30(0.0736731067f/M_PI), Q30(0.0791663304f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0195873566f/M_PI), Q30(0.0303316917f/M_PI), Q30(0.0448668823f/M_PI), Q30(0.0522258915f/M_PI), Q30(0.0561044961f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0110366223f/M_PI), Q30(0.0170863140f/M_PI), Q30(0.0252620988f/M_PI), Q30(0.0293955617f/M_PI), Q30(0.0315726399f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0062100487f/M_PI), Q30(0.0096135242f/M_PI), Q30(0.0142110568f/M_PI), Q30(0.0165348612f/M_PI), Q30(0.0177587029f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0034913034f/M_PI), Q30(0.0054070661f/M_PI), Q30(0.0079917293f/M_PI), Q30(0.0092999367f/M_PI), Q30(0.0099875759f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0019607407f/M_PI), Q30(0.0030395309f/M_PI), Q30(0.0044951206f/M_PI), Q30(0.0052305623f/M_PI), Q30(0.0056152637f/M_PI),
+      Q30(0.0000000000f/M_PI), Q30(0.0011053939f/M_PI), Q30(0.0017089852f/M_PI), Q30(0.0025254129f/M_PI), Q30(0.0029398468f/M_PI), Q30(0.0031597170f/M_PI)
+    };
+
+    static const int iid_par_dequant_c1[] = {
+        //iid_par_dequant_default
+        Q30(1.41198278375959f), Q30(1.40313815268360f), Q30(1.38687670404960f), Q30(1.34839972492648f),
+        Q30(1.29124937110028f), Q30(1.19603741667993f), Q30(1.10737240362323f), Q30(1),
+        Q30(0.87961716655242f), Q30(0.75464859232732f), Q30(0.57677990744575f), Q30(0.42640143271122f),
+        Q30(0.27671828230984f), Q30(0.17664462766713f), Q30(0.07940162697653f),
+        //iid_par_dequant_fine
+        Q30(1.41420649135832f), Q30(1.41419120222364f), Q30(1.41414285699784f), Q30(1.41399000859438f),
+        Q30(1.41350698548044f), Q30(1.41198278375959f), Q30(1.40977302262355f), Q30(1.40539479488545f),
+        Q30(1.39677960498402f), Q30(1.38005309967827f), Q30(1.34839972492648f), Q30(1.31392017367631f),
+        Q30(1.26431008149654f), Q30(1.19603741667993f), Q30(1.10737240362323f), Q30(1),
+        Q30(0.87961716655242f), Q30(0.75464859232732f), Q30(0.63365607219232f), Q30(0.52308104267543f),
+        Q30(0.42640143271122f), Q30(0.30895540465965f), Q30(0.22137464873077f), Q30(0.15768788954414f),
+        Q30(0.11198225164225f), Q30(0.07940162697653f), Q30(0.04469901562677f), Q30(0.02514469318284f),
+        Q30(0.01414142856998f), Q30(0.00795258154731f), Q30(0.00447211359449f),
+    };
+
+    static const int acos_icc_invq[] = {
+        Q31(0), Q31(0.178427635f/M_PI), Q31(0.28566733f/M_PI), Q31(0.46307236f/M_PI), Q31(0.59716315f/M_PI), Q31(0.78539816f/M_PI), Q31(1.10030855f/M_PI), Q31(1.57079633f/M_PI)
+    };
+    int iid, icc;
+
+    int k, m;
+    static const int8_t f_center_20[] = {
+        -3, -1, 1, 3, 5, 7, 10, 14, 18, 22,
+    };
+    static const int32_t f_center_34[] = {
+      Q31(  2/768.0),Q31(  6/768.0),Q31(10/768.0),Q31(14/768.0),Q31( 18/768.0),Q31( 22/768.0),Q31( 26/768.0),Q31(30/768.0),
+      Q31( 34/768.0),Q31(-10/768.0),Q31(-6/768.0),Q31(-2/768.0),Q31( 51/768.0),Q31( 57/768.0),Q31( 15/768.0),Q31(21/768.0),
+      Q31( 27/768.0),Q31( 33/768.0),Q31(39/768.0),Q31(45/768.0),Q31( 54/768.0),Q31( 66/768.0),Q31( 78/768.0),Q31(42/768.0),
+      Q31(102/768.0),Q31( 66/768.0),Q31(78/768.0),Q31(90/768.0),Q31(102/768.0),Q31(114/768.0),Q31(126/768.0),Q31(90/768.0)
+    };
+    static const int fractional_delay_links[] = { Q31(0.43f), Q31(0.75f), Q31(0.347f) };
+    const int fractional_delay_gain = Q31(0.39f);
+
+    for (pd0 = 0; pd0 < 8; pd0++) {
+        int pd0_re = (ipdopd_cos[pd0]+2)>>2;
+        int pd0_im = (ipdopd_sin[pd0]+2)>>2;
+        for (pd1 = 0; pd1 < 8; pd1++) {
+            int pd1_re = ipdopd_cos[pd1] >> 1;
+            int pd1_im = ipdopd_sin[pd1] >> 1;
+            for (pd2 = 0; pd2 < 8; pd2++) {
+                int shift, round;
+                int pd2_re = ipdopd_cos[pd2];
+                int pd2_im = ipdopd_sin[pd2];
+                int re_smooth = pd0_re + pd1_re + pd2_re;
+                int im_smooth = pd0_im + pd1_im + pd2_im;
+
+                SoftFloat pd_mag = av_int2sf(((ipdopd_cos[(pd0-pd1)&7]+8)>>4) + ((ipdopd_cos[(pd0-pd2)&7]+4)>>3) +
+                                               ((ipdopd_cos[(pd1-pd2)&7]+2)>>2) + 0x15000000, 28);
+                pd_mag = av_div_sf(FLOAT_1, av_sqrt_sf(pd_mag));
