avcodec/g2meet: fix src pointer checks in kempf_decode_tile()

Fixes Ticket2842

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 2960576378)

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

.gitignore

@@ -63,7 +63,6 @@
 /tools/aviocat
 /tools/ffbisect
 /tools/bisect.need
-/tools/crypto_bench
 /tools/cws2fws
 /tools/fourcc2pixfmt
 /tools/ffescape

Changelog

@@ -1,55 +1,6 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.
 
-version <next>
-
-
-version 2.1:
-
-- aecho filter
-- perspective filter ported from libmpcodecs
-- ffprobe -show_programs option
-- compand filter
-- RTMP seek support
-- when transcoding with ffmpeg (i.e. not streamcopying), -ss is now accurate
-  even when used as an input option. Previous behavior can be restored with
-  the -noaccurate_seek option.
-- ffmpeg -t option can now be used for inputs, to limit the duration of
-  data read from an input file
-- incomplete Voxware MetaSound decoder
-- read EXIF metadata from JPEG
-- DVB teletext decoder
-- phase filter ported from libmpcodecs
-- w3fdif filter
-- Opus support in Matroska
-- FFV1 version 1.3 is stable and no longer experimental
-- FFV1: YUVA(444,422,420) 9, 10 and 16 bit support
-- changed DTS stream id in lavf mpeg ps muxer from 0x8a to 0x88, to be
-  more consistent with other muxers.
-- adelay filter
-- pullup filter ported from libmpcodecs
-- ffprobe -read_intervals option
-- Lossless and alpha support for WebP decoder
-- Error Resilient AAC syntax (ER AAC LC) decoding
-- Low Delay AAC (ER AAC LD) decoding
-- mux chapters in ASF files
-- SFTP protocol (via libssh)
-- libx264: add ability to encode in YUVJ422P and YUVJ444P
-- Fraps: use BT.709 colorspace by default for yuv, as reference fraps decoder does
-- make decoding alpha optional for prores, ffv1 and vp6 by setting
-  the skip_alpha flag.
-- ladspa wrapper filter
-- native VP9 decoder
-- dpx parser
-- max_error_rate parameter in ffmpeg
-- PulseAudio output device
-- ReplayGain scanner
-- Enhanced Low Delay AAC (ER AAC ELD) decoding (no LD SBR support)
-- Linux framebuffer output device
-- HEVC decoder, raw HEVC demuxer, HEVC demuxing in TS, Matroska and MP4
-- mergeplanes filter
-
-
 version 2.0:
 
 - curves filter
@@ -656,7 +607,7 @@ version 0.6:
 - LPCM support in MPEG-TS (HDMV RID as found on Blu-ray disks)
 - WMA Pro decoder
 - Core Audio Format demuxer
-- ATRAC1 decoder
+- Atrac1 decoder
 - MD STUDIO audio demuxer
 - RF64 support in WAV demuxer
 - MPEG-4 Audio Lossless Coding (ALS) decoder
@@ -756,7 +707,7 @@ version 0.5:
 - MXF demuxer
 - VC-1/WMV3/WMV9 video decoder
 - MacIntel support
-- AviSynth support
+- AVISynth support
 - VMware video decoder
 - VP5 video decoder
 - VP6 video decoder
@@ -784,7 +735,7 @@ version 0.5:
 - Interplay C93 demuxer and video decoder
 - Bethsoft VID demuxer and video decoder
 - CRYO APC demuxer
-- ATRAC3 decoder
+- Atrac3 decoder
 - V.Flash PTX decoder
 - RoQ muxer, RoQ audio encoder
 - Renderware TXD demuxer and decoder

LICENSE

@@ -33,14 +33,13 @@ Specifically, the GPL parts of FFmpeg are
     - vf_geq.c
     - vf_histeq.c
     - vf_hqdn3d.c
+    - vf_hue.c
     - vf_kerndeint.c
     - vf_mcdeint.c
     - vf_mp.c
+    - vf_noise.c
     - vf_owdenoise.c
-    - vf_perspective.c
-    - vf_phase.c
     - vf_pp.c
-    - vf_pullup.c
     - vf_sab.c
     - vf_smartblur.c
     - vf_spp.c

MAINTAINERS

@@ -43,24 +43,16 @@ QuickTime faststart:
 Miscellaneous Areas
 ===================
 
-documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu
+documentation                           Mike Melanson
+website                                 Robert Swain, Lou Logan
 build system (configure,Makefiles)      Diego Biurrun, Mans Rullgard
 project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger, Alexander Strasser
+mailinglists                            Michael Niedermayer, Baptiste Coudurier, Lou Logan
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
 release management                      Michael Niedermayer
 
 
-Communication
-=============
-
-website                                 Robert Swain, Lou Logan
-mailinglists                            Michael Niedermayer, Baptiste Coudurier, Lou Logan
-Google+                                 Paul B Mahol, Michael Niedermayer, Alexander Strasser
-Twitter                                 Lou Logan
-Launchpad                               Timothy Gu
-
-
 libavutil
 =========
 
@@ -73,16 +65,13 @@ Other:
   bprint                                Nicolas George
   bswap.h
   des                                   Reimar Doeffinger
-  eval.c, eval.h                        Michael Niedermayer
   float_dsp                             Loren Merritt
   hash                                  Reimar Doeffinger
   intfloat*                             Michael Niedermayer
   integer.c, integer.h                  Michael Niedermayer
   lzo                                   Reimar Doeffinger
   mathematics.c, mathematics.h          Michael Niedermayer
-  mem.c, mem.h                          Michael Niedermayer
   opencl.c, opencl.h                    Wei Gao
-  opt.c, opt.h                          Michael Niedermayer
   rational.c, rational.h                Michael Niedermayer
   rc4                                   Reimar Doeffinger
   ripemd.c, ripemd.h                    James Almer
@@ -97,6 +86,10 @@ Generic Parts:
     avcodec.h                           Michael Niedermayer
   utility code:
     utils.c                             Michael Niedermayer
+    mem.c                               Michael Niedermayer
+    opt.c, opt.h                        Michael Niedermayer
+  arithmetic expression evaluator:
+    eval.c                              Michael Niedermayer
   audio and video frame extraction:
     parser.c                            Michael Niedermayer
   bitstream reading:
@@ -127,8 +120,6 @@ Generic Parts:
     libpostproc/*                       Michael Niedermayer
   table generation:
     tableprint.c, tableprint.h          Reimar Doeffinger
-  fixed point FFT:
-    fft*                                Zeljko Lukac
 
 Codecs:
   4xm.c                                 Michael Niedermayer
@@ -162,7 +153,6 @@ Codecs:
   dv.c                                  Roman Shaposhnik
   dxa.c                                 Kostya Shishkov
   eacmv*, eaidct*, eat*                 Peter Ross
-  exif.c, exif.h                        Thilo Borgmann
   ffv1.c                                Michael Niedermayer
   ffwavesynth.c                         Nicolas George
   flac*                                 Justin Ruggles
@@ -274,7 +264,6 @@ Codecs:
   wnv1.c                                Kostya Shishkov
   xan.c                                 Mike Melanson
   xbm*                                  Paul B Mahol
-  xface                                 Stefano Sabatini
   xl.c                                  Kostya Shishkov
   xvmc.c                                Ivan Kalvachev
   xwd*                                  Paul B Mahol
@@ -297,12 +286,8 @@ libavdevice
 
 
   dshow.c                               Roger Pack
-  fbdev_enc.c                           Lukasz Marek
   iec61883.c                            Georg Lippitsch
-  lavfi                                 Stefano Sabatini
   libdc1394.c                           Roman Shaposhnik
-  pulse_audio_enc.c                     Lukasz Marek
-  sdl                                   Stefano Sabatini
   v4l2.c                                Luca Abeni
   vfwcap.c                              Ramiro Polla
 
@@ -313,33 +298,14 @@ Generic parts:
   graphdump.c                           Nicolas George
 
 Filters:
-  af_adelay.c                           Paul B Mahol
-  af_aecho.c                            Paul B Mahol
-  af_afade.c                            Paul B Mahol
   af_amerge.c                           Nicolas George
-  af_aphaser.c                          Paul B Mahol
   af_aresample.c                        Michael Niedermayer
-  af_astats.c                           Paul B Mahol
   af_astreamsync.c                      Nicolas George
   af_atempo.c                           Pavel Koshevoy
-  af_biquads.c                          Paul B Mahol
-  af_compand.c                          Paul B Mahol
-  af_ladspa.c                           Paul B Mahol
   af_pan.c                              Nicolas George
-  avf_avectorscope.c                    Paul B Mahol
-  vf_blend.c                            Paul B Mahol
-  vf_colorbalance.c                     Paul B Mahol
   vf_delogo.c                           Jean Delvare (CC <khali@linux-fr.org>)
   vf_drawbox.c/drawgrid                 Andrey Utkin
-  vf_extractplanes.c                    Paul B Mahol
-  vf_histogram.c                        Paul B Mahol
-  vf_il.c                               Paul B Mahol
-  vf_mergeplanes.c                      Paul B Mahol
-  vf_psnr.c                             Paul B Mahol
   vf_scale.c                            Michael Niedermayer
-  vf_separatefields.c                   Paul B Mahol
-  vf_stereo3d.c                         Paul B Mahol
-  vf_telecine.c                         Paul B Mahol
   vf_yadif.c                            Michael Niedermayer
 
 Sources:
@@ -462,7 +428,6 @@ Protocols:
   bluray.c                              Petri Hintukainen
   ftp.c                                 Lukasz Marek
   http.c                                Ronald S. Bultje
-  libssh.c                              Lukasz Marek
   mms*.c                                Ronald S. Bultje
   udp.c                                 Luca Abeni
 
@@ -487,7 +452,7 @@ Operating systems / CPU architectures
 Alpha                                   Mans Rullgard, Falk Hueffner
 ARM                                     Mans Rullgard
 AVR32                                   Mans Rullgard
-MIPS                                    Mans Rullgard, Nedeljko Babic
+MIPS                                    Mans Rullgard
 Mac OS X / PowerPC                      Romain Dolbeau, Guillaume Poirier
 Amiga / PowerPC                         Colin Ward
 Linux / PowerPC                         Luca Barbato
@@ -501,8 +466,8 @@ x86                                     Michael Niedermayer
 Releases
 ========
 
-2.1                                     Michael Niedermayer
 2.0                                     Michael Niedermayer
+1.2                                     Michael Niedermayer
 
 If you want to maintain an older release, please contact us
 
@@ -510,7 +475,6 @@ If you want to maintain an older release, please contact us
 GnuPG Fingerprints of maintainers and contributors
 ==================================================
 
-Alexander Strasser            1C96 78B7 83CB 8AA7 9AF5 D1EB A7D8 A57B A876 E58F
 Anssi Hannula                 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
 Anton Khirnov                 6D0C 6625 56F8 65D1 E5F5 814B B50A 1241 C067 07AB
 Ash Hughes                    694D 43D2 D180 C7C7 6421 ABD3 A641 D0B7 623D 6029

RELEASE

@@ -1 +1 @@
-2.1.4
+2.0.1

VERSION

@@ -1 +1 @@
-2.1.4
+2.0.1

cmdutils.c

@@ -47,7 +47,6 @@
 #include "libavutil/eval.h"
 #include "libavutil/dict.h"
 #include "libavutil/opt.h"
-#include "libavutil/cpu.h"
 #include "cmdutils.h"
 #include "version.h"
 #if CONFIG_NETWORK
@@ -68,7 +67,7 @@ struct SwsContext *sws_opts;
 AVDictionary *swr_opts;
 AVDictionary *format_opts, *codec_opts, *resample_opts;
 
-const int this_year = 2014;
+const int this_year = 2013;
 
 static FILE *report_file;
 
@@ -214,10 +213,7 @@ static const OptionDef *find_option(const OptionDef *po, const char *name)
     return po;
 }
 
-/* _WIN32 means using the windows libc - cygwin doesn't define that
- * by default. HAVE_COMMANDLINETOARGVW is true on cygwin, while
- * it doesn't provide the actual command line via GetCommandLineW(). */
-#if HAVE_COMMANDLINETOARGVW && defined(_WIN32)
+#if HAVE_COMMANDLINETOARGVW
 #include <windows.h>
 #include <shellapi.h>
 /* Will be leaked on exit */
@@ -498,15 +494,6 @@ void parse_loglevel(int argc, char **argv, const OptionDef *options)
     }
 }
 
-static const AVOption *opt_find(void *obj, const char *name, const char *unit,
-                            int opt_flags, int search_flags)
-{
-    const AVOption *o = av_opt_find(obj, name, unit, opt_flags, search_flags);
-    if(o && !o->flags)
-        return NULL;
-    return o;
-}
-
 #define FLAGS (o->type == AV_OPT_TYPE_FLAGS) ? AV_DICT_APPEND : 0
 int opt_default(void *optctx, const char *opt, const char *arg)
 {
@@ -527,14 +514,14 @@ int opt_default(void *optctx, const char *opt, const char *arg)
         p = opt + strlen(opt);
     av_strlcpy(opt_stripped, opt, FFMIN(sizeof(opt_stripped), p - opt + 1));
 
-    if ((o = opt_find(&cc, opt_stripped, NULL, 0,
+    if ((o = av_opt_find(&cc, opt_stripped, NULL, 0,
                          AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ)) ||
         ((opt[0] == 'v' || opt[0] == 'a' || opt[0] == 's') &&
-         (o = opt_find(&cc, opt + 1, NULL, 0, AV_OPT_SEARCH_FAKE_OBJ)))) {
+         (o = av_opt_find(&cc, opt + 1, NULL, 0, AV_OPT_SEARCH_FAKE_OBJ)))) {
         av_dict_set(&codec_opts, opt, arg, FLAGS);
         consumed = 1;
     }
-    if ((o = opt_find(&fc, opt, NULL, 0,
+    if ((o = av_opt_find(&fc, opt, NULL, 0,
                          AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
         av_dict_set(&format_opts, opt, arg, FLAGS);
         if (consumed)
@@ -543,7 +530,7 @@ 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,
+    if (!consumed && av_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);
@@ -556,7 +543,7 @@ int opt_default(void *optctx, const char *opt, const char *arg)
 #endif
 #if CONFIG_SWRESAMPLE
     swr_class = swr_get_class();
-    if (!consumed && (o=opt_find(&swr_class, opt, NULL, 0,
+    if (!consumed && (o=av_opt_find(&swr_class, opt, NULL, 0,
                                     AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
         struct SwrContext *swr = swr_alloc();
         int ret = av_opt_set(swr, opt, arg, 0);
@@ -570,7 +557,7 @@ int opt_default(void *optctx, const char *opt, const char *arg)
     }
 #endif
 #if CONFIG_AVRESAMPLE
-    if ((o=opt_find(&rc, opt, NULL, 0,
+    if ((o=av_opt_find(&rc, opt, NULL, 0,
                        AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
         av_dict_set(&resample_opts, opt, arg, FLAGS);
         consumed = 1;
@@ -809,18 +796,6 @@ do {                                                                           \
     return 0;
 }
 
-int opt_cpuflags(void *optctx, const char *opt, const char *arg)
-{
-    int ret;
-    unsigned flags = av_get_cpu_flags();
-
-    if ((ret = av_parse_cpu_caps(&flags, arg)) < 0)
-        return ret;
-
-    av_force_cpu_flags(flags);
-    return 0;
-}
-
 int opt_loglevel(void *optctx, const char *opt, const char *arg)
 {
     const struct { const char *name; int level; } log_levels[] = {
@@ -973,6 +948,18 @@ int opt_max_alloc(void *optctx, const char *opt, const char *arg)
     return 0;
 }
 
+int opt_cpuflags(void *optctx, const char *opt, const char *arg)
+{
+    int ret;
+    unsigned flags = av_get_cpu_flags();
+
+    if ((ret = av_parse_cpu_caps(&flags, arg)) < 0)
+        return ret;
+
+    av_force_cpu_flags(flags);
+    return 0;
+}
+
 int opt_timelimit(void *optctx, const char *opt, const char *arg)
 {
 #if HAVE_SETRLIMIT
@@ -1482,9 +1469,8 @@ int show_filters(void *optctx, const char *opt, const char *arg)
     const AVFilterPad *pad;
 
     printf("Filters:\n"
-           "  T.. = Timeline support\n"
-           "  .S. = Slice threading\n"
-           "  ..C = Commmand support\n"
+           "  T. = Timeline support\n"
+           "  .S = Slice threading\n"
            "  A = Audio input/output\n"
            "  V = Video input/output\n"
            "  N = Dynamic number and/or type of input/output\n"
@@ -1508,28 +1494,15 @@ int show_filters(void *optctx, const char *opt, const char *arg)
                                   ( i && (filter->flags & AVFILTER_FLAG_DYNAMIC_OUTPUTS))) ? 'N' : '|';
         }
         *descr_cur = 0;
-        printf(" %c%c%c %-16s %-10s %s\n",
+        printf(" %c%c %-16s %-10s %s\n",
                filter->flags & AVFILTER_FLAG_SUPPORT_TIMELINE ? 'T' : '.',
                filter->flags & AVFILTER_FLAG_SLICE_THREADS    ? 'S' : '.',
-               filter->process_command                        ? 'C' : '.',
                filter->name, descr, filter->description);
     }
 #endif
     return 0;
 }
 
-void show_colors(void *optctx, const char *opt, const char *arg)
-{
-    const char *name;
-    const uint8_t *rgb;
-    int i;
-
-    printf("%-32s #RRGGBB\n", "name");
-
-    for (i = 0; name = av_get_known_color_name(i, &rgb); i++)
-        printf("%-32s #%02x%02x%02x\n", name, rgb[0], rgb[1], rgb[2]);
-}
-
 int show_pix_fmts(void *optctx, const char *opt, const char *arg)
 {
     const AVPixFmtDescriptor *pix_desc = NULL;

cmdutils.h

@@ -81,11 +81,6 @@ void uninit_opts(void);
  */
 void log_callback_help(void* ptr, int level, const char* fmt, va_list vl);
 
-/**
- * Override the cpuflags.
- */
-int opt_cpuflags(void *optctx, const char *opt, const char *arg);
-
 /**
  * Fallback for options that are not explicitly handled, these will be
  * parsed through AVOptions.
@@ -101,6 +96,8 @@ int opt_report(const char *opt);
 
 int opt_max_alloc(void *optctx, const char *opt, const char *arg);
 
+int opt_cpuflags(void *optctx, const char *opt, const char *arg);
+
 int opt_codec_debug(void *optctx, const char *opt, const char *arg);
 
 int opt_opencl(void *optctx, const char *opt, const char *arg);
@@ -488,12 +485,6 @@ int show_layouts(void *optctx, const char *opt, const char *arg);
  */
 int show_sample_fmts(void *optctx, const char *opt, const char *arg);
 
-/**
- * Print a listing containing all the color names and values recognized
- * by the program.
- */
-void show_colors(void *optctx, const char *opt, const char *arg);
-
 /**
  * Return a positive value if a line read from standard input
  * starts with [yY], otherwise return 0.
@@ -507,7 +498,7 @@ int read_yesno(void);
  * @param filename file to read from
  * @param bufptr location where pointer to buffer is returned
  * @param size   location where size of buffer is returned
- * @return >= 0 in case of success, a negative value corresponding to an
+ * @return 0 in case of success, a negative value corresponding to an
  * AVERROR error code in case of failure.
  */
 int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);

cmdutils_common_opts.h

@@ -14,12 +14,11 @@
     { "pix_fmts"   , OPT_EXIT, {.func_arg = show_pix_fmts },    "show available pixel formats" },
     { "layouts"    , OPT_EXIT, {.func_arg = show_layouts  },    "show standard channel layouts" },
     { "sample_fmts", OPT_EXIT, {.func_arg = show_sample_fmts }, "show available audio sample formats" },
-    { "colors"     , OPT_EXIT, {.func_arg = show_colors },      "show available color names" },
     { "loglevel"   , HAS_ARG,  {.func_arg = opt_loglevel},      "set logging level", "loglevel" },
     { "v",           HAS_ARG,  {.func_arg = opt_loglevel},      "set logging level", "loglevel" },
     { "report"     , 0,        {(void*)opt_report}, "generate a report" },
     { "max_alloc"  , HAS_ARG,  {.func_arg = opt_max_alloc},     "set maximum size of a single allocated block", "bytes" },
-    { "cpuflags"   , HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags }, "force specific cpu flags", "flags" },
+    { "cpuflags"   , HAS_ARG | OPT_EXPERT, {.func_arg = opt_cpuflags}, "force specific cpu flags", "flags" },
 #if CONFIG_OPENCL
     { "opencl_options", HAS_ARG, {.func_arg = opt_opencl},      "set OpenCL environment options" },
 #endif

common.mak

@@ -99,8 +99,7 @@ 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)))
+DEP_LIBS := $(foreach NAME,$(FFLIBS),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
 
 SRC_DIR    := $(SRC_PATH)/lib$(NAME)
 ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))

compat/aix/math.h

@@ -1,26 +1,9 @@
 /*
- * Work around the class() function in AIX math.h clashing with
- * identifiers named "class".
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ * Workaround aix-specific class() function clashing with ffmpeg class usage
  */
 
-#ifndef FFMPEG_COMPAT_AIX_MATH_H
-#define FFMPEG_COMPAT_AIX_MATH_H
+#ifndef COMPAT_AIX_MATH_H
+#define COMPAT_AIX_MATH_H
 
 #define class class_in_math_h_causes_problems
 
@@ -28,4 +11,4 @@
 
 #undef class
 
-#endif /* FFMPEG_COMPAT_AIX_MATH_H */
+#endif /* COMPAT_AIX_MATH_H */

compat/avisynth/avisynth_c_25.h

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

compat/tms470/math.h

@@ -1,24 +1,3 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef FFMPEG_COMPAT_TMS470_MATH_H
-#define FFMPEG_COMPAT_TMS470_MATH_H
-
 #include_next <math.h>
 
 #undef INFINITY
@@ -26,5 +5,3 @@
 
 #define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 }))
 #define NAN      (*(const float*)((const unsigned []){ 0x7fc00000 }))
-
-#endif /* FFMPEG_COMPAT_TMS470_MATH_H */

compat/w32pthreads.h

@@ -26,8 +26,8 @@
  * w32threads to pthreads wrapper
  */
 
-#ifndef FFMPEG_COMPAT_W32PTHREADS_H
-#define FFMPEG_COMPAT_W32PTHREADS_H
+#ifndef LIBAV_W32PTHREADS_H
+#define LIBAV_W32PTHREADS_H
 
 /* Build up a pthread-like API using underlying Windows API. Have only static
  * methods so as to not conflict with a potentially linked in pthread-win32
@@ -62,18 +62,11 @@ typedef struct pthread_cond_t {
 } pthread_cond_t;
 
 /* function pointers to conditional variable API on windows 6.0+ kernels */
-#if _WIN32_WINNT < 0x0600
 static void (WINAPI *cond_broadcast)(pthread_cond_t *cond);
 static void (WINAPI *cond_init)(pthread_cond_t *cond);
 static void (WINAPI *cond_signal)(pthread_cond_t *cond);
 static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
                                 DWORD milliseconds);
-#else
-#define cond_init      InitializeConditionVariable
-#define cond_broadcast WakeAllConditionVariable
-#define cond_signal    WakeConditionVariable
-#define cond_wait      SleepConditionVariableCS
-#endif
 
 static unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
 {
@@ -275,8 +268,13 @@ static void w32thread_init(void)
         (void*)GetProcAddress(kernel_dll, "WakeConditionVariable");
     cond_wait      =
         (void*)GetProcAddress(kernel_dll, "SleepConditionVariableCS");
+#else
+    cond_init      = InitializeConditionVariable;
+    cond_broadcast = WakeAllConditionVariable;
+    cond_signal    = WakeConditionVariable;
+    cond_wait      = SleepConditionVariableCS;
 #endif
 
 }
 
-#endif /* FFMPEG_COMPAT_W32PTHREADS_H */
+#endif /* LIBAV_W32PTHREADS_H */

configure

@@ -82,7 +82,6 @@ Standard options:
   --prefix=PREFIX          install in PREFIX [$prefix]
   --bindir=DIR             install binaries in DIR [PREFIX/bin]
   --datadir=DIR            install data files in DIR [PREFIX/share/ffmpeg]
-  --docdir=DIR             install documentation in DIR [PREFIX/share/doc/ffmpeg]
   --libdir=DIR             install libs in DIR [PREFIX/lib]
   --shlibdir=DIR           install shared libs in DIR [PREFIX/lib]
   --incdir=DIR             install includes in DIR [PREFIX/include]
@@ -105,7 +104,6 @@ Configuration options:
   --disable-all            disable building components, libraries and programs
   --enable-incompatible-libav-abi enable incompatible Libav fork ABI [no]
   --enable-incompatible-fork-abi  enable incompatible Libav fork ABI (deprecated) [no]
-  --enable-raise-major     increase major version numbers in sonames [no]
 
 Program options:
   --disable-programs       do not build command line programs
@@ -188,13 +186,12 @@ Individual component options:
   --disable-filters        disable all filters
 
 External library support:
-  --enable-avisynth        enable reading of AviSynth script files [no]
+  --enable-avisynth        enable reading of AVISynth script files [no]
   --disable-bzlib          disable bzlib [autodetect]
   --enable-fontconfig      enable fontconfig
   --enable-frei0r          enable frei0r video filtering
   --enable-gnutls          enable gnutls [no]
   --disable-iconv          disable iconv [autodetect]
-  --enable-ladspa          enable LADSPA audio filtering
   --enable-libaacplus      enable AAC+ encoding via libaacplus [no]
   --enable-libass          enable libass subtitles rendering [no]
   --enable-libbluray       enable BluRay reading using libbluray [no]
@@ -204,7 +201,7 @@ External library support:
   --enable-libdc1394       enable IIDC-1394 grabbing using libdc1394
                            and libraw1394 [no]
   --enable-libfaac         enable AAC encoding via libfaac [no]
-  --enable-libfdk-aac      enable AAC de/encoding via libfdk-aac [no]
+  --enable-libfdk-aac      enable AAC encoding via libfdk-aac [no]
   --enable-libflite        enable flite (voice synthesis) support via libflite [no]
   --enable-libfreetype     enable libfreetype [no]
   --enable-libgme          enable Game Music Emu via libgme [no]
@@ -227,7 +224,6 @@ External library support:
   --enable-libshine        enable fixed-point MP3 encoding via libshine [no]
   --enable-libsoxr         enable Include libsoxr resampling [no]
   --enable-libspeex        enable Speex de/encoding via libspeex [no]
-  --enable-libssh          enable SFTP protocol via libssh [no]
   --enable-libstagefright-h264  enable H.264 decoding via libstagefright [no]
   --enable-libtheora       enable Theora encoding via libtheora [no]
   --enable-libtwolame      enable MP2 encoding via libtwolame [no]
@@ -245,7 +241,6 @@ External library support:
   --enable-libxvid         enable Xvid encoding via xvidcore,
                            native MPEG-4/Xvid encoder exists [no]
   --enable-libzmq          enable message passing via libzmq [no]
-  --enable-libzvbi         enable teletext support via libzvbi [no]
   --enable-openal          enable OpenAL 1.1 capture support [no]
   --enable-opencl          enable OpenCL code
   --enable-openssl         enable openssl [no]
@@ -261,7 +256,6 @@ Advanced options (experts only):
   --target-exec=CMD        command to run executables on target
   --target-path=DIR        path to view of build directory on target
   --target-samples=DIR     path to samples directory on target
-  --tempprefix=PATH        force fixed dir/prefix instead of mktemp for checks
   --toolchain=NAME         set tool defaults according to NAME
   --nm=NM                  use nm tool NM [$nm_default]
   --ar=AR                  use archive tool AR [$ar_default]
@@ -316,7 +310,6 @@ Optimization options (experts only):
   --disable-sse42          disable SSE4.2 optimizations
   --disable-avx            disable AVX optimizations
   --disable-fma4           disable FMA4 optimizations
-  --disable-avx2           disable AVX2 optimizations
   --disable-armv5te        disable armv5te optimizations
   --disable-armv6          disable armv6 optimizations
   --disable-armv6t2        disable armv6t2 optimizations
@@ -426,7 +419,7 @@ sh_quote(){
 }
 
 cleanws(){
-    echo "$@" | sed 's/^ *//;s/  */ /g;s/ *$//;s/\r//g'
+    echo "$@" | sed 's/^ *//;s/  */ /g;s/ *$//'
 }
 
 filter(){
@@ -729,10 +722,6 @@ add_ldflags(){
     append LDFLAGS $($ldflags_filter "$@")
 }
 
-add_stripflags(){
-    append ASMSTRIPFLAGS "$@"
-}
-
 add_extralibs(){
     prepend extralibs $($ldflags_filter "$@")
 }
@@ -890,20 +879,6 @@ check_ldflags(){
     test_ldflags "$@" && add_ldflags "$@"
 }
 
-test_stripflags(){
-    log test_stripflags "$@"
-    # call check_cc to get a fresh TMPO
-    check_cc <<EOF
-int main(void) { return 0; }
-EOF
-    check_cmd $strip $ASMSTRIPFLAGS "$@" $TMPO
-}
-
-check_stripflags(){
-    log check_stripflags "$@"
-    test_stripflags "$@" && add_stripflags "$@"
-}
-
 check_header(){
     log check_header "$@"
     header=$1
@@ -1018,7 +993,7 @@ check_pkg_config(){
     headers="$2"
     funcs="$3"
     shift 3
-    check_cmd $pkg_config --exists --print-errors $pkg || return
+    $pkg_config --exists $pkg 2>/dev/null || return
     pkg_cflags=$($pkg_config --cflags $pkg)
     pkg_libs=$($pkg_config --libs $pkg)
     check_func_headers "$headers" "$funcs" $pkg_cflags $pkg_libs "$@" &&
@@ -1121,26 +1096,6 @@ require_pkg_config(){
     add_extralibs $(get_safe ${pkg}_libs)
 }
 
-require_libfreetype(){
-    log require_libfreetype "$@"
-    pkg="freetype2"
-    check_cmd $pkg_config --exists --print-errors $pkg \
-      || die "ERROR: $pkg not found"
-    pkg_cflags=$($pkg_config --cflags $pkg)
-    pkg_libs=$($pkg_config --libs $pkg)
-    {
-        echo "#include <ft2build.h>"
-        echo "#include FT_FREETYPE_H"
-        echo "long check_func(void) { return (long) FT_Init_FreeType; }"
-        echo "int main(void) { return 0; }"
-    } | check_ld "cc" $pkg_cflags $pkg_libs \
-      && set_safe ${pkg}_cflags $pkg_cflags \
-      && set_safe ${pkg}_libs   $pkg_libs \
-      || die "ERROR: $pkg not found"
-    add_cflags    $(get_safe ${pkg}_cflags)
-    add_extralibs $(get_safe ${pkg}_libs)
-}
-
 hostcc_o(){
     eval printf '%s\\n' $HOSTCC_O
 }
@@ -1204,7 +1159,6 @@ EXTERNAL_LIBRARY_LIST="
     frei0r
     gnutls
     iconv
-    ladspa
     libaacplus
     libass
     libbluray
@@ -1235,7 +1189,6 @@ EXTERNAL_LIBRARY_LIST="
     libshine
     libsoxr
     libspeex
-    libssh
     libstagefright_h264
     libtheora
     libtwolame
@@ -1251,7 +1204,6 @@ EXTERNAL_LIBRARY_LIST="
     libxavs
     libxvid
     libzmq
-    libzvbi
     openal
     opencl
     openssl
@@ -1319,7 +1271,6 @@ CONFIG_LIST="
     network
     nonfree
     pic
-    raise_major
     rdft
     runtime_cpudetect
     safe_bitstream_reader
@@ -1385,9 +1336,7 @@ ARCH_EXT_LIST_X86='
     amd3dnow
     amd3dnowext
     avx
-    avx2
     fma4
-    i686
     mmx
     mmxext
     sse
@@ -1512,10 +1461,10 @@ HAVE_LIST="
     glob
     gnu_as
     gsm_h
+    i686
     ibm_asm
     inet_aton
     io_h
-    inline_asm_labels
     isatty
     jack_port_get_latency_range
     kbhit
@@ -1547,11 +1496,9 @@ HAVE_LIST="
     pod2man
     poll_h
     posix_memalign
-    pragma_deprecated
     pthread_cancel
     rdtsc
     rsync_contimeout
-    sarestart
     sched_getaffinity
     sdl
     SetConsoleTextAttribute
@@ -1585,7 +1532,6 @@ HAVE_LIST="
     sys_select_h
     sys_soundcard_h
     sys_time_h
-    sys_un_h
     sys_videoio_h
     termios_h
     texi2html
@@ -1606,7 +1552,6 @@ CONFIG_EXTRA="
     ac3dsp
     audio_frame_queue
     dsputil
-    exif
     frame_thread_encoder
     gcrypt
     golomb
@@ -1625,8 +1570,6 @@ CONFIG_EXTRA="
     mpegvideoenc
     nettle
     rangecoder
-    riffdec
-    riffenc
     rtpdec
     rtpenc_chain
     sinewin
@@ -1652,7 +1595,6 @@ CMDLINE_SELECT="
 PATHS_LIST='
     bindir
     datadir
-    docdir
     incdir
     libdir
     mandir
@@ -1696,7 +1638,6 @@ CMDLINE_SET="
     target_os
     target_path
     target_samples
-    tempprefix
     toolchain
     valgrind
     yasmexe
@@ -1731,9 +1672,7 @@ ppc4xx_deps="ppc"
 
 vis_deps="sparc"
 
-cpunop_deps="i686"
-x86_64_select="i686"
-x86_64_suggest="fast_cmov"
+x86_64_suggest="fast_cmov i686"
 
 amd3dnow_deps="mmx"
 amd3dnowext_deps="amd3dnow"
@@ -1747,7 +1686,6 @@ sse4_deps="ssse3"
 sse42_deps="sse4"
 avx_deps="sse42"
 fma4_deps="avx"
-avx2_deps="avx"
 
 mmx_external_deps="yasm"
 mmx_inline_deps="inline_asm"
@@ -1796,7 +1734,7 @@ alac_encoder_select="lpc"
 als_decoder_select="dsputil"
 amrnb_decoder_select="lsp"
 amrwb_decoder_select="lsp"
-amv_decoder_select="dsputil hpeldsp exif"
+amv_decoder_select="dsputil hpeldsp"
 amv_encoder_select="aandcttables"
 ape_decoder_select="dsputil"
 asv1_decoder_select="dsputil"
@@ -1805,7 +1743,6 @@ asv2_decoder_select="dsputil"
 asv2_encoder_select="dsputil"
 atrac1_decoder_select="mdct sinewin"
 atrac3_decoder_select="mdct"
-avrn_decoder_select="exif"
 bink_decoder_select="dsputil hpeldsp"
 binkaudio_dct_decoder_select="mdct rdft dct sinewin"
 binkaudio_rdft_decoder_select="mdct rdft sinewin"
@@ -1852,24 +1789,22 @@ h263i_decoder_select="h263_decoder"
 h263p_encoder_select="h263_encoder"
 h264_decoder_select="golomb h264chroma h264dsp h264pred h264qpel videodsp"
 h264_decoder_suggest="error_resilience"
-hevc_decoder_select="dsputil golomb videodsp"
 huffyuv_decoder_select="dsputil"
 huffyuv_encoder_select="dsputil huffman"
 iac_decoder_select="dsputil fft mdct sinewin"
 imc_decoder_select="dsputil fft mdct sinewin"
 indeo3_decoder_select="hpeldsp"
 interplay_video_decoder_select="hpeldsp"
-jpegls_decoder_select="dsputil golomb hpeldsp exif"
+jpegls_decoder_select="dsputil golomb hpeldsp"
 jpegls_encoder_select="golomb"
 jv_decoder_select="dsputil"
 lagarith_decoder_select="dsputil"
 ljpeg_encoder_select="aandcttables mpegvideoenc"
 loco_decoder_select="golomb"
 mdec_decoder_select="dsputil error_resilience mpegvideo"
-metasound_decoder_select="lsp mdct sinewin"
 mimic_decoder_select="dsputil hpeldsp"
-mjpeg_decoder_select="dsputil hpeldsp exif"
-mjpegb_decoder_select="dsputil hpeldsp exif"
+mjpeg_decoder_select="dsputil hpeldsp"
+mjpegb_decoder_select="dsputil hpeldsp"
 mjpeg_encoder_select="aandcttables dsputil mpegvideoenc"
 mlp_decoder_select="dsputil mlp_parser"
 motionpixels_decoder_select="dsputil"
@@ -1899,7 +1834,7 @@ msmpeg4v2_encoder_select="h263_encoder"
 msmpeg4v3_decoder_select="h263_decoder"
 msmpeg4v3_encoder_select="h263_encoder"
 mss2_decoder_select="error_resilience vc1_decoder"
-mxpeg_decoder_select="dsputil hpeldsp exif"
+mxpeg_decoder_select="dsputil hpeldsp"
 nellymoser_decoder_select="mdct sinewin"
 nellymoser_encoder_select="audio_frame_queue mdct sinewin"
 nuv_decoder_select="dsputil lzo"
@@ -1925,7 +1860,7 @@ snow_encoder_select="aandcttables dsputil dwt h264qpel hpeldsp mpegvideoenc rang
 sonic_decoder_select="golomb"
 sonic_encoder_select="golomb"
 sonic_ls_encoder_select="golomb"
-sp5x_decoder_select="dsputil hpeldsp exif"
+sp5x_decoder_select="dsputil hpeldsp"
 svq1_decoder_select="hpeldsp"
 svq1_encoder_select="aandcttables dsputil hpeldsp mpegvideoenc"
 svq3_decoder_select="golomb h264chroma h264dsp h264pred h264qpel hpeldsp mpegvideo videodsp"
@@ -1934,7 +1869,7 @@ tak_decoder_select="dsputil"
 theora_decoder_select="vp3_decoder"
 tiff_decoder_suggest="zlib"
 tiff_encoder_suggest="zlib"
-thp_decoder_select="dsputil hpeldsp exif"
+thp_decoder_select="dsputil hpeldsp"
 truehd_decoder_select="mlp_parser"
 truemotion2_decoder_select="dsputil"
 truespeech_decoder_select="dsputil"
@@ -1953,7 +1888,6 @@ vp6_decoder_select="h264chroma hpeldsp huffman videodsp vp3dsp"
 vp6a_decoder_select="vp6_decoder"
 vp6f_decoder_select="vp6_decoder"
 vp8_decoder_select="h264pred videodsp"
-vp9_decoder_select="videodsp"
 webp_decoder_select="vp8_decoder"
 wmapro_decoder_select="mdct sinewin"
 wmav1_decoder_select="mdct sinewin"
@@ -2036,7 +1970,6 @@ wmv3_vdpau_hwaccel_select="vc1_vdpau_hwaccel"
 