+                shift = 30 - pd_mag.exp;
+                round = 1 << (shift-1);
+                pd_re_smooth[pd0*64+pd1*8+pd2] = (int)(((int64_t)re_smooth * pd_mag.mant + round) >> shift);
+                pd_im_smooth[pd0*64+pd1*8+pd2] = (int)(((int64_t)im_smooth * pd_mag.mant + round) >> shift);
+            }
+        }
+    }
+
+    idx = 0;
+    for (iid = 0; iid < 46; iid++) {
+        int c1, c2;
+
+        c1 = iid_par_dequant_c1[iid];
+        if (iid < 15)
+          c2 = iid_par_dequant_c1[14-iid];
+        else
+          c2 = iid_par_dequant_c1[60-iid];
+
+        for (icc = 0; icc < 8; icc++) {
+            /*if (PS_BASELINE || ps->icc_mode < 3)*/{
+                int alpha, beta;
+                int ca, sa, cb, sb;
+
+                alpha = acos_icc_invq[icc];
+                beta = (int)(((int64_t)alpha * 1518500250 + 0x40000000) >> 31);
+                alpha >>= 1;
+                beta = (int)(((int64_t)beta * (c1 - c2) + 0x40000000) >> 31);
+                av_sincos_sf(beta + alpha, &sa, &ca);
+                av_sincos_sf(beta - alpha, &sb, &cb);
+
+                HA[iid][icc][0] = (int)(((int64_t)c2 * ca + 0x20000000) >> 30);
+                HA[iid][icc][1] = (int)(((int64_t)c1 * cb + 0x20000000) >> 30);
+                HA[iid][icc][2] = (int)(((int64_t)c2 * sa + 0x20000000) >> 30);
+                HA[iid][icc][3] = (int)(((int64_t)c1 * sb + 0x20000000) >> 30);
+            } /* else */ {
+                int alpha_int, gamma_int;
+                int alpha_c_int, alpha_s_int, gamma_c_int, gamma_s_int;
+
+                alpha_int = alpha_tab[idx];
+                gamma_int = gamma_tab[idx];
+
+                av_sincos_sf(alpha_int, &alpha_s_int, &alpha_c_int);
+                av_sincos_sf(gamma_int, &gamma_s_int, &gamma_c_int);
+
+                alpha_c_int = (int)(((int64_t)alpha_c_int * 1518500250 + 0x20000000) >> 30);
+                alpha_s_int = (int)(((int64_t)alpha_s_int * 1518500250 + 0x20000000) >> 30);
+
+                HB[iid][icc][0] = (int)(((int64_t)alpha_c_int * gamma_c_int + 0x20000000) >> 30);
+                HB[iid][icc][1] = (int)(((int64_t)alpha_s_int * gamma_c_int + 0x20000000) >> 30);
+                HB[iid][icc][2] = -(int)(((int64_t)alpha_s_int * gamma_s_int + 0x20000000) >> 30);
+                HB[iid][icc][3] = (int)(((int64_t)alpha_c_int * gamma_s_int + 0x20000000) >> 30);
+            }
+
+            if (icc < 5 || icc > 6)
+              idx++;
+        }
+    }
+
+    for (k = 0; k < NR_ALLPASS_BANDS20; k++) {
+        int theta;
+        int64_t f_center;
+        int c, s;
+
+        if (k < FF_ARRAY_ELEMS(f_center_20))
+          f_center = f_center_20[k];
+        else
+          f_center = (k << 3) - 52;
+
+        for (m = 0; m < PS_AP_LINKS; m++) {
+            theta = (int)(((int64_t)fractional_delay_links[m] * f_center + 8) >> 4);
+            av_sincos_sf(-theta, &s, &c);
+            Q_fract_allpass[0][k][m][0] = c;
+            Q_fract_allpass[0][k][m][1] = s;
+        }
+
+        theta = (int)(((int64_t)fractional_delay_gain * f_center + 8) >> 4);
+        av_sincos_sf(-theta, &s, &c);
+        phi_fract[0][k][0] = c;
+        phi_fract[0][k][1] = s;
+    }
+
+    for (k = 0; k < NR_ALLPASS_BANDS34; k++) {
+        int theta, f_center;
+        int c, s;
+
+        if (k < FF_ARRAY_ELEMS(f_center_34))
+            f_center = f_center_34[k];
+        else
+            f_center = ((int64_t)k << 26) - (53 << 25);
+
+        for (m = 0; m < PS_AP_LINKS; m++) {
+            theta = (int)(((int64_t)fractional_delay_links[m] * f_center + 0x10000000) >> 27);
+            av_sincos_sf(-theta, &s, &c);
+            Q_fract_allpass[1][k][m][0] = c;
+            Q_fract_allpass[1][k][m][1] = s;
+        }
+
+        theta = (int)(((int64_t)fractional_delay_gain * f_center + 0x10000000) >> 27);
+        av_sincos_sf(-theta, &s, &c);
+        phi_fract[1][k][0] = c;
+        phi_fract[1][k][1] = s;
+    }
+
+    make_filters_from_proto(f20_0_8,  g0_Q8,   8);
+    make_filters_from_proto(f34_0_12, g0_Q12, 12);
+    make_filters_from_proto(f34_1_8,  g1_Q8,   8);
+    make_filters_from_proto(f34_2_4,  g2_Q4,   4);
+}
+#endif /* CONFIG_HARDCODED_TABLES */
+
+#endif /* AACPS_FIXED_TABLEGEN_H */