 # parsers
 h264_parser_select="golomb h264chroma h264dsp h264pred h264qpel videodsp"
-hevc_parser_select="hevc_decoder"
 mpeg4video_parser_select="error_resilience mpegvideo"
 mpegvideo_parser_select="error_resilience mpegvideo"
 vc1_parser_select="mpegvideo"
@@ -2046,7 +1979,6 @@ libaacplus_encoder_deps="libaacplus"
 libcelt_decoder_deps="libcelt"
 libfaac_encoder_deps="libfaac"
 libfaac_encoder_select="audio_frame_queue"
-libfdk_aac_decoder_deps="libfdk_aac"
 libfdk_aac_encoder_deps="libfdk_aac"
 libfdk_aac_encoder_select="audio_frame_queue"
 libgme_demuxer_deps="libgme"
@@ -2096,45 +2028,30 @@ libxavs_encoder_deps="libxavs"
 libxvid_encoder_deps="libxvid"
 libutvideo_decoder_deps="libutvideo"
 libutvideo_encoder_deps="libutvideo"
-libzvbi_teletext_decoder_deps="libzvbi"
 
 # demuxers / muxers
 ac3_demuxer_select="ac3_parser"
-asf_demuxer_select="riffdec"
-asf_muxer_select="riffenc"
 asf_stream_muxer_select="asf_muxer"
-avi_demuxer_select="riffdec"
-avi_muxer_select="riffenc"
 avisynth_demuxer_deps="avisynth"
-avisynth_demuxer_select="riffdec"
-caf_demuxer_select="riffdec"
 dirac_demuxer_select="dirac_parser"
 dts_demuxer_select="dca_parser"
 dtshd_demuxer_select="dca_parser"
-dxa_demuxer_select="riffdec"
 eac3_demuxer_select="ac3_parser"
 f4v_muxer_select="mov_muxer"
 flac_demuxer_select="flac_parser"
-hls_muxer_select="mpegts_muxer"
 ipod_muxer_select="mov_muxer"
 ismv_muxer_select="mov_muxer"
 libnut_demuxer_deps="libnut"
 libnut_muxer_deps="libnut"
 matroska_audio_muxer_select="matroska_muxer"
-matroska_demuxer_select="riffdec"
 matroska_demuxer_suggest="bzlib lzo zlib"
-matroska_muxer_select="riffenc"
-mmf_muxer_select="riffenc"
-mov_demuxer_select="riffdec"
 mov_demuxer_suggest="zlib"
-mov_muxer_select="riffenc rtpenc_chain"
+mov_muxer_select="rtpenc_chain"
 mp3_demuxer_select="mpegaudio_parser"
 mp4_muxer_select="mov_muxer"
 mpegts_muxer_select="adts_muxer latm_muxer"
 mpegtsraw_demuxer_select="mpegts_demuxer"
 mxf_d10_muxer_select="mxf_muxer"
-nut_muxer_select="riffenc"
-nuv_demuxer_select="riffdec"
 ogg_demuxer_select="golomb"
 psp_muxer_select="mov_muxer"
 rtp_demuxer_select="sdp_demuxer"
@@ -2150,14 +2067,8 @@ tak_demuxer_select="tak_parser"
 tg2_muxer_select="mov_muxer"
 tgp_muxer_select="mov_muxer"
 vobsub_demuxer_select="mpegps_demuxer"
-w64_demuxer_select="wav_demuxer"
-w64_muxer_select="wav_muxer"
-wav_demuxer_select="riffdec"
-wav_muxer_select="riffenc"
-webm_muxer_select="riffenc"
-wtv_demuxer_select="riffdec"
-xmv_demuxer_select="riffdec"
-xwma_demuxer_select="riffdec"
+w64_demuxer_deps="wav_demuxer"
+w64_muxer_deps="wav_muxer"
 
 # indevs / outdevs
 alsa_indev_deps="alsa_asoundlib_h snd_pcm_htimestamp"
@@ -2166,10 +2077,8 @@ bktr_indev_deps_any="dev_bktr_ioctl_bt848_h machine_ioctl_bt848_h dev_video_bktr
 caca_outdev_deps="libcaca"
 dshow_indev_deps="IBaseFilter"
 dshow_indev_extralibs="-lpsapi -lole32 -lstrmiids -luuid"
-dv1394_indev_deps="dv1394"
-dv1394_indev_select="dv_demuxer"
+dv1394_indev_deps="dv1394 dv_demuxer"
 fbdev_indev_deps="linux_fb_h"
-fbdev_outdev_deps="linux_fb_h"
 iec61883_indev_deps="libiec61883"
 jack_indev_deps="jack_jack_h sem_timedwait"
 lavfi_indev_deps="avfilter"
@@ -2180,7 +2089,6 @@ openal_indev_deps="openal"
 oss_indev_deps_any="soundcard_h sys_soundcard_h"
 oss_outdev_deps_any="soundcard_h sys_soundcard_h"
 pulse_indev_deps="libpulse"
-pulse_outdev_deps="libpulse"
 sdl_outdev_deps="sdl"
 sndio_indev_deps="sndio_h"
 sndio_outdev_deps="sndio_h"
@@ -2210,7 +2118,6 @@ librtmpe_protocol_deps="librtmp"
 librtmps_protocol_deps="librtmp"
 librtmpt_protocol_deps="librtmp"
 librtmpte_protocol_deps="librtmp"
-libssh_protocol_deps="libssh"
 mmsh_protocol_select="http_protocol"
 mmst_protocol_select="network"
 rtmp_protocol_deps="!librtmp_protocol"
@@ -2229,8 +2136,6 @@ tcp_protocol_select="network"
 tls_protocol_deps_any="openssl gnutls"
 tls_protocol_select="tcp_protocol"
 udp_protocol_select="network"
-unix_protocol_deps="sys_un_h"
-unix_protocol_select="network"
 
 # filters
 aconvert_filter_deps="swresample"
@@ -2238,15 +2143,13 @@ amovie_filter_deps="avcodec avformat"
 aresample_filter_deps="swresample"
 ass_filter_deps="libass"
 asyncts_filter_deps="avresample"
-atempo_filter_deps="avcodec"
-atempo_filter_select="rdft"
+atempo_filter_deps="avcodec rdft"
 azmq_filter_deps="libzmq"
 blackframe_filter_deps="gpl"
 boxblur_filter_deps="gpl"
 colormatrix_filter_deps="gpl"
 cropdetect_filter_deps="gpl"
-dctdnoiz_filter_deps="avcodec"
-dctdnoiz_filter_select="fft"
+dctdnoiz_filter_deps="avcodec fft"
 delogo_filter_deps="gpl"
 deshake_filter_deps="avcodec"
 deshake_filter_select="dsputil"
@@ -2260,31 +2163,27 @@ frei0r_src_filter_extralibs='$ldl'
 geq_filter_deps="gpl"
 histeq_filter_deps="gpl"
 hqdn3d_filter_deps="gpl"
+hue_filter_deps="gpl"
 interlace_filter_deps="gpl"
 kerndeint_filter_deps="gpl"
-ladspa_filter_deps="ladspa dlopen"
 mcdeint_filter_deps="avcodec gpl"
 movie_filter_deps="avcodec avformat"
 mp_filter_deps="gpl avcodec swscale inline_asm"
 mpdecimate_filter_deps="gpl avcodec"
 mptestsrc_filter_deps="gpl"
 negate_filter_deps="lut_filter"
-perspective_filter_deps="gpl"
+noise_filter_deps="gpl"
 resample_filter_deps="avresample"
 ocv_filter_deps="libopencv"
 owdenoise_filter_deps="gpl"
 pan_filter_deps="swresample"
-phase_filter_deps="gpl"
 pp_filter_deps="gpl postproc"
-pullup_filter_deps="gpl"
 removelogo_filter_deps="avcodec avformat swscale"
 sab_filter_deps="gpl swscale"
 scale_filter_deps="swscale"
 smartblur_filter_deps="gpl swscale"
-showspectrum_filter_deps="avcodec"
-showspectrum_filter_select="rdft"
-spp_filter_deps="gpl avcodec"
-spp_filter_select="fft"
+showspectrum_filter_deps="avcodec rdft"
+spp_filter_deps="gpl avcodec fft"
 stereo3d_filter_deps="gpl"
 subtitles_filter_deps="avformat avcodec libass"
 super2xsai_filter_deps="gpl"
@@ -2307,15 +2206,14 @@ postproc_deps="avutil gpl"
 swscale_deps="avutil"
 
 # programs
-ffmpeg_deps="avcodec avfilter avformat swresample"
+ffmpeg_deps="avcodec avfilter avformat swscale swresample"
 ffmpeg_select="aformat_filter anull_filter atrim_filter format_filter
                null_filter
                setpts_filter trim_filter"
 ffplay_deps="avcodec avformat swscale swresample sdl"
 ffplay_select="rdft crop_filter"
 ffprobe_deps="avcodec avformat"
-ffserver_deps="avformat fork sarestart"
-ffserver_select="ffm_muxer rtp_protocol rtsp_demuxer"
+ffserver_deps="avformat ffm_muxer fork rtp_protocol rtsp_demuxer"
 ffserver_extralibs='$ldl'
 
 # documentation
@@ -2333,7 +2231,6 @@ logfile="config.log"
 prefix_default="/usr/local"
 bindir_default='${prefix}/bin'
 datadir_default='${prefix}/share/ffmpeg'
-docdir_default='${prefix}/share/doc/ffmpeg'
 incdir_default='${prefix}/include'
 libdir_default='${prefix}/lib'
 mandir_default='${prefix}/share/man'
@@ -2384,6 +2281,7 @@ enable dxva2 vaapi vdpau
 
 # build settings
 SHFLAGS='-shared -Wl,-soname,$$(@F)'
+FFSERVERLDFLAGS=-Wl,-E
 LIBPREF="lib"
 LIBSUF=".a"
 FULLNAME='$(NAME)$(BUILDSUF)'
@@ -2702,11 +2600,7 @@ HOSTEXESUF=$(exesuf $host_os)
 : ${TMPDIR:=$TMP}
 : ${TMPDIR:=/tmp}
 
-if [ -n "$tempprefix" ] ; then
-    mktemp(){
-        echo $tempprefix.${HOSTNAME}.${UID}
-    }
-elif ! check_cmd mktemp -u XXXXXX; then
+if ! check_cmd mktemp -u XXXXXX; then
     # simple replacement for missing mktemp
     # NOT SAFE FOR GENERAL USE
     mktemp(){
@@ -2769,18 +2663,6 @@ ccc_flags(){
    done
 }
 
-cparser_flags(){
-    for flag; do
-        case $flag in
-            -Wno-switch)             echo -Wno-switch-enum ;;
-            -Wno-format-zero-length) ;;
-            -Wdisabled-optimization) ;;
-            -Wno-pointer-sign)       echo -Wno-other ;;
-            *)                       echo $flag ;;
-        esac
-    done
-}
-
 msvc_common_flags(){
     for flag; do
         case $flag in
@@ -2815,7 +2697,7 @@ msvc_flags(){
                                        -wd4146 -wd4057 -wd4204 -wd4706 -wd4305 \
                                        -wd4152 -wd4324 -we4013 -wd4100 -wd4214 \
                                        -wd4554 \
-                                       -wd4273 ;;
+                                       -wd4996 -wd4273 ;;
         esac
     done
 }
@@ -2919,9 +2801,7 @@ probe_cc(){
     unset _depflags _DEPCMD _DEPFLAGS
     _flags_filter=echo
 
-    if $_cc --version 2>&1 | grep -q '^GNU assembler'; then
-        true # no-op to avoid reading stdin in following checks
-    elif $_cc -v 2>&1 | grep -q '^gcc.*LLVM'; then
+    if $_cc -v 2>&1 | grep -q '^gcc.*LLVM'; then
         _type=llvm_gcc
         gcc_extra_ver=$(expr "$($_cc --version | head -n1)" : '.*\((.*)\)')
         _ident="llvm-gcc $($_cc -dumpversion) $gcc_extra_ver"
@@ -3042,7 +2922,7 @@ probe_cc(){
         _ld_lib='lib%.a'
         _ld_path='-libpath:'
         _flags='-nologo'
-        _cflags='-D_USE_MATH_DEFINES -D_CRT_SECURE_NO_WARNINGS -Dinline=__inline -FIstdlib.h -Dstrtoll=_strtoi64'
+        _cflags='-D_USE_MATH_DEFINES -Dinline=__inline -FIstdlib.h -Dstrtoll=_strtoi64'
         if [ $pfx = hostcc ]; then
             append _cflags -Dsnprintf=_snprintf
         fi
@@ -3075,13 +2955,6 @@ probe_cc(){
         if [ $pfx = hostcc ]; then
             append _cflags -Dsnprintf=_snprintf
         fi
-    elif $_cc --version 2>/dev/null | grep -q ^cparser; then
-        _type=cparser
-        _ident=$($_cc --version | head -n1)
-        _depflags='-MMD'
-        _cflags_speed='-O4'
-        _cflags_size='-O2'
-        _flags_filter=cparser_flags
     fi
 
     eval ${pfx}_type=\$_type
@@ -3156,9 +3029,6 @@ if $ar 2>&1 | grep -q Microsoft; then
 elif $ar 2>&1 | grep -q 'Texas Instruments'; then
     arflags="rq"
     ar_o='$@'
-elif $ar 2>&1 | grep -q 'Usage: ar.*-X.*any'; then
-    arflags='-Xany -r -c'
-    ar_o='$@'
 else
     arflags="rc"
     ar_o='$@'
@@ -3208,7 +3078,7 @@ case "$arch" in
     aarch64|arm64)
         arch="aarch64"
     ;;
-    arm*|iPad*|iPhone*)
+    arm*|iPad*)
         arch="arm"
     ;;
     mips*|IP*)
@@ -3531,9 +3401,7 @@ enabled spic && enable_weak pic
 # OS specific
 case $target_os in
     aix)
-        SHFLAGS=-shared
         add_cppflags '-I\$(SRC_PATH)/compat/aix'
-        enabled shared && add_ldflags -Wl,-brtl
         ;;
     haiku)
         prefix_default="/boot/common"
@@ -3541,6 +3409,7 @@ case $target_os in
         host_libs=
         ;;
     sunos)
+        FFSERVERLDFLAGS=""
         SHFLAGS='-shared -Wl,-h,$$(@F)'
         enabled x86 && SHFLAGS="-mimpure-text $SHFLAGS"
         network_extralibs="-lsocket -lnsl"
@@ -3588,6 +3457,7 @@ case $target_os in
         SLIBSUF=".dylib"
         SLIBNAME_WITH_VERSION='$(SLIBPREF)$(FULLNAME).$(LIBVERSION)$(SLIBSUF)'
         SLIBNAME_WITH_MAJOR='$(SLIBPREF)$(FULLNAME).$(LIBMAJOR)$(SLIBSUF)'
+        FFSERVERLDFLAGS=-Wl,-bind_at_load
         objformat="macho"
         enabled x86_64 && objformat="macho64"
         enabled_any pic shared ||
@@ -3635,7 +3505,7 @@ case $target_os in
             # Link to the import library instead of the normal static library
             # for shared libs.
             LD_LIB='%.lib'
-            # Cannot build both shared and static libs with MSVC or icl.
+            # Cannot build shared and static libs at the same time with MSVC.
             disable static
         fi
         shlibdir_default="$bindir_default"
@@ -3687,6 +3557,7 @@ case $target_os in
         add_cppflags -D_GNU_SOURCE
         add_ldflags -Zomf -Zbin-files -Zargs-wild -Zmap
         SHFLAGS='$(SUBDIR)$(NAME).def -Zdll -Zomf'
+        FFSERVERLDFLAGS=""
         LIBSUF="_s.a"
         SLIBPREF=""
         SLIBSUF=".dll"
@@ -3727,6 +3598,7 @@ case $target_os in
         ;;
     osf1)
         add_cppflags -D_OSF_SOURCE -D_POSIX_PII -D_REENTRANT
+        FFSERVERLDFLAGS=
         ;;
     minix)
         ;;
@@ -3853,7 +3725,6 @@ die_license_disabled gpl libvidstab
 die_license_disabled gpl libx264
 die_license_disabled gpl libxavs
 die_license_disabled gpl libxvid
-die_license_disabled gpl libzvbi
 die_license_disabled gpl x11grab
 
 die_license_disabled nonfree libaacplus
@@ -3903,10 +3774,6 @@ void foo(char * $restrict_keyword p);
 EOF
 done
 
-check_cc <<EOF && enable pragma_deprecated
-void foo(void) { _Pragma("GCC diagnostic ignored \"-Wdeprecated-declarations\"") }
-EOF
-
 check_cc <<EOF && enable attribute_packed
 struct { int x; } __attribute__((packed)) x;
 EOF
@@ -3920,8 +3787,6 @@ unsigned int endian = 'B' << 24 | 'I' << 16 | 'G' << 8 | 'E';
 EOF
 od -t x1 $TMPO | grep -q '42 *49 *47 *45' && enable bigendian
 
-check_inline_asm inline_asm_labels '"1:\n"'
-
 if enabled alpha; then
 
     check_cflags -mieee
@@ -3995,9 +3860,7 @@ elif enabled ppc; then
 
     # AltiVec flags: The FSF version of GCC differs from the Apple version
     if enabled altivec; then
-        if ! enabled_any pic ppc64; then
-            nogas=warn
-        fi
+        nogas=warn
         check_cflags -maltivec -mabi=altivec &&
         { check_header altivec.h && inc_altivec_h="#include <altivec.h>" ; } ||
         check_cflags -faltivec
@@ -4065,11 +3928,11 @@ EOF
             elf*) enabled debug && append YASMFLAGS $yasm_debug ;;
         esac
 
-        check_yasm "movbe ecx, [5]" && enable yasm ||
-            die "yasm/nasm not found or too old. Use --disable-yasm for a crippled build."
-        check_yasm "vextractf128 xmm0, ymm0, 0"      || disable avx_external avresample
+        check_yasm "pextrd [eax], xmm0, 1" && enable yasm ||
+            die "yasm not found, use --disable-yasm for a crippled build"
+        check_yasm "vextractf128 xmm0, ymm0, 0"      || disable avx_external
         check_yasm "vfmaddps ymm0, ymm1, ymm2, ymm3" || disable fma4_external
-        check_yasm "CPU amdnop" && enable cpunop
+        check_yasm "CPU amdnop" && enabled i686 && enable cpunop
     fi
 
     case "$cpu" in
@@ -4165,7 +4028,6 @@ check_builtin sync_val_compare_and_swap "" "int *ptr; int oldval, newval; __sync
 check_builtin machine_rw_barrier mbarrier.h "__machine_rw_barrier()"
 check_builtin atomic_cas_ptr atomic.h "void **ptr; void *oldval, *newval; atomic_cas_ptr(ptr, oldval, newval)"
 check_builtin MemoryBarrier windows.h "MemoryBarrier()"
-check_builtin sarestart signal.h "SA_RESTART"
 check_func  sysconf
 check_func  sysctl
 check_func  usleep
@@ -4200,7 +4062,6 @@ check_header sys/param.h
 check_header sys/resource.h
 check_header sys/select.h
 check_header sys/time.h
-check_header sys/un.h
 check_header termios.h
 check_header unistd.h
 check_header vdpau/vdpau.h
@@ -4270,7 +4131,6 @@ enabled avisynth          && { { check_lib2 "windows.h" LoadLibrary; } ||
 enabled fontconfig        && require_pkg_config fontconfig "fontconfig/fontconfig.h" FcInit
 enabled frei0r            && { check_header frei0r.h || die "ERROR: frei0r.h header not found"; }
 enabled gnutls            && require_pkg_config gnutls gnutls/gnutls.h gnutls_global_init
-enabled ladspa            && { check_header ladspa.h || die "ERROR: ladspa.h header not found"; }
 enabled libiec61883       && require libiec61883 libiec61883/iec61883.h iec61883_cmp_connect -lraw1394 -lavc1394 -lrom1394 -liec61883
 enabled libaacplus        && require "libaacplus >= 2.0.0" aacplus.h aacplusEncOpen -laacplus
 enabled libass            && require_pkg_config libass ass/ass.h ass_library_init
@@ -4283,8 +4143,8 @@ enabled libfaac           && require2 libfaac "stdint.h faac.h" faacEncGetVersio
 enabled libfdk_aac        && require libfdk_aac fdk-aac/aacenc_lib.h aacEncOpen -lfdk-aac
 flite_libs="-lflite_cmu_time_awb -lflite_cmu_us_awb -lflite_cmu_us_kal -lflite_cmu_us_kal16 -lflite_cmu_us_rms -lflite_cmu_us_slt -lflite_usenglish -lflite_cmulex -lflite"
 enabled libflite          && require2 libflite "flite/flite.h" flite_init $flite_libs
-enabled libfreetype       && require_libfreetype
-enabled libgme            && require  libgme gme/gme.h gme_new_emu -lgme -lstdc++
+enabled libfreetype       && require_pkg_config freetype2 "ft2build.h freetype/freetype.h" FT_Init_FreeType
+enabled libgme            && require  libgme gme/gme.h gme_new_emu -lgme
 enabled libgsm            && { for gsm_hdr in "gsm.h" "gsm/gsm.h"; do
                                    check_lib "${gsm_hdr}" gsm_create -lgsm && break;
                                done || die "ERROR: libgsm not found"; }
@@ -4303,9 +4163,8 @@ enabled libpulse          && require_pkg_config libpulse-simple pulse/simple.h p
 enabled libquvi           && require_pkg_config libquvi quvi/quvi.h quvi_init
 enabled librtmp           && require_pkg_config librtmp librtmp/rtmp.h RTMP_Socket
 enabled libschroedinger   && require_pkg_config schroedinger-1.0 schroedinger/schro.h schro_init
-enabled libshine          && require_pkg_config shine shine/layer3.h shine_encode_buffer
+enabled libshine          && require_pkg_config shine shine/layer3.h shine_encode_frame
 enabled libsoxr           && require libsoxr soxr.h soxr_create -lsoxr
-enabled libssh            && require_pkg_config libssh libssh/sftp.h sftp_init
 enabled libspeex          && require libspeex speex/speex.h speex_decoder_init -lspeex
 enabled libstagefright_h264 && require_cpp libstagefright_h264 "binder/ProcessState.h media/stagefright/MetaData.h
     media/stagefright/MediaBufferGroup.h media/stagefright/MediaDebug.h media/stagefright/MediaDefs.h
@@ -4334,7 +4193,6 @@ enabled libx264           && require libx264 x264.h x264_encoder_encode -lx264 &
 enabled libxavs           && require libxavs xavs.h xavs_encoder_encode -lxavs
 enabled libxvid           && require libxvid xvid.h xvid_global -lxvidcore
 enabled libzmq            && require_pkg_config libzmq zmq.h zmq_ctx_new
-enabled libzvbi           && require libzvbi libzvbi.h vbi_decoder_new -lzvbi
 enabled openal            && { { for al_libs in "${OPENAL_LIBS}" "-lopenal" "-lOpenAL32"; do
                                check_lib 'AL/al.h' alGetError "${al_libs}" && break; done } ||
                                die "ERROR: openal not found"; } &&
@@ -4343,7 +4201,7 @@ enabled openal            && { { for al_libs in "${OPENAL_LIBS}" "-lopenal" "-lO
 enabled opencl            && { check_lib2 OpenCL/cl.h clEnqueueNDRangeKernel -Wl,-framework,OpenCL ||
                                check_lib2 CL/cl.h clEnqueueNDRangeKernel -lOpenCL ||
                                die "ERROR: opencl not found"; } &&
-                             { ! enabled_any w32threads os2threads ||
+                             { enabled_any w32threads os2threads &&
                                die "opencl currently needs --enable-pthreads or --disable-w32threads"; } &&
                              { check_cpp_condition "OpenCL/cl.h" "defined(CL_VERSION_1_2)" ||
                                check_cpp_condition "CL/cl.h" "defined(CL_VERSION_1_2)" ||
@@ -4479,10 +4337,6 @@ check_ldflags -Wl,--warn-common
 check_ldflags -Wl,-rpath-link=libpostproc:libswresample:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil:libavresample
 test_ldflags -Wl,-Bsymbolic && append SHFLAGS -Wl,-Bsymbolic
 
-# add some strip flags
-# -wN '..@*' is more selective than -x, but not available everywhere.
-check_stripflags -wN \'..@*\' || check_stripflags -x
-
 enabled xmm_clobber_test &&
     check_ldflags -Wl,--wrap,avcodec_open2              \
                   -Wl,--wrap,avcodec_decode_audio4      \
@@ -4490,7 +4344,6 @@ enabled xmm_clobber_test &&
                   -Wl,--wrap,avcodec_decode_subtitle2   \
                   -Wl,--wrap,avcodec_encode_audio2      \
                   -Wl,--wrap,avcodec_encode_video       \
-                  -Wl,--wrap,avcodec_encode_video2      \
                   -Wl,--wrap,avcodec_encode_subtitle    \
                   -Wl,--wrap,sws_scale ||
     disable xmm_clobber_test
@@ -4584,9 +4437,6 @@ elif enabled clang; then
     check_cflags -Werror=implicit-function-declaration
     check_cflags -Werror=missing-prototypes
     check_cflags -Werror=return-type
-elif enabled cparser; then
-    add_cflags -Wno-missing-variable-declarations
-    add_cflags -Wno-empty-statement
 elif enabled armcc; then
     # 2523: use of inline assembler is deprecated
     add_cflags -W${armcc_opt},--diag_suppress=2523
@@ -4602,16 +4452,11 @@ elif enabled pathscale; then
 elif enabled_any msvc icl; then
     enabled x86_32 && disable aligned_stack
     enabled_all x86_32 debug && add_cflags -Oy-
-    enabled debug && add_ldflags -debug
-    enable pragma_deprecated
     if enabled icl; then
-        # -Qansi-alias is basically -fstrict-aliasing, but does not work
-        # (correctly) on icl 13.x.
-        check_cpp_condition "windows.h" "__ICL < 1300 || __ICL >= 1400" &&
-            add_cflags -Qansi-alias
+        # basically -fstrict-aliasing that does not work (correctly) on icl 13.x
+        check_cpp_condition "windows.h" "__ICL < 1300" && add_cflags -Qansi-alias
         # icl will pass the inline asm tests but inline asm is currently
         # not supported (build will fail)
-        disabled inline_asm || warn "inline asm disabled due to issues with it in ICL"
         disable inline_asm
     fi
 fi
@@ -4629,8 +4474,6 @@ enable frame_thread_encoder
 enabled_any $THREADS_LIST      && enable threads
 enabled_any $ATOMICS_LIST      && enable atomics_native
 
-enabled threads && ! enabled pthreads && ! enabled atomics_native && die "non pthread threading without atomics not supported, try adding --enable-pthreads or --cpu=i486 or higher if you are on x86"
-
 enabled asm || { arch=c; disable $ARCH_LIST $ARCH_EXT_LIST; }
 
 check_deps $CONFIG_LIST       \
@@ -4739,7 +4582,6 @@ echo "threading support         ${thread_type-no}"
 echo "safe bitstream reader     ${safe_bitstream_reader-no}"
 echo "SDL support               ${sdl-no}"
 echo "opencl enabled            ${opencl-no}"
-echo "libzvbi enabled           ${libzvbi-no}"
 echo "texi2html enabled         ${texi2html-no}"
 echo "perl enabled              ${perl-no}"
 echo "pod2man enabled           ${pod2man-no}"
@@ -4791,7 +4633,6 @@ SHLIBDIR=\$(DESTDIR)$shlibdir
 INCDIR=\$(DESTDIR)$incdir
 BINDIR=\$(DESTDIR)$bindir
 DATADIR=\$(DESTDIR)$datadir
-DOCDIR=\$(DESTDIR)$docdir
 MANDIR=\$(DESTDIR)$mandir
 SRC_PATH=$source_path
 ifndef MAIN_MAKEFILE
@@ -4813,9 +4654,9 @@ AR=$ar
 ARFLAGS=$arflags
 AR_O=$ar_o
 RANLIB=$ranlib
-STRIP=$strip
 CP=cp -p
 LN_S=$ln_s
+STRIP=$strip
 CPPFLAGS=$CPPFLAGS
 CFLAGS=$CFLAGS
 CXXFLAGS=$CXXFLAGS
@@ -4832,8 +4673,8 @@ LD_LIB=$LD_LIB
 LD_PATH=$LD_PATH
 DLLTOOL=$dlltool
 LDFLAGS=$LDFLAGS
+LDFLAGS-ffserver=$FFSERVERLDFLAGS
 SHFLAGS=$(echo $($ldflags_filter $SHFLAGS))
-ASMSTRIPFLAGS=$ASMSTRIPFLAGS
 YASMFLAGS=$YASMFLAGS
 BUILDSUF=$build_suffix
 PROGSSUF=$progs_suffix
@@ -4897,7 +4738,6 @@ get_version(){
     name=$(toupper $lcname)
     file=$source_path/$lcname/version.h
     eval $(awk "/#define ${name}_VERSION_M/ { print \$2 \"=\" \$3 }" "$file")
-    enabled raise_major && eval ${name}_VERSION_MAJOR=$((${name}_VERSION_MAJOR+100))
     eval ${name}_VERSION=\$${name}_VERSION_MAJOR.\$${name}_VERSION_MINOR.\$${name}_VERSION_MICRO
     eval echo "${lcname}_VERSION=\$${name}_VERSION" >> config.mak
     eval echo "${lcname}_VERSION_MAJOR=\$${name}_VERSION_MAJOR" >> config.mak
@@ -5013,7 +4853,7 @@ Description: $comment
 Version: $version
 Requires: $requires
 Conflicts:
-Libs: -L\${libdir} -Wl,-rpath,\${libdir} -l${shortname} $(enabled shared || echo $libs)
+Libs: -L\${libdir} -l${shortname} $(enabled shared || echo $libs)
 Cflags: -I\${includedir}
 EOF
 }

doc/APIchanges

@@ -15,96 +15,24 @@ libavutil:     2012-10-22
 
 API changes, most recent first:
 
-2013-10-27 - xxxxxxx - lavc 55.39.100 - avcodec.h
-  Add CODEC_CAP_DELAY support to avcodec_decode_subtitle2.
-
-2013-10-27 - xxxxxxx - lavu 52.48.100 - parseutils.h
-  Add av_get_known_color_name().
-
-2013-10-17 - xxxxxxx - lavu 52.47.100 - opt.h
-  Add AV_OPT_TYPE_CHANNEL_LAYOUT and channel layout option handlers
-  av_opt_get_channel_layout() and av_opt_set_channel_layout().
-
-2013-10-xx - xxxxxxx -libswscale 2.5.101 - options.c
-  Change default scaler to bicubic
-
-2013-10-03 - xxxxxxx - lavc 55.34.100 - avcodec.h
-  Add av_codec_get_max_lowres()
-
-2013-10-02 - xxxxxxx - lavf 55.19.100 - avformat.h
-  Add audio/video/subtitle AVCodec fields to AVFormatContext to force specific
-  decoders
-
-2013-08-xx - xxxxxxx - lavfi 3.11.0 - avfilter.h
-  Add AVFilterGraph.execute and AVFilterGraph.opaque for custom slice threading
-  implementations.
-
-2013-09-21 - xxxxxxx - lavu 52.16.0 - pixfmt.h
-  Add interleaved 4:2:2 8/10-bit formats AV_PIX_FMT_NV16 and
-  AV_PIX_FMT_NV20.
-
-2013-09-16 - c74c3fb / 3feb3d6 - lavu 52.44.100 / 52.15.0 - mem.h
-  Add av_reallocp.
-
-2013-09-04 - 3e1f507 - lavc 55.31.101 - avcodec.h
-  avcodec_close() argument can be NULL.
-
-2013-09-04 - 36cd017 - lavf 55.16.101 - avformat.h
-  avformat_close_input() argument can be NULL and point on NULL.
-
-2013-08-29 - e31db62 - lavf 55.15.100 - avformat.h
-  Add av_format_get_probe_score().
-
-2013-08-15 - 1e0e193 - lsws 2.5.100 -
-  Add a sws_dither AVOption, allowing to set the dither algorithm used
-
-2013-08-xx - xxxxxxx - lavc 55.27.100 - vdpau.h
-  Add a render2 alternative to the render callback function.
-
-2013-08-xx - xxxxxxx - lavc 55.26.100 - vdpau.h
-  Add allocation function for AVVDPAUContext, allowing
-  to extend it in the future without breaking ABI/API.
-
-2013-08-10 - 67a580f / 5a9a9d4 - lavc 55.25.100 / 55.16.0 - avcodec.h
-  Extend AVPacket API with av_packet_unref, av_packet_ref,
-  av_packet_move_ref, av_packet_copy_props, av_packet_free_side_data.
-
-2013-08-05 - 9547e3e / f824535 - lavc 55.22.100 / 55.13.0 - avcodec.h
-  Deprecate the bitstream-related members from struct AVVDPAUContext.
-  The bistream buffers no longer need to be explicitly freed.
-
-2013-08-05 - 3b805dc / 549294f - lavc 55.21.100 / 55.12.0 - avcodec.h
-  Deprecate the CODEC_CAP_HWACCEL_VDPAU codec capability. Use CODEC_CAP_HWACCEL
-  and select the AV_PIX_FMT_VDPAU format with get_format() instead.
-
-2013-08-05 - 4ee0984 / a0ad5d0 - lavu 52.41.100 / 52.14.0 - pixfmt.h
-  Deprecate AV_PIX_FMT_VDPAU_*. Use AV_PIX_FMT_VDPAU instead.
-
-2013-08-02 - 82fdfe8 / a8b1927 - lavc 55.20.100 / 55.11.0 - avcodec.h
-  Add output_picture_number to AVCodecParserContext.
-
-2013-07-23 - abc8110 - lavc 55.19.100 - avcodec.h
-  Add avcodec_chroma_pos_to_enum()
-  Add avcodec_enum_to_chroma_pos()
-
-2013-07-03 - 838bd73 - lavfi 3.78.100 - avfilter.h
+2013-07-03 - xxxxxxx - lavfi 3.78.100 - avfilter.h
   Deprecate avfilter_graph_parse() in favor of the equivalent
   avfilter_graph_parse_ptr().
 
-2013-06-24 - af5f9c0 / 95d5246 - lavc 55.17.100 / 55.10.0 - avcodec.h
+2013-06-xx - xxxxxxx - lavc 55.10.0 - avcodec.h
   Add MPEG-2 AAC profiles
 
-2013-06-25 - af5f9c0 / 95d5246 - lavf 55.10.100 - avformat.h
+2013-06-xx - xxxxxxx - lavf 55.10.100 - avformat.h
   Add AV_DISPOSITION_* flags to indicate text track kind.
 
-2013-06-15 - 99b8cd0 - lavu 52.36.100
+2013-06-xx - xxxxxxx - lavu 52.36.100
   Add AVRIPEMD:
    av_ripemd_alloc()
    av_ripemd_init()
    av_ripemd_update()
    av_ripemd_final()
 
-2013-06-04 - 30b491f / fc962d4 - lavu 52.35.100 / 52.13.0 - mem.h
+2013-06-05 - fc962d4 - lavu 52.13.0 - mem.h
   Add av_realloc_array and av_reallocp_array
 
 2013-05-30 - 682b227 - lavu 52.35.100
@@ -114,55 +42,55 @@ API changes, most recent first:
    av_sha512_update()
    av_sha512_final()
 
-2013-05-24 - 8d4e969 / 129bb23 - lavfi 3.10.0 / 3.70.100 - avfilter.h
+2013-05-24 - xxxxxxx - lavfi 3.70.100 - avfilter.h
   Add support for slice multithreading to lavfi. Filters supporting threading
   are marked with AVFILTER_FLAG_SLICE_THREADS.
   New fields AVFilterContext.thread_type, AVFilterGraph.thread_type and
   AVFilterGraph.nb_threads (accessible directly or through AVOptions) may be
   used to configure multithreading.
 
-2013-05-24 - fe40a9f / 2a6eaea - lavu 52.12.0 / 52.34.100 - cpu.h
+2013-05-24 - xxxxxxx - lavu 52.34.100 - cpu.h
   Add av_cpu_count() function for getting the number of logical CPUs.
 
-2013-05-24 - 0c25c39 / b493847 - lavc 55.7.0 / 55.12.100 - avcodec.h
+2013-05-24 - xxxxxxx - lavc 55.12.100 - avcodec.h
   Add picture_structure to AVCodecParserContext.
 
-2013-05-17 - 3a751ea - lavu 52.33.100 - opt.h
+2013-05-17 - xxxxxxx - lavu 52.33.100 - opt.h
   Add AV_OPT_TYPE_COLOR value to AVOptionType enum.
 
-2013-05-13 - e398416 - lavu 52.31.100 - mem.h
+2013-05-13 - xxxxxxx - lavu 52.31.100 - mem.h
   Add av_dynarray2_add().
 
-2013-05-12 - 1776177 - lavfi 3.65.100
+2013-05-12 - xxxxxxx - lavfi 3.65.100
   Add AVFILTER_FLAG_SUPPORT_TIMELINE* filter flags.
 
-2013-04-19 - 380cfce - lavc 55.4.100
+2013-04-19 - xxxxxxx - lavc 55.4.100
   Add AV_CODEC_PROP_TEXT_SUB property for text based subtitles codec.
 
-2013-04-18 - 7c1a002 - lavf 55.3.100
+2013-04-18 - xxxxxxx - lavf 55.3.100
   The matroska demuxer can now output proper verbatim ASS packets. It will
   become the default starting lavf 56.0.100.
 
-2013-04-10 - af0d270 - lavu 25.26.100 - avutil.h,opt.h
+2013-04-10 - xxxxxxx - lavu 25.26.100 - avutil.h,opt.h
   Add av_int_list_length()
   and av_opt_set_int_list().
 
-2013-03-30 - 5c73645 - lavu 52.24.100 - samplefmt.h
+2013-03-30 - xxxxxxx - lavu 52.24.100 - samplefmt.h
   Add av_samples_alloc_array_and_samples().
 
-2013-03-29 - ef7b6b4 - lavf 55.1.100 - avformat.h
+2013-03-29 - xxxxxxx - lavf 55.1.100 - avformat.h
   Add av_guess_frame_rate()
 
-2013-03-20 - 8d928a9 - lavu 52.22.100 - opt.h
+2013-03-20 - xxxxxxx - lavu 52.22.100 - opt.h
   Add AV_OPT_TYPE_DURATION value to AVOptionType enum.
 
-2013-03-17 - 7aa9af5 - lavu 52.20.100 - opt.h
+2013-03-17 - xxxxxx - lavu 52.20.100 - opt.h
   Add AV_OPT_TYPE_VIDEO_RATE value to AVOptionType enum.
 
-2013-03-07 - 9767ec6 - lavu 52.18.100 - avstring.h,bprint.h
+2013-03-07 - xxxxxx - lavu 52.18.100 - avstring.h,bprint.h
   Add av_escape() and av_bprint_escape() API.
 
-2013-02-24 - b59cd08 - lavfi 3.41.100 - buffersink.h
+2013-02-24 - xxxxxx - lavfi 3.41.100 - buffersink.h
   Add sample_rates field to AVABufferSinkParams.
 
 2013-01-17 - a1a707f - lavf 54.61.100
@@ -175,7 +103,7 @@ API changes, most recent first:
   Add AVFilterLink.channels, avfilter_link_get_channels()
   and avfilter_ref_get_channels().
 
-2012-12-15 - 96d815fc - lavc 54.80.100 - avcodec.h
+2012-12-15 - 2ada584d - lavc 54.80.100 - avcodec.h
   Add pkt_size field to AVFrame.
 
 2012-11-25 - c70ec631 - lavu 52.9.100 - opt.h
@@ -288,52 +216,52 @@ API changes, most recent first:
 2012-03-26 - a67d9cf - lavfi 2.66.100
   Add avfilter_fill_frame_from_{audio_,}buffer_ref() functions.
 
-2013-05-15 - ff46809 / e6c4ac7 - lavu 52.32.100 / 52.11.0 - pixdesc.h
+2013-05-xx - xxxxxxx - lavu 52.11.0 - pixdesc.h
   Replace PIX_FMT_* flags with AV_PIX_FMT_FLAG_*.
 
-2013-04-03 - 6fc58a8 / 507b1e4 - lavc 55.7.100 / 55.4.0 - avcodec.h
+2013-04-xx - xxxxxxx - lavc 55.4.0 - avcodec.h
   Add field_order to AVCodecParserContext.
 
-2013-04-19 - f4b05cd / 5e83d9a - lavc 55.5.100 / 55.2.0 - avcodec.h
+2013-03-xx - xxxxxxx - lavc 55.2.0 - avcodec.h
   Add CODEC_FLAG_UNALIGNED to allow decoders to produce unaligned output.
 
-2013-04-11 - lavfi 3.53.100 / 3.8.0
-  231fd44 / 38f0c07 - Move all content from avfiltergraph.h to avfilter.h. Deprecate
+2013-04-11 - lavfi 3.8.0
+  38f0c07 - Move all content from avfiltergraph.h to avfilter.h. Deprecate
             avfilterhraph.h, user applications should include just avfilter.h
-  86070b8 / bc1a985 - Add avfilter_graph_alloc_filter(), deprecate avfilter_open() and
+  bc1a985 - Add avfilter_graph_alloc_filter(), deprecate avfilter_open() and
             avfilter_graph_add_filter().
-  4fde705 / 1113672 - Add AVFilterContext.graph pointing to the AVFilterGraph that contains the
+  1113672 - Add AVFilterContext.graph pointing to the AVFilterGraph that contains the
             filter.
-  710b0aa / 48a5ada - Add avfilter_init_str(), deprecate avfilter_init_filter().
-  46de9ba / 1ba95a9 - Add avfilter_init_dict().
-  16fc24b / 7cdd737 - Add AVFilter.flags field and AVFILTER_FLAG_DYNAMIC_{INPUTS,OUTPUTS} flags.
-  f4db6bf / 7e8fe4b - Add avfilter_pad_count() for counting filter inputs/outputs.
-  835cc0f / fa2a34c - Add avfilter_next(), deprecate av_filter_next().
+  48a5ada - Add avfilter_init_str(), deprecate avfilter_init_filter().
+  1ba95a9 - Add avfilter_init_dict().
+  7cdd737 - Add AVFilter.flags field and AVFILTER_FLAG_DYNAMIC_{INPUTS,OUTPUTS} flags.
+  7e8fe4b - Add avfilter_pad_count() for counting filter inputs/outputs.
+  fa2a34c - Add avfilter_next(), deprecate av_filter_next().
             Deprecate avfilter_uninit().
 
-2013-04-09 - lavfi 3.51.100 / 3.7.0 - avfilter.h
-  0594ef0 / b439c99 - Add AVFilter.priv_class for exporting filter options through the
+2013-04-09 - lavfi 3.7.0 - avfilter.h
+  b439c99 - Add AVFilter.priv_class for exporting filter options through the
             AVOptions API in the similar way private options work in lavc and lavf.
-  44d4488 / 8114c10 - Add avfilter_get_class().
+  8114c10 - Add avfilter_get_class().
   Switch all filters to use AVOptions.
 
-2013-03-19 - 17ebef2 / 2c328a9 - lavu 52.20.100 / 52.9.0 - pixdesc.h
+2013-03-19 - 2c328a9 - lavu 52.9.0 - pixdesc.h
   Add av_pix_fmt_count_planes() function for counting planes in a pixel format.
 
-2013-03-16 - ecade98 / 42c7c61 - lavfi 3.47.100 / 3.6.0
+2013-03-16 - 42c7c61 - lavfi 3.6.0
   Add AVFilterGraph.nb_filters, deprecate AVFilterGraph.filter_count.
 
-2013-03-08 - Reference counted buffers - lavu 52.8.0, lavc 55.0.100 / 55.0.0, lavf 55.0.100 / 55.0.0,
-lavd 54.4.100 / 54.0.0, lavfi 3.5.0
-  36099df / 8e401db, 532f31a / 1cec062 - add a new API for reference counted buffers and buffer
+2013-03-08 - Reference counted buffers - lavu 52.8.0, lavc 55.0.0, lavf 55.0.0,
+lavd 54.0.0, lavfi 3.5.0
+  8e401db, 1cec062 - add a new API for reference counted buffers and buffer
                      pools (new header libavutil/buffer.h).
-  2653e12 / 1afddbe - add AVPacket.buf to allow reference counting for the AVPacket data.
+  1afddbe - add AVPacket.buf to allow reference counting for the AVPacket data.
             Add av_packet_from_data() function for constructing packets from
             av_malloc()ed data.
-  c4e8821 / 7ecc2d4 - move AVFrame from lavc to lavu (new header libavutil/frame.h), add
+  7ecc2d4 - move AVFrame from lavc to lavu (new header libavutil/frame.h), add
             AVFrame.buf/extended_buf to allow reference counting for the AVFrame
             data. Add new API for working with reference-counted AVFrames.
-  80e9e63 / 759001c - add the refcounted_frames field to AVCodecContext to make audio and
+  759001c - add the refcounted_frames field to AVCodecContext to make audio and
             video decoders return reference-counted frames. Add get_buffer2()
             callback to AVCodecContext which allocates reference-counted frames.
             Add avcodec_default_get_buffer2() as the default get_buffer2()
@@ -351,30 +279,30 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
                 * qscale_table, qstride, qscale_type, mbskip_table, motion_val,
                   mb_type, dct_coeff, ref_index -- mpegvideo-specific tables,
                   which are not exported anymore.
-  a05a44e / 7e35037 - switch libavfilter to use AVFrame instead of AVFilterBufferRef. Add
+  7e35037 - switch libavfilter to use AVFrame instead of AVFilterBufferRef. Add
             av_buffersrc_add_frame(), deprecate av_buffersrc_buffer().
             Add av_buffersink_get_frame() and av_buffersink_get_samples(),
             deprecate av_buffersink_read() and av_buffersink_read_samples().
             Deprecate AVFilterBufferRef and all functions for working with it.
 
-2013-03-17 - 6c17ff8 / 12c5c1d - lavu 52.19.100 / 52.8.0 - avstring.h
+2013-03-17 - 12c5c1d - lavu 52.8.0 - avstring.h
   Add av_isdigit, av_isgraph, av_isspace, av_isxdigit.
 
-2013-02-23 - 71cf094 / 9f12235 - lavfi 3.40.100 / 3.4.0 - avfiltergraph.h
+2013-02-23 - 9f12235 - lavfi 3.4.0 - avfiltergraph.h
   Add resample_lavr_opts to AVFilterGraph for setting libavresample options
   for auto-inserted resample filters.
 
-2013-01-25 - e7e14bc / 38c1466 - lavu 52.17.100 / 52.7.0 - dict.h
+2013-01-25 - 38c1466 - lavu 52.7.0 - dict.h
   Add av_dict_parse_string() to set multiple key/value pairs at once from a
   string.
 
-2013-01-25 - 25be630 / b85a5e8 - lavu 52.16.100 / 52.6.0 - avstring.h
+2013-01-25 - b85a5e8 - lavu 52.6.0 - avstring.h
   Add av_strnstr()
 
-2013-01-15 - e7e0186 / 8ee288d - lavu 52.15.100 / 52.5.0 - hmac.h
+2013-01-15 - 8ee288d - lavu 52.5.0 - hmac.h
   Add AVHMAC.
 
-2013-01-13 - 8ee7b38 / 44e065d - lavc 54.87.100 / 54.36.0 - vdpau.h
+2013-01-13 - 44e065d - lavc 54.87.100 / 54.36.0 - vdpau.h
   Add AVVDPAUContext struct for VDPAU hardware-accelerated decoding.
 
 2013-01-12 - dae382b / 169fb94 - lavu 52.14.100 / 52.4.0 - pixdesc.h
@@ -462,7 +390,7 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
 2012-07-29 - 7c26761 / 681ed00 - lavf 54.22.100 / 54.13.0 - avformat.h
   Add AVFMT_FLAG_NOBUFFER for low latency use cases.
 
-2012-07-10 - fbe0245 / f3e5e6f - lavu 51.65.100 / 51.37.0
+2012-07-10 - 5fade8a - lavu 51.37.0
   Add av_malloc_array() and av_mallocz_array()
 
 2012-06-22 - e847f41 / d3d3a32 - lavu 51.61.100 / 51.34.0

doc/Doxyfile

@@ -31,9 +31,9 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.
 
-PROJECT_NUMBER         = 2.1.4
+PROJECT_NUMBER         = 2.0.1
 
-# With the PROJECT_LOGO tag one can specify a logo or icon that is included
+# With the PROJECT_LOGO tag one can specify an logo or icon that is included
 # in the documentation. The maximum height of the logo should not exceed 55
 # pixels and the maximum width should not exceed 200 pixels. Doxygen will
 # copy the logo to the output directory.

doc/Makefile

@@ -88,20 +88,8 @@ $(DOCS) doc/doxy/html: | doc/
 doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(INSTHEADERS)
 	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $^
 
-install-doc: install-html install-man
-
-install-html:
-
 install-man:
 
-ifdef CONFIG_HTMLPAGES
-install-progs-$(CONFIG_DOC): install-html
-
-install-html: $(HTMLPAGES)
-	$(Q)mkdir -p "$(DOCDIR)"
-	$(INSTALL) -m 644 $(HTMLPAGES) "$(DOCDIR)"
-endif
-
 ifdef CONFIG_MANPAGES
 install-progs-$(CONFIG_DOC): install-man
 
@@ -112,16 +100,11 @@ install-man: $(MANPAGES)
 	$(INSTALL) -m 644 $(MANPAGES3) "$(MANDIR)/man3"
 endif
 
-uninstall: uninstall-doc
-
-uninstall-doc: uninstall-html uninstall-man
-
-uninstall-html:
-	$(RM) -r "$(DOCDIR)"
+uninstall: uninstall-man
 
 uninstall-man:
-	$(RM) $(addprefix "$(MANDIR)/man1/",$(PROGS-yes:%=%.1) $(PROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
-	$(RM) $(addprefix "$(MANDIR)/man3/",$(LIBRARIES-yes:%=%.3))
+	$(RM) $(addprefix "$(MANDIR)/man1/",$(MANPAGES1))
+	$(RM) $(addprefix "$(MANDIR)/man3/",$(MANPAGES3))
 
 clean:: docclean
 

doc/RELEASE_NOTES

@@ -1,7 +1,7 @@
 Release Notes
 =============
 
-* 2.1 "Fourier"  October, 2013
+* 2.0 "Nameless"  July, 2013
 
 
 General notes
@@ -14,3 +14,7 @@ accepted. If you are experiencing issues with any formally released version of
 FFmpeg, please try git master to check if the issue still exists. If it does,
 make your report against the development code following the usual bug reporting
 guidelines.
+
+AVI/AVXSynth
+--------
+If you want to use FFmpeg with AVISynth, you need AVISynth 2.6.0 at minimum.

doc/avtools-common-opts.texi

@@ -59,18 +59,10 @@ Show license.
 
 @item -h, -?, -help, --help [@var{arg}]
 Show help. An optional parameter may be specified to print help about a specific
-item. If no argument is specified, only basic (non advanced) tool
-options are shown.
+item.
 
 Possible values of @var{arg} are:
 @table @option
-@item long
-Print advanced tool options in addition to the basic tool options.
-
-@item full
-Print complete list of options, including shared and private options
-for encoders, decoders, demuxers, muxers, filters, etc.
-
 @item decoder=@var{decoder_name}
 Print detailed information about the decoder named @var{decoder_name}. Use the
 @option{-decoders} option to get a list of all decoders.
@@ -90,6 +82,7 @@ Print detailed information about the muxer named @var{muxer_name}. Use the
 @item filter=@var{filter_name}
 Print detailed information about the filter name @var{filter_name}. Use the
 @option{-filters} option to get a list of all filters.
+
 @end table
 
 @item -version
@@ -128,9 +121,6 @@ Show available sample formats.
 @item -layouts
 Show channel names and standard channel layouts.
 
-@item -colors
-Show recognized color names.
-
 @item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel}
 Set the logging level used by the library.
 Adding "repeat+" indicates that repeated log output should not be compressed
@@ -281,12 +271,11 @@ muxer:
 ffmpeg -i input.flac -id3v2_version 3 out.mp3
 @end example
 
-All codec AVOptions are per-stream, and thus a stream specifier
-should be attached to them.
+All codec AVOptions are obviously per-stream, so the chapter on stream
+specifiers applies to them
 
-Note: the @option{-nooption} syntax cannot be used for boolean
-AVOptions, use @option{-option 0}/@option{-option 1}.
+Note @option{-nooption} syntax cannot be used for boolean AVOptions,
+use @option{-option 0}/@option{-option 1}.
 
-Note: the old undocumented way of specifying per-stream AVOptions by
-prepending v/a/s to the options name is now obsolete and will be
-removed soon.
+Note2 old undocumented way of specifying per-stream AVOptions by prepending
+v/a/s to the options name is now obsolete and will be removed soon.

doc/bitstream_filters.texi

@@ -30,33 +30,7 @@ ADTS AAC container to a FLV or a MOV/MP4 file.
 
 Remove zero padding at the end of a packet.
 
-@section dump_extra
-
-Add extradata to the beginning of the filtered packets.
-
-The additional argument specifies which packets should be filtered.
-It accepts the values:
-@table @samp
-@item a
-add extradata to all key packets, but only if @var{local_header} is
-set in the @option{flags2} codec context field
-
-@item k
-add extradata to all key packets
-
-@item e
-add extradata to all packets
-@end table
-
-If not specified it is assumed @samp{k}.
-
-For example the following @command{ffmpeg} command forces a global
-header (thus disabling individual packet headers) in the H.264 packets
-generated by the @code{libx264} encoder, but corrects them by adding
-the header stored in extradata to the key packets:
-@example
-ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
-@end example
+@section dump_extradata
 
 @section h264_mp4toannexb
 
@@ -123,6 +97,6 @@ ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
 
 @section noise
 
-@section remove_extra
+@section remove_extradata
 
 @c man end BITSTREAM FILTERS

doc/codecs.texi

@@ -1,4 +1,3 @@
-@anchor{codec-options}
 @chapter Codec Options
 @c man begin CODEC OPTIONS
 
@@ -84,6 +83,9 @@ Apply interlaced motion estimation.
 Use closed gop.
 @end table
 
+@item sub_id @var{integer}
+Deprecated, currently unused.
+
 @item me_method @var{integer} (@emph{encoding,video})
 Set motion estimation method.
 
@@ -383,6 +385,10 @@ Possible values:
 
 @item simplemmx
 
+@item libmpeg2mmx
+
+@item mmi
+
 @item arm
 
 @item altivec
@@ -399,6 +405,10 @@ Possible values:
 
 @item simplealpha
 
+@item h264
+
+@item vp3
+
 @item ipp
 
 @item xvidmmx
@@ -761,26 +771,26 @@ Set noise reduction.
 Set number of bits which should be loaded into the rc buffer before
 decoding starts.
 
+@item inter_threshold @var{integer} (@emph{encoding,video})
+
 @item flags2 @var{flags} (@emph{decoding/encoding,audio,video})
 
 Possible values:
 @table @samp
 @item fast
-Allow non spec compliant speedup tricks.
+allow non spec compliant speedup tricks
 @item sgop
-Deprecated, use mpegvideo private options instead.
+Deprecated, use mpegvideo private options instead
 @item noout
-Skip bitstream encoding.
-@item ignorecrop
-Ignore cropping information from sps.
+skip bitstream encoding
 @item local_header
-Place global headers at every keyframe instead of in extradata.
+place global headers at every keyframe instead of in extradata
 @item chunks
-Frame data might be split into multiple chunks.
+Frame data might be split into multiple chunks
 @item showall
-Show all frames before the first keyframe.
+Show all frames before the first keyframe
 @item skiprd
-Deprecated, use mpegvideo private options instead.
+Deprecated, use mpegvideo private options instead
 @end table
 
 @item error @var{integer} (@emph{encoding,video})
@@ -836,10 +846,6 @@ Possible values:
 
 @item aac_eld
 
-@item mpeg2_aac_low
-
-@item mpeg2_aac_he
-
 @item dts
 
 @item dts_es
@@ -1058,27 +1064,6 @@ Set sample format audio decoders should prefer. Default value is
 
 @item sub_charenc @var{encoding} (@emph{decoding,subtitles})
 Set the input subtitles character encoding.
-
-@item field_order  @var{field_order} (@emph{video})
-Set/override the field order of the video.
-Possible values:
-@table @samp
-@item progressive
-Progressive video
-@item tt
-Interlaced video, top field coded and displayed first
-@item bb
-Interlaced video, bottom field coded and displayed first
-@item tb
-Interlaced video, top coded first, bottom displayed first
-@item bt
-Interlaced video, bottom coded first, top displayed first
-@end table
-
-@item skip_alpha @var{integer} (@emph{decoding,video})
-Set to 1 to disable processing alpha (transparency). This works like the
-@samp{gray} flag in the @option{flags} option which skips chroma information
-instead of alpha. Default is 0.
 @end table
 
 @c man end CODEC OPTIONS

doc/decoders.texi

@@ -158,45 +158,4 @@ ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
 @end table
 
-@section libzvbi-teletext
-
-Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext
-subtitles. Requires the presence of the libzvbi headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libzvbi}.
-
-@subsection Options
-
-@table @option
-@item txt_page
-List of teletext page numbers to decode. You may use the special * string to
-match all pages. Pages that do not match the specified list are dropped.
-Default value is *.
-@item txt_chop_top
-Discards the top teletext line. Default value is 1.
-@item txt_format
-Specifies the format of the decoded subtitles. The teletext decoder is capable
-of decoding the teletext pages to bitmaps or to simple text, you should use
-"bitmap" for teletext pages, because certain graphics and colors cannot be
-expressed in simple text. You might use "text" for teletext based subtitles if
-your application can handle simple text based subtitles. Default value is
-bitmap.
-@item txt_left
-X offset of generated bitmaps, default is 0.
-@item txt_top
-Y offset of generated bitmaps, default is 0.
-@item txt_chop_spaces
-Chops leading and trailing spaces and removes empty lines from the generated
-text. This option is useful for teletext based subtitles where empty spaces may
-be present at the start or at the end of the lines or empty lines may be
-present between the subtitle lines because of double-sized teletext charactes.
-Default value is 1.
-@item txt_duration
-Sets the display duration of the decoded teletext pages or subtitles in
-miliseconds. Default value is 30000 which is 30 seconds.
-@item txt_transparent
-Force transparent background of the generated teletext bitmaps. Default value
-is 0 which means an opaque (black) background.
-@end table
-
 @c man end SUBTILES DECODERS

doc/default.css

@@ -1,7 +1,3 @@
-a.summary-letter {
-    text-decoration: none;
-}
-
 a {
     color: #2D6198;
 }
@@ -49,16 +45,11 @@ body {
     text-align: center;
 }
 
-h1 a, h2 a, h3 a, h4 a {
-    text-decoration: inherit;
-    color: inherit;
-}
-
-h1, h2, h3, h4 {
+h1, h2, h3 {
     padding-left: 0.4em;
     border-radius: 4px;
-    padding-bottom: 0.25em;
-    padding-top: 0.25em;
+    padding-bottom: 0.2em;
+    padding-top: 0.2em;
     border: 1px solid #6A996A;
 }
 
@@ -72,22 +63,15 @@ h1 {
 
 h2 {
     color: #313131;
-    font-size: 1.0em;
+    font-size: 0.9em;
     background-color: #ABE3AB;
 }
 
 h3 {
-    color: #313131;
-    font-size: 0.9em;
-    margin-bottom: -6px;
-    background-color: #BBF3BB;
-}
-
-h4 {
     color: #313131;
     font-size: 0.8em;
     margin-bottom: -8px;
-    background-color: #D1FDD1;
+    background-color: #BBF3BB;
 }
 
 img {

doc/demuxers.texi

@@ -1,7 +1,7 @@
 @chapter Demuxers
 @c man begin DEMUXERS
 
-Demuxers are configured elements in FFmpeg that can read the
+Demuxers are configured elements in FFmpeg which allow to read the
 multimedia streams from a particular type of file.
 
 When you configure your FFmpeg build, all the supported demuxers
@@ -29,17 +29,6 @@ the caller can decide which variant streams to actually receive.
 The total bitrate of the variant that the stream belongs to is
 available in a metadata key named "variant_bitrate".
 
-@section asf
-
-Advanced Systems Format demuxer.
-
-This demuxer is used to demux ASF files and MMS network streams.
-
-@table @option
-@item -no_resync_search @var{bool}
-Do not try to resynchronize by looking for a certain optional start code.
-@end table
-
 @anchor{concat}
 @section concat
 
@@ -114,17 +103,6 @@ probed and 0 otherwise.
 
 @end table
 
-@section flv
-
-Adobe Flash Video Format demuxer.
-
-This demuxer is used to demux FLV files and RTMP network streams.
-
-@table @option
-@item -flv_metadata @var{bool}
-Allocate the streams according to the onMetaData array content.
-@end table
-
 @section libgme
 
 The Game Music Emu library is a collection of video game music file emulators.
@@ -262,36 +240,23 @@ Use @command{ffmpeg} for creating a video from the images in the file
 sequence @file{img-001.jpeg}, @file{img-002.jpeg}, ..., assuming an
 input frame rate of 10 frames per second:
 @example
-ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv
+ffmpeg -i 'img-%03d.jpeg' -r 10 out.mkv
 @end example
 
 @item
 As above, but start by reading from a file with index 100 in the sequence:
 @example
-ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv
+ffmpeg -start_number 100 -i 'img-%03d.jpeg' -r 10 out.mkv
 @end example
 
 @item
 Read images matching the "*.png" glob pattern , that is all the files
 terminating with the ".png" suffix:
 @example
-ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
+ffmpeg -pattern_type glob -i "*.png" -r 10 out.mkv
 @end example
 @end itemize
 
-@section mpegts
-
-MPEG-2 transport stream demuxer.
-
-@table @option
-
-@item fix_teletext_pts
-Overrides teletext packet PTS and DTS values with the timestamps calculated
-from the PCR of the first program which the teletext stream is part of and is
-not discarded. Default value is 1, set this option to 0 if you want your
-teletext packet PTS and DTS values untouched.
-@end table
-
 @section rawvideo
 
 Raw video demuxer.

doc/developer.texi

@@ -11,23 +11,29 @@
 
 @chapter Developers Guide
 
-@section Notes for external developers
+@section API
+@itemize @bullet
+@item libavcodec is the library containing the codecs (both encoding and
+decoding). Look at @file{doc/examples/decoding_encoding.c} to see how to use
+it.
 
-This document is mostly useful for internal FFmpeg developers.
-External developers who need to use the API in their application should
-refer to the API doxygen documentation in the public headers, and
-check the examples in @file{doc/examples} and in the source code to
-see how the public API is employed.
+@item libavformat is the library containing the file format handling (mux and
+demux code for several formats). Look at @file{ffplay.c} to use it in a
+player. See @file{doc/examples/muxing.c} to use it to generate audio or video
+streams.
 
-You can use the FFmpeg libraries in your commercial program, but you
-are encouraged to @emph{publish any patch you make}. In this case the
-best way to proceed is to send your patches to the ffmpeg-devel
-mailing list following the guidelines illustrated in the remainder of
-this document.
+@end itemize
 
-For more detailed legal information about the use of FFmpeg in
-external programs read the @file{LICENSE} file in the source tree and
-consult @url{http://ffmpeg.org/legal.html}.
+@section Integrating libavcodec or libavformat in your program
+
+You can integrate all the source code of the libraries to link them
+statically to avoid any version problem. All you need is to provide a
+'config.mak' and a 'config.h' in the parent directory. See the defines
+generated by ./configure to understand what is needed.
+
+You can use libavcodec or libavformat in your commercial program, but
+@emph{any patch you make must be published}. The best way to proceed is
+to send your patches to the FFmpeg mailing list.
 
 @section Contributing
 
@@ -51,16 +57,13 @@ and should try to fix issues their commit causes.
 @subsection Code formatting conventions
 
 There are the following guidelines regarding the indentation in files:
-
 @itemize @bullet
 @item
 Indent size is 4.
-
 @item
 The TAB character is forbidden outside of Makefiles as is any
 form of trailing whitespace. Commits containing either will be
 rejected by the git repository.
-
 @item
 You should try to limit your code lines to 80 characters; however, do so if
 and only if this improves readability.
@@ -114,17 +117,13 @@ int myfunc(int my_parameter)
 
 FFmpeg is programmed in the ISO C90 language with a few additional
 features from ISO C99, namely:
-
 @itemize @bullet
 @item
 the @samp{inline} keyword;
-
 @item
 @samp{//} comments;
-
 @item
 designated struct initializers (@samp{struct s x = @{ .i = 17 @};})
-
 @item
 compound literals (@samp{x = (struct s) @{ 17, 23 @};})
 @end itemize
@@ -136,17 +135,13 @@ clarity and performance.
 All code must compile with recent versions of GCC and a number of other
 currently supported compilers. To ensure compatibility, please do not use
 additional C99 features or GCC extensions. Especially watch out for:
-
 @itemize @bullet
 @item
 mixing statements and declarations;
-
 @item
 @samp{long long} (use @samp{int64_t} instead);
-
 @item
 @samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
-
 @item
 GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
 @end itemize
@@ -158,25 +153,20 @@ All names should be composed with underscores (_), not CamelCase. For example,
 for example structs and enums; they should always be in the CamelCase
 
 There are the following conventions for naming variables and functions:
-
 @itemize @bullet
 @item
 For local variables no prefix is required.
-
 @item
 For file-scope variables and functions declared as @code{static}, no prefix
 is required.
-
 @item
 For variables and functions visible outside of file scope, but only used
 internally by a library, an @code{ff_} prefix should be used,
 e.g. @samp{ff_w64_demuxer}.
-
 @item
 For variables and functions visible outside of file scope, used internally
 across multiple libraries, use @code{avpriv_} as prefix, for example,
 @samp{avpriv_aac_parse_header}.
-
 @item
 Each library has its own prefix for public symbols, in addition to the
 commonly used @code{av_} (@code{avformat_} for libavformat,
@@ -196,12 +186,10 @@ are reserved at the file level and may not be used for externally visible
 symbols. If in doubt, just avoid names starting with @code{_} altogether.
 
 @subsection Miscellaneous conventions
-
 @itemize @bullet
 @item
 fprintf and printf are forbidden in libavformat and libavcodec,
 please use av_log() instead.
-
 @item
 Casts should be used only when necessary. Unneeded parentheses
 should also be avoided if they don't make the code easier to understand.
@@ -244,154 +232,131 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
 
 @enumerate
 @item
-Contributions should be licensed under the
-@uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
-including an "or any later version" clause, or, if you prefer
-a gift-style license, the
-@uref{http://www.isc.org/software/license/, ISC} or
-@uref{http://mit-license.org/, MIT} license.
-@uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
-an "or any later version" clause is also acceptable, but LGPL is
-preferred.
-If you add a new file, give it a proper license header. Do not copy and
-paste it from a random place, use an existing file as template.
-
+   Contributions should be licensed under the
+   @uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
+   including an "or any later version" clause, or, if you prefer
+   a gift-style license, the
+   @uref{http://www.isc.org/software/license/, ISC} or
+   @uref{http://mit-license.org/, MIT} license.
+   @uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
+   an "or any later version" clause is also acceptable, but LGPL is
+   preferred.
 @item
-You must not commit code which breaks FFmpeg! (Meaning unfinished but
-enabled code which breaks compilation or compiles but does not work or
-breaks the regression tests)
-You can commit unfinished stuff (for testing etc), but it must be disabled
-(#ifdef etc) by default so it does not interfere with other developers'
-work.
-
+   You must not commit code which breaks FFmpeg! (Meaning unfinished but
+   enabled code which breaks compilation or compiles but does not work or
+   breaks the regression tests)
+   You can commit unfinished stuff (for testing etc), but it must be disabled
+   (#ifdef etc) by default so it does not interfere with other developers'
+   work.
 @item
-The commit message should have a short first line in the form of
-a @samp{topic: short description} as a header, separated by a newline
-from the body consisting of an explanation of why the change is necessary.
-If the commit fixes a known bug on the bug tracker, the commit message
-should include its bug ID. Referring to the issue on the bug tracker does
-not exempt you from writing an excerpt of the bug in the commit message.
-
+   The commit message should have a short first line in the form of
+   a @samp{topic: short description} as a header, separated by a newline
+   from the body consisting of an explanation of why the change is necessary.
+   If the commit fixes a known bug on the bug tracker, the commit message
+   should include its bug ID. Referring to the issue on the bug tracker does
+   not exempt you from writing an excerpt of the bug in the commit message.
 @item
-You do not have to over-test things. If it works for you, and you think it
-should work for others, then commit. If your code has problems
-(portability, triggers compiler bugs, unusual environment etc) they will be
-reported and eventually fixed.
-
+   You do not have to over-test things. If it works for you, and you think it
+   should work for others, then commit. If your code has problems
+   (portability, triggers compiler bugs, unusual environment etc) they will be
+   reported and eventually fixed.
 @item
-Do not commit unrelated changes together, split them into self-contained
-pieces. Also do not forget that if part B depends on part A, but A does not
-depend on B, then A can and should be committed first and separate from B.
-Keeping changes well split into self-contained parts makes reviewing and
-understanding them on the commit log mailing list easier. This also helps
-in case of debugging later on.
-Also if you have doubts about splitting or not splitting, do not hesitate to
-ask/discuss it on the developer mailing list.
-
+   Do not commit unrelated changes together, split them into self-contained
+   pieces. Also do not forget that if part B depends on part A, but A does not
+   depend on B, then A can and should be committed first and separate from B.
+   Keeping changes well split into self-contained parts makes reviewing and
+   understanding them on the commit log mailing list easier. This also helps
+   in case of debugging later on.
+   Also if you have doubts about splitting or not splitting, do not hesitate to
+   ask/discuss it on the developer mailing list.
 @item
-Do not change behavior of the programs (renaming options etc) or public
-API or ABI without first discussing it on the ffmpeg-devel mailing list.
-Do not remove functionality from the code. Just improve!
-
-Note: Redundant code can be removed.
+   Do not change behavior of the programs (renaming options etc) or public
+   API or ABI without first discussing it on the ffmpeg-devel mailing list.
+   Do not remove functionality from the code. Just improve!
 
+   Note: Redundant code can be removed.
 @item
-Do not commit changes to the build system (Makefiles, configure script)
-which change behavior, defaults etc, without asking first. The same
-applies to compiler warning fixes, trivial looking fixes and to code
-maintained by other developers. We usually have a reason for doing things
-the way we do. Send your changes as patches to the ffmpeg-devel mailing
-list, and if the code maintainers say OK, you may commit. This does not
-apply to files you wrote and/or maintain.
-
+   Do not commit changes to the build system (Makefiles, configure script)
+   which change behavior, defaults etc, without asking first. The same
+   applies to compiler warning fixes, trivial looking fixes and to code
+   maintained by other developers. We usually have a reason for doing things
+   the way we do. Send your changes as patches to the ffmpeg-devel mailing
+   list, and if the code maintainers say OK, you may commit. This does not
+   apply to files you wrote and/or maintain.
 @item
-We refuse source indentation and other cosmetic changes if they are mixed
-with functional changes, such commits will be rejected and removed. Every
-developer has his own indentation style, you should not change it. Of course
-if you (re)write something, you can use your own style, even though we would
-prefer if the indentation throughout FFmpeg was consistent (Many projects
-force a given indentation style - we do not.). If you really need to make
-indentation changes (try to avoid this), separate them strictly from real
-changes.
-
-NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
-then either do NOT change the indentation of the inner part within (do not
-move it to the right)! or do so in a separate commit
+   We refuse source indentation and other cosmetic changes if they are mixed
+   with functional changes, such commits will be rejected and removed. Every
+   developer has his own indentation style, you should not change it. Of course
+   if you (re)write something, you can use your own style, even though we would
+   prefer if the indentation throughout FFmpeg was consistent (Many projects
+   force a given indentation style - we do not.). If you really need to make
+   indentation changes (try to avoid this), separate them strictly from real
+   changes.
 
+   NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
+   then either do NOT change the indentation of the inner part within (do not
+   move it to the right)! or do so in a separate commit
 @item
-Always fill out the commit log message. Describe in a few lines what you
-changed and why. You can refer to mailing list postings if you fix a
-particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
-Recommended format:
-area changed: Short 1 line description
-
-details describing what and why and giving references.
+   Always fill out the commit log message. Describe in a few lines what you
+   changed and why. You can refer to mailing list postings if you fix a
+   particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
+   Recommended format:
+   area changed: Short 1 line description
 
+   details describing what and why and giving references.
 @item
-Make sure the author of the commit is set correctly. (see git commit --author)
-If you apply a patch, send an
-answer to ffmpeg-devel (or wherever you got the patch from) saying that
-you applied the patch.
-
+   Make sure the author of the commit is set correctly. (see git commit --author)
+   If you apply a patch, send an
+   answer to ffmpeg-devel (or wherever you got the patch from) saying that
+   you applied the patch.
 @item
-When applying patches that have been discussed (at length) on the mailing
-list, reference the thread in the log message.
-
+   When applying patches that have been discussed (at length) on the mailing
+   list, reference the thread in the log message.
 @item
-Do NOT commit to code actively maintained by others without permission.
-Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
-timeframe (12h for build failures and security fixes, 3 days small changes,
-1 week for big patches) then commit your patch if you think it is OK.
-Also note, the maintainer can simply ask for more time to review!
-
+    Do NOT commit to code actively maintained by others without permission.
+    Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
+    timeframe (12h for build failures and security fixes, 3 days small changes,
+    1 week for big patches) then commit your patch if you think it is OK.
+    Also note, the maintainer can simply ask for more time to review!
 @item
-Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
-are sent there and reviewed by all the other developers. Bugs and possible
-improvements or general questions regarding commits are discussed there. We
-expect you to react if problems with your code are uncovered.
-
+    Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
+    are sent there and reviewed by all the other developers. Bugs and possible
+    improvements or general questions regarding commits are discussed there. We
+    expect you to react if problems with your code are uncovered.
 @item
-Update the documentation if you change behavior or add features. If you are
-unsure how best to do this, send a patch to ffmpeg-devel, the documentation
-maintainer(s) will review and commit your stuff.
-
+    Update the documentation if you change behavior or add features. If you are
+    unsure how best to do this, send a patch to ffmpeg-devel, the documentation
+    maintainer(s) will review and commit your stuff.
 @item
-Try to keep important discussions and requests (also) on the public
-developer mailing list, so that all developers can benefit from them.
-
+    Try to keep important discussions and requests (also) on the public
+    developer mailing list, so that all developers can benefit from them.
 @item
-Never write to unallocated memory, never write over the end of arrays,
-always check values read from some untrusted source before using them
-as array index or other risky things.
-
+    Never write to unallocated memory, never write over the end of arrays,
+    always check values read from some untrusted source before using them
+    as array index or other risky things.
 @item
-Remember to check if you need to bump versions for the specific libav*
-parts (libavutil, libavcodec, libavformat) you are changing. You need
-to change the version integer.
-Incrementing the first component means no backward compatibility to
-previous versions (e.g. removal of a function from the public API).
-Incrementing the second component means backward compatible change
-(e.g. addition of a function to the public API or extension of an
-existing data structure).
-Incrementing the third component means a noteworthy binary compatible
-change (e.g. encoder bug fix that matters for the decoder). The third
-component always starts at 100 to distinguish FFmpeg from Libav.
-
+    Remember to check if you need to bump versions for the specific libav*
+    parts (libavutil, libavcodec, libavformat) you are changing. You need
+    to change the version integer.
+    Incrementing the first component means no backward compatibility to
+    previous versions (e.g. removal of a function from the public API).
+    Incrementing the second component means backward compatible change
+    (e.g. addition of a function to the public API or extension of an
+    existing data structure).
+    Incrementing the third component means a noteworthy binary compatible
+    change (e.g. encoder bug fix that matters for the decoder). The third
+    component always starts at 100 to distinguish FFmpeg from Libav.
 @item
-Compiler warnings indicate potential bugs or code with bad style. If a type of
-warning always points to correct and clean code, that warning should
-be disabled, not the code changed.
-Thus the remaining warnings can either be bugs or correct code.
-If it is a bug, the bug has to be fixed. If it is not, the code should
-be changed to not generate a warning unless that causes a slowdown
-or obfuscates the code.
-
+    Compiler warnings indicate potential bugs or code with bad style. If a type of
+    warning always points to correct and clean code, that warning should
+    be disabled, not the code changed.
+    Thus the remaining warnings can either be bugs or correct code.
+    If it is a bug, the bug has to be fixed. If it is not, the code should
+    be changed to not generate a warning unless that causes a slowdown
+    or obfuscates the code.
 @item
-Make sure that no parts of the codebase that you maintain are missing from the
-@file{MAINTAINERS} file. If something that you want to maintain is missing add it with
-your name after it.
-If at some point you no longer want to maintain some code, then please help
-finding a new maintainer and also don't forget updating the @file{MAINTAINERS} file.
+    If you add a new file, give it a proper license header. Do not copy and
+    paste it from a random place, use an existing file as template.
 @end enumerate
 
 We think our rules are not too hard. If you have comments, contact us.
@@ -446,51 +411,40 @@ send a reminder by email. Your patch should eventually be dealt with.
 
 @enumerate
 @item
-Did you use av_cold for codec initialization and close functions?
-
+    Did you use av_cold for codec initialization and close functions?
 @item
-Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
-AVInputFormat/AVOutputFormat struct?
-
+    Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
+    AVInputFormat/AVOutputFormat struct?
 @item
-Did you bump the minor version number (and reset the micro version
-number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
-
+    Did you bump the minor version number (and reset the micro version
+    number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
 @item
-Did you register it in @file{allcodecs.c} or @file{allformats.c}?
-
+    Did you register it in @file{allcodecs.c} or @file{allformats.c}?
 @item
-Did you add the AVCodecID to @file{avcodec.h}?
-When adding new codec IDs, also add an entry to the codec descriptor
-list in @file{libavcodec/codec_desc.c}.
-
+    Did you add the AVCodecID to @file{avcodec.h}?
+    When adding new codec IDs, also add an entry to the codec descriptor
+    list in @file{libavcodec/codec_desc.c}.
 @item
-If it has a FourCC, did you add it to @file{libavformat/riff.c},
-even if it is only a decoder?
-
+    If it has a FourCC, did you add it to @file{libavformat/riff.c},
+    even if it is only a decoder?
 @item
-Did you add a rule to compile the appropriate files in the Makefile?
-Remember to do this even if you're just adding a format to a file that is
-already being compiled by some other rule, like a raw demuxer.
-
+    Did you add a rule to compile the appropriate files in the Makefile?
+    Remember to do this even if you're just adding a format to a file that is
+    already being compiled by some other rule, like a raw demuxer.
 @item
-Did you add an entry to the table of supported formats or codecs in
-@file{doc/general.texi}?
-
+    Did you add an entry to the table of supported formats or codecs in
+    @file{doc/general.texi}?
 @item
-Did you add an entry in the Changelog?
-
+    Did you add an entry in the Changelog?
 @item
-If it depends on a parser or a library, did you add that dependency in
-configure?
-
+    If it depends on a parser or a library, did you add that dependency in
+    configure?
 @item
-Did you @code{git add} the appropriate files before committing?
-
+    Did you @code{git add} the appropriate files before committing?
 @item
-Did you make sure it compiles standalone, i.e. with
-@code{configure --disable-everything --enable-decoder=foo}
-(or @code{--enable-demuxer} or whatever your component is)?
+    Did you make sure it compiles standalone, i.e. with
+    @code{configure --disable-everything --enable-decoder=foo}
+    (or @code{--enable-demuxer} or whatever your component is)?
 @end enumerate
 
 
@@ -498,109 +452,82 @@ Did you make sure it compiles standalone, i.e. with
 
 @enumerate
 @item
-Does @code{make fate} pass with the patch applied?
-
+    Does @code{make fate} pass with the patch applied?
 @item
-Was the patch generated with git format-patch or send-email?
-
+    Was the patch generated with git format-patch or send-email?
 @item
-Did you sign off your patch? (git commit -s)
-See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
-of sign off.
-
+    Did you sign off your patch? (git commit -s)
+    See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
+    of sign off.
 @item
-Did you provide a clear git commit log message?
-
+    Did you provide a clear git commit log message?
 @item
-Is the patch against latest FFmpeg git master branch?
-
+    Is the patch against latest FFmpeg git master branch?
 @item
-Are you subscribed to ffmpeg-devel?
-(the list is subscribers only due to spam)
-
+    Are you subscribed to ffmpeg-devel?
+    (the list is subscribers only due to spam)
 @item
-Have you checked that the changes are minimal, so that the same cannot be
-achieved with a smaller patch and/or simpler final code?
-
+    Have you checked that the changes are minimal, so that the same cannot be
+    achieved with a smaller patch and/or simpler final code?
 @item
-If the change is to speed critical code, did you benchmark it?
-
+    If the change is to speed critical code, did you benchmark it?
 @item
-If you did any benchmarks, did you provide them in the mail?
-
+    If you did any benchmarks, did you provide them in the mail?
 @item
-Have you checked that the patch does not introduce buffer overflows or
-other security issues?
-
+    Have you checked that the patch does not introduce buffer overflows or
+    other security issues?
 @item
-Did you test your decoder or demuxer against damaged data? If no, see
-tools/trasher, the noise bitstream filter, and
-@uref{http://caca.zoy.org/wiki/zzuf, zzuf}. Your decoder or demuxer
-should not crash, end in a (near) infinite loop, or allocate ridiculous
-amounts of memory when fed damaged data.
-
+    Did you test your decoder or demuxer against damaged data? If no, see
+    tools/trasher, the noise bitstream filter, and
+    @uref{http://caca.zoy.org/wiki/zzuf, zzuf}. Your decoder or demuxer
+    should not crash, end in a (near) infinite loop, or allocate ridiculous
+    amounts of memory when fed damaged data.
 @item
-Does the patch not mix functional and cosmetic changes?
-
+    Does the patch not mix functional and cosmetic changes?
 @item
-Did you add tabs or trailing whitespace to the code? Both are forbidden.
-
+    Did you add tabs or trailing whitespace to the code? Both are forbidden.
 @item
-Is the patch attached to the email you send?
-
+    Is the patch attached to the email you send?
 @item
-Is the mime type of the patch correct? It should be text/x-diff or
-text/x-patch or at least text/plain and not application/octet-stream.
-
+    Is the mime type of the patch correct? It should be text/x-diff or
+    text/x-patch or at least text/plain and not application/octet-stream.
 @item
-If the patch fixes a bug, did you provide a verbose analysis of the bug?
-
+    If the patch fixes a bug, did you provide a verbose analysis of the bug?
 @item
-If the patch fixes a bug, did you provide enough information, including
-a sample, so the bug can be reproduced and the fix can be verified?
-Note please do not attach samples >100k to mails but rather provide a
-URL, you can upload to ftp://upload.ffmpeg.org
-
+    If the patch fixes a bug, did you provide enough information, including
+    a sample, so the bug can be reproduced and the fix can be verified?
+    Note please do not attach samples >100k to mails but rather provide a
+    URL, you can upload to ftp://upload.ffmpeg.org
 @item
-Did you provide a verbose summary about what the patch does change?
-
+    Did you provide a verbose summary about what the patch does change?
 @item
-Did you provide a verbose explanation why it changes things like it does?
-
+    Did you provide a verbose explanation why it changes things like it does?
 @item
-Did you provide a verbose summary of the user visible advantages and
-disadvantages if the patch is applied?
-
+    Did you provide a verbose summary of the user visible advantages and
+    disadvantages if the patch is applied?
 @item
-Did you provide an example so we can verify the new feature added by the
-patch easily?
-
+    Did you provide an example so we can verify the new feature added by the
+    patch easily?
 @item
-If you added a new file, did you insert a license header? It should be
-taken from FFmpeg, not randomly copied and pasted from somewhere else.
-
+    If you added a new file, did you insert a license header? It should be
+    taken from FFmpeg, not randomly copied and pasted from somewhere else.
 @item
-You should maintain alphabetical order in alphabetically ordered lists as
-long as doing so does not break API/ABI compatibility.
-
+    You should maintain alphabetical order in alphabetically ordered lists as
+    long as doing so does not break API/ABI compatibility.
 @item
-Lines with similar content should be aligned vertically when doing so
-improves readability.
-
+    Lines with similar content should be aligned vertically when doing so
+    improves readability.
 @item
-Consider to add a regression test for your code.
-
+    Consider to add a regression test for your code.
 @item
-If you added YASM code please check that things still work with --disable-yasm
-
+    If you added YASM code please check that things still work with --disable-yasm
 @item
-Make sure you check the return values of function and return appropriate
-error codes. Especially memory allocation functions like @code{av_malloc()}
-are notoriously left unchecked, which is a serious problem.
-
+    Make sure you check the return values of function and return appropriate
+    error codes. Especially memory allocation functions like @code{av_malloc()}
+    are notoriously left unchecked, which is a serious problem.
 @item
-Test your code with valgrind and or Address Sanitizer to ensure it's free
-of leaks, out of array accesses, etc.
+    Test your code with valgrind and or Address Sanitizer to ensure it's free
+    of leaks, out of array accesses, etc.
 @end enumerate
 
 @section Patch review process
@@ -663,15 +590,12 @@ the following steps:
 @item
     Configure to compile with instrumentation enabled:
     @code{configure --toolchain=gcov}.
-
 @item
     Run your test case, either manually or via FATE. This can be either
     the full FATE regression suite, or any arbitrary invocation of any
     front-end tool provided by FFmpeg, in any combination.
-
 @item
     Run @code{make lcov} to generate coverage data in HTML format.
-
 @item
     View @code{lcov/index.html} in your preferred HTML viewer.
 @end enumerate
@@ -706,13 +630,12 @@ There are two kinds of releases:
 
 @enumerate
 @item
-@strong{Major releases} always include the latest and greatest
-features and functionality.
-
+    @strong{Major releases} always include the latest and greatest
+    features and functionality.
 @item
-@strong{Point releases} are cut from @strong{release} branches,
-which are named @code{release/X}, with @code{X} being the release
-version number.
+    @strong{Point releases} are cut from @strong{release} branches,
+    which are named @code{release/X}, with @code{X} being the release
+    version number.
 @end enumerate
 
 Note that we promise to our users that shared libraries from any FFmpeg
@@ -733,18 +656,15 @@ inclusion into a point release:
 
 @enumerate
 @item
-Fixes a security issue, preferably identified by a @strong{CVE
-number} issued by @url{http://cve.mitre.org/}.
-
+    Fixes a security issue, preferably identified by a @strong{CVE
+    number} issued by @url{http://cve.mitre.org/}.
 @item
-Fixes a documented bug in @url{https://trac.ffmpeg.org}.
-
+    Fixes a documented bug in @url{https://trac.ffmpeg.org}.
 @item
-Improves the included documentation.
-
+    Improves the included documentation.
 @item
-Retains both source code and binary compatibility with previous
-point releases of the same release branch.
+    Retains both source code and binary compatibility with previous
+    point releases of the same release branch.
 @end enumerate
 
 The order for checking the rules is (1 OR 2 OR 3) AND 4.
@@ -756,42 +676,33 @@ The release process involves the following steps:
 
 @enumerate
 @item
-Ensure that the @file{RELEASE} file contains the version number for
-the upcoming release.
-
+    Ensure that the @file{RELEASE} file contains the version number for
+    the upcoming release.
 @item
-Add the release at @url{https://trac.ffmpeg.org/admin/ticket/versions}.
-
+    Add the release at @url{https://trac.ffmpeg.org/admin/ticket/versions}.
 @item
-Announce the intent to do a release to the mailing list.
-
+    Announce the intent to do a release to the mailing list.
 @item
-Make sure all relevant security fixes have been backported. See
-@url{https://ffmpeg.org/security.html}.
-
+    Make sure all relevant security fixes have been backported. See
+    @url{https://ffmpeg.org/security.html}.
 @item
-Ensure that the FATE regression suite still passes in the release
-branch on at least @strong{i386} and @strong{amd64}
-(cf. @ref{Regression tests}).
-
+    Ensure that the FATE regression suite still passes in the release
+    branch on at least @strong{i386} and @strong{amd64}
+    (cf. @ref{Regression tests}).
 @item
-Prepare the release tarballs in @code{bz2} and @code{gz} formats, and
-supplementing files that contain @code{gpg} signatures
-
+    Prepare the release tarballs in @code{bz2} and @code{gz} formats, and
+    supplementing files that contain @code{gpg} signatures
 @item
-Publish the tarballs at @url{http://ffmpeg.org/releases}. Create and
-push an annotated tag in the form @code{nX}, with @code{X}
-containing the version number.
-
+    Publish the tarballs at @url{http://ffmpeg.org/releases}. Create and
+    push an annotated tag in the form @code{nX}, with @code{X}
+    containing the version number.
 @item
-Propose and send a patch to the @strong{ffmpeg-devel} mailing list
-with a news entry for the website.
-
+    Propose and send a patch to the @strong{ffmpeg-devel} mailing list
+    with a news entry for the website.
 @item
-Publish the news entry.
-
+    Publish the news entry.
 @item
-Send announcement to the mailing list.
+    Send announcement to the mailing list.
 @end enumerate
 
 @bye

doc/examples/Makefile

@@ -25,7 +25,6 @@ OBJS=$(addsuffix .o,$(EXAMPLES))
 # the following examples make explicit use of the math library
 decoding_encoding: LDLIBS += -lm
 muxing:            LDLIBS += -lm
-resampling_audio:  LDLIBS += -lm
 
 .phony: all clean-test clean
 

doc/examples/README

@@ -14,5 +14,5 @@ correctly configured.
 2) Build the examples in-tree
 
 Assuming you are in the source FFmpeg checkout directory, you need to build
-FFmpeg (no need to make install in any prefix). Then you can go into
+FFmpeg (no need to make install in any prefix). Then you can go into the
 doc/examples and run a command such as PKG_CONFIG_PATH=pc-uninstalled make.

doc/examples/demuxing.c

@@ -47,6 +47,10 @@ static uint8_t *video_dst_data[4] = {NULL};
 static int      video_dst_linesize[4];
 static int video_dst_bufsize;
 
+static uint8_t **audio_dst_data = NULL;
+static int       audio_dst_linesize;
+static int audio_dst_bufsize;
+
 static int video_stream_idx = -1, audio_stream_idx = -1;
 static AVFrame *frame = NULL;
 static AVPacket pkt;
@@ -56,7 +60,6 @@ static int audio_frame_count = 0;
 static int decode_packet(int *got_frame, int cached)
 {
     int ret = 0;
-    int decoded = pkt.size;
 
     if (pkt.stream_index == video_stream_idx) {
         /* decode video frame */
@@ -88,32 +91,37 @@ static int decode_packet(int *got_frame, int cached)
             fprintf(stderr, "Error decoding audio frame\n");
             return ret;
         }
-        /* Some audio decoders decode only part of the packet, and have to be
-         * called again with the remainder of the packet data.
-         * Sample: fate-suite/lossless-audio/luckynight-partial.shn
-         * Also, some decoders might over-read the packet. */
-        decoded = FFMIN(ret, pkt.size);
 
         if (*got_frame) {
-            size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
             printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
                    cached ? "(cached)" : "",
                    audio_frame_count++, frame->nb_samples,
                    av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
 
-            /* Write the raw audio data samples of the first plane. This works
-             * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
-             * most audio decoders output planar audio, which uses a separate
-             * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
-             * In other words, this code will write only the first audio channel
-             * in these cases.
-             * You should use libswresample or libavfilter to convert the frame
-             * to packed data. */
-            fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
+            ret = av_samples_alloc(audio_dst_data, &audio_dst_linesize, av_frame_get_channels(frame),
+                                   frame->nb_samples, frame->format, 1);
+            if (ret < 0) {
+                fprintf(stderr, "Could not allocate audio buffer\n");
+                return AVERROR(ENOMEM);
+            }
+
+            /* TODO: extend return code of the av_samples_* functions so that this call is not needed */
+            audio_dst_bufsize =
+                av_samples_get_buffer_size(NULL, av_frame_get_channels(frame),
+                                           frame->nb_samples, frame->format, 1);
+
+            /* copy audio data to destination buffer:
+             * this is required since rawaudio expects non aligned data */
+            av_samples_copy(audio_dst_data, frame->data, 0, 0,
+                            frame->nb_samples, av_frame_get_channels(frame), frame->format);
+
+            /* write to rawaudio file */
+            fwrite(audio_dst_data[0], 1, audio_dst_bufsize, audio_dst_file);
+            av_freep(&audio_dst_data[0]);
         }
     }
 
-    return decoded;
+    return ret;
 }
 
 static int open_codec_context(int *stream_idx,
@@ -236,6 +244,8 @@ int main (int argc, char **argv)
     }
 
     if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
+        int nb_planes;
+
         audio_stream = fmt_ctx->streams[audio_stream_idx];
         audio_dec_ctx = audio_stream->codec;
         audio_dst_file = fopen(audio_dst_filename, "wb");
@@ -244,6 +254,15 @@ int main (int argc, char **argv)
             ret = 1;
             goto end;
         }
+
+        nb_planes = av_sample_fmt_is_planar(audio_dec_ctx->sample_fmt) ?
+            audio_dec_ctx->channels : 1;
+        audio_dst_data = av_mallocz(sizeof(uint8_t *) * nb_planes);
+        if (!audio_dst_data) {
+            fprintf(stderr, "Could not allocate audio data buffers\n");
+            ret = AVERROR(ENOMEM);
+            goto end;
+        }
     }
 
     /* dump input information to stderr */
@@ -274,15 +293,8 @@ int main (int argc, char **argv)
 
     /* read frames from the file */
     while (av_read_frame(fmt_ctx, &pkt) >= 0) {
-        AVPacket orig_pkt = pkt;
-        do {
-            ret = decode_packet(&got_frame, 0);
-            if (ret < 0)
-                break;
-            pkt.data += ret;
-            pkt.size -= ret;
-        } while (pkt.size > 0);
-        av_free_packet(&orig_pkt);
+        decode_packet(&got_frame, 0);
+        av_free_packet(&pkt);
     }
 
     /* flush cached frames */
@@ -302,25 +314,13 @@ int main (int argc, char **argv)
     }
 
     if (audio_stream) {
-        enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
-        int n_channels = audio_dec_ctx->channels;
         const char *fmt;
 
-        if (av_sample_fmt_is_planar(sfmt)) {
-            const char *packed = av_get_sample_fmt_name(sfmt);
-            printf("Warning: the sample format the decoder produced is planar "
-                   "(%s). This example will output the first channel only.\n",
-                   packed ? packed : "?");
-            sfmt = av_get_packed_sample_fmt(sfmt);
-            n_channels = 1;
-        }
-
-        if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
+        if ((ret = get_format_from_sample_fmt(&fmt, audio_dec_ctx->sample_fmt)) < 0)
             goto end;
-
         printf("Play the output audio file with the command:\n"
                "ffplay -f %s -ac %d -ar %d %s\n",
-               fmt, n_channels, audio_dec_ctx->sample_rate,
+               fmt, audio_dec_ctx->channels, audio_dec_ctx->sample_rate,
                audio_dst_filename);
     }
 
@@ -336,6 +336,7 @@ end:
         fclose(audio_dst_file);
     av_free(frame);
     av_free(video_dst_data[0]);
+    av_free(audio_dst_data);
 
     return ret < 0;
 }

doc/examples/filtering_audio.c

@@ -38,8 +38,8 @@
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>
 
-static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
-static const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";
+const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
+const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";
 
 static AVFormatContext *fmt_ctx;
 static AVCodecContext *dec_ctx;
@@ -90,9 +90,9 @@ static int init_filters(const char *filters_descr)
     AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
     AVFilterInOut *outputs = avfilter_inout_alloc();
     AVFilterInOut *inputs  = avfilter_inout_alloc();
-    static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
-    static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
-    static const int out_sample_rates[] = { 8000, -1 };
+    const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
+    const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
+    const int out_sample_rates[] = { 8000, -1 };
     const AVFilterLink *outlink;
     AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
 

doc/faq.texi

@@ -105,7 +105,7 @@ For example, img1.jpg, img2.jpg, img3.jpg,...
 Then you may run:
 
 @example
-ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
+  ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
 @end example
 
 Notice that @samp{%d} is replaced by the image number.
@@ -118,7 +118,7 @@ the sequence. This is useful if your sequence does not start with
 example will start with @file{img100.jpg}:
 
 @example
-ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
+  ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
 @end example
 
 If you have large number of pictures to rename, you can use the
@@ -128,7 +128,7 @@ that match @code{*jpg} to the @file{/tmp} directory in the sequence of
 @file{img001.jpg}, @file{img002.jpg} and so on.
 
 @example
-x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
+  x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
 @end example
 
 If you want to sequence them by oldest modified first, substitute
@@ -137,7 +137,7 @@ If you want to sequence them by oldest modified first, substitute
 Then run:
 
 @example
-ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
+  ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
 @end example
 
 The same logic is used for any image format that ffmpeg reads.
@@ -145,7 +145,7 @@ The same logic is used for any image format that ffmpeg reads.
 You can also use @command{cat} to pipe images to ffmpeg:
 
 @example
-cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
+  cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
 @end example
 
 @section How do I encode movie to single pictures?
@@ -153,7 +153,7 @@ cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
 Use:
 
 @example
-ffmpeg -i movie.mpg movie%d.jpg
+  ffmpeg -i movie.mpg movie%d.jpg
 @end example
 
 The @file{movie.mpg} used as input will be converted to
@@ -169,7 +169,7 @@ to force the encoding.
 
 Applying that to the previous example:
 @example
-ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
+  ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
 @end example
 
 Beware that there is no "jpeg" codec. Use "mjpeg" instead.
@@ -227,15 +227,15 @@ then you may use any file that DirectShow can read as input.
 
 Just create an "input.avs" text file with this single line ...
 @example
-DirectShowSource("C:\path to your file\yourfile.asf")
+  DirectShowSource("C:\path to your file\yourfile.asf")
 @end example
 ... and then feed that text file to ffmpeg:
 @example
-ffmpeg -i input.avs
+  ffmpeg -i input.avs
 @end example
 
-For ANY other help on AviSynth, please visit the
-@uref{http://www.avisynth.org/, AviSynth homepage}.
+For ANY other help on Avisynth, please visit the
+@uref{http://www.avisynth.org/, Avisynth homepage}.
 
 @section How can I join video files?
 
@@ -475,10 +475,9 @@ read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.
 
 @section Why are the ffmpeg programs devoid of debugging symbols?
 
-The build process creates @command{ffmpeg_g}, @command{ffplay_g}, etc. which
-contain full debug information. Those binaries are stripped to create
-@command{ffmpeg}, @command{ffplay}, etc. If you need the debug information, use
-the *_g versions.
+The build process creates ffmpeg_g, ffplay_g, etc. which contain full debug
+information. Those binaries are stripped to create ffmpeg, ffplay, etc. If
+you need the debug information, use the *_g versions.
 
 @section I do not like the LGPL, can I contribute code under the GPL instead?
 
@@ -498,7 +497,7 @@ An easy way to get the full list of required libraries in dependency order
 is to use @code{pkg-config}.
 
 @example
-c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
+  c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
 @end example
 
 See @file{doc/example/Makefile} and @file{doc/example/pc-uninstalled} for
@@ -522,6 +521,10 @@ to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS
 You have to create a custom AVIOContext using @code{avio_alloc_context},
 see @file{libavformat/aviobuf.c} in FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer or MPlayer2 sources.
 
+@section Where can I find libav* headers for Pascal/Delphi?
+
+see @url{http://www.iversenit.dk/dev/ffmpeg-headers/}
+
 @section Where is the documentation about ffv1, msmpeg4, asv1, 4xm?
 
 see @url{http://www.ffmpeg.org/~michael/}
@@ -534,12 +537,11 @@ In this specific case please look at RFC 4629 to see how it should be done.
 
 @section AVStream.r_frame_rate is wrong, it is much larger than the frame rate.
 
-@code{r_frame_rate} is NOT the average frame rate, it is the smallest frame rate
+r_frame_rate is NOT the average frame rate, it is the smallest frame rate
 that can accurately represent all timestamps. So no, it is not
 wrong if it is larger than the average!
-For example, if you have mixed 25 and 30 fps content, then @code{r_frame_rate}
-will be 150 (it is the least common multiple).
-If you are looking for the average frame rate, see @code{AVStream.avg_frame_rate}.
+For example, if you have mixed 25 and 30 fps content, then r_frame_rate
+will be 150.
 
 @section Why is @code{make fate} not running all tests?
 

doc/fate.texi

@@ -153,20 +153,20 @@ the synchronisation of the samples directory.
 
 @table @option
 @item fate-rsync
-Download/synchronize sample files to the configured samples directory.
+    Download/synchronize sample files to the configured samples directory.
 
 @item fate-list
-Will list all fate/regression test targets.
+    Will list all fate/regression test targets.
 
 @item fate
-Run the FATE test suite (requires the fate-suite dataset).
+    Run the FATE test suite (requires the fate-suite dataset).
 @end table
 
 @section Makefile variables
 
 @table @option
 @item V
-Verbosity level, can be set to 0, 1 or 2.
+    Verbosity level, can be set to 0, 1 or 2.
     @itemize
         @item 0: show just the test arguments
         @item 1: show just the command used in the test
@@ -174,26 +174,22 @@ Verbosity level, can be set to 0, 1 or 2.
     @end itemize
 
 @item SAMPLES
-Specify or override the path to the FATE samples at make time, it has a
-meaning only while running the regression tests.
+    Specify or override the path to the FATE samples at make time, it has a
+    meaning only while running the regression tests.
 
 @item THREADS
-Specify how many threads to use while running regression tests, it is
-quite useful to detect thread-related regressions.
-
+    Specify how many threads to use while running regression tests, it is
+    quite useful to detect thread-related regressions.
 @item THREAD_TYPE
-Specify which threading strategy test, either @var{slice} or @var{frame},
-by default @var{slice+frame}
-
+    Specify which threading strategy test, either @var{slice} or @var{frame},
+    by default @var{slice+frame}
 @item CPUFLAGS
-Specify CPU flags.
-
+    Specify CPU flags.
 @item TARGET_EXEC
-Specify or override the wrapper used to run the tests.
-The @var{TARGET_EXEC} option provides a way to run FATE wrapped in
-@command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
-through @command{ssh}.
-
+    Specify or override the wrapper used to run the tests.
+    The @var{TARGET_EXEC} option provides a way to run FATE wrapped in
+    @command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
+    through @command{ssh}.
 @item GEN
 Set to @var{1} to generate the missing or mismatched references.
 @end table

doc/ffmpeg-resampler.texi

@@ -12,7 +12,7 @@
 @chapter Description
 @c man begin DESCRIPTION
 
-The FFmpeg resampler provides a high-level interface to the
+The FFmpeg resampler provides an high-level interface to the
 libswresample library audio resampling utilities. In particular it
 allows to perform audio resampling, audio channel layout rematrixing,
 and convert audio format and packing layout.

doc/ffmpeg-scaler.texi

@@ -12,7 +12,7 @@
 @chapter Description
 @c man begin DESCRIPTION
 
-The FFmpeg rescaler provides a high-level interface to the libswscale
+The FFmpeg rescaler provides an high-level interface to the libswscale
 library image conversion utilities. In particular it allows to perform
 image rescaling and pixel format conversion.
 

doc/ffmpeg.texi

@@ -214,7 +214,7 @@ described.
 @chapter Options
 @c man begin OPTIONS
 
-@include fftools-common-opts.texi
+@include avtools-common-opts.texi
 
 @section Main options
 
@@ -272,15 +272,9 @@ 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}.
-When transcoding and @option{-accurate_seek} is enabled (the default), this
-extra segment between the seek point and @var{position} will be decoded and
-discarded. When doing stream copy or when @option{-noaccurate_seek} is used, it
-will be preserved.
-
-When used as an output option (before an output filename), decodes but discards
-input until the timestamps reach @var{position}.
+@var{position}. When used as an output option (before an output filename),
+decodes but discards input until the timestamps reach @var{position}. This is
+slower, but more accurate.
 
 @var{position} may be either in seconds or in @code{hh:mm:ss[.xxx]} form.
 
@@ -877,12 +871,13 @@ Dump each input packet to stderr.
 When dumping packets, also dump the payload.
 @item -re (@emph{input})
 Read input at native frame rate. Mainly used to simulate a grab device.
-or live input stream (e.g. when reading from a file). Should not be used
-with actual grab devices or live input streams (where it can cause packet
-loss).
 By default @command{ffmpeg} attempts to read the input(s) as fast as possible.
 This option will slow down the reading of the input(s) to the native frame rate
-of the input(s). It is useful for real-time output (e.g. live streaming).
+of the input(s). It is useful for real-time output (e.g. live streaming). If
+your input(s) is coming from some other live streaming source (through HTTP or
+UDP for example) the server might already be in real-time, thus the option will
+likely not be required. On the other hand, this is meaningful if your input(s)
+is a file you are trying to push in real-time.
 @item -loop_input
 Loop over the input stream. Currently it works only for image
 streams. This option is used for automatic FFserver testing.
@@ -1065,12 +1060,6 @@ This option is similar to @option{-filter_complex}, the only difference is that
 its argument is the name of the file from which a complex filtergraph
 description is to be read.
 
-@item -accurate_seek (@emph{input})
-This option enables or disables accurate seeking in input files with the
-@option{-ss} option. It is enabled by default, so seeking is accurate when
-transcoding. Use @option{-noaccurate_seek} to disable it, which may be useful
-e.g. when copying some streams and transcoding the others.
-
 @item -override_ffserver (@emph{global})
 Overrides the input specifications from ffserver. Using this option you can
 map any input stream to ffserver and control many aspects of the encoding from
@@ -1136,7 +1125,7 @@ then it will search for the file @file{libvpx-1080p.ffpreset}.
 
 @itemize
 @item
-For streaming at very low bitrates, use a low frame rate
+For streaming at very low bitrate application, 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:
@@ -1215,14 +1204,14 @@ standard mixer.
 Grab the X11 display with ffmpeg via
 
 @example
-ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0 /tmp/out.mpg
+ffmpeg -f x11grab -s cif -r 25 -i :0.0 /tmp/out.mpg
 @end example
 
 0.0 is display.screen number of your X11 server, same as
 the DISPLAY environment variable.
 
 @example
-ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0+10,20 /tmp/out.mpg
+ffmpeg -f x11grab -s cif -r 25 -i :0.0+10,20 /tmp/out.mpg
 @end example
 
 0.0 is display.screen number of your X11 server, same as the DISPLAY environment

doc/ffplay.texi

@@ -24,7 +24,7 @@ various FFmpeg APIs.
 @chapter Options
 @c man begin OPTIONS
 
-@include fftools-common-opts.texi
+@include avtools-common-opts.texi
 
 @section Main options
 
@@ -174,16 +174,13 @@ Toggle full screen.
 Pause.
 
 @item a
-Cycle audio channel in the curret program.
+Cycle audio channel.
 
 @item v
 Cycle video channel.
 
 @item t
-Cycle subtitle channel in the current program.
-
-@item c
-Cycle program.
+Cycle subtitle channel.
 
 @item w
 Show audio waves.

doc/ffprobe.texi

@@ -44,15 +44,14 @@ name (which may be shared by other sections), and an unique
 name. See the output of @option{sections}.
 
 Metadata tags stored in the container or in the streams are recognized
-and printed in the corresponding "FORMAT", "STREAM" or "PROGRAM_STREAM"
-section.
+and printed in the corresponding "FORMAT" or "STREAM" section.
 
 @c man end
 
 @chapter Options
 @c man begin OPTIONS
 
-@include fftools-common-opts.texi
+@include avtools-common-opts.texi
 
 @section Main options
 
@@ -113,7 +112,7 @@ ffprobe -show_packets -select_streams v:1 INPUT
 @end example
 
 @item -show_data
-Show payload data, as a hexadecimal and ASCII dump. Coupled with
+Show payload data, as an hexadecimal and ASCII dump. Coupled with
 @option{-show_packets}, it will dump the packets' data. Coupled with
 @option{-show_streams}, it will dump the codec extradata.
 
@@ -210,13 +209,6 @@ multimedia stream.
 Each media stream information is printed within a dedicated section
 with name "STREAM".
 
-@item -show_programs
-Show information about programs and their streams contained in the input
-multimedia stream.
-
-Each media stream information is printed within a dedicated section
-with name "PROGRAM_STREAM".
-
 @item -show_chapters
 Show information about chapters stored in the format.
 
@@ -230,70 +222,6 @@ corresponding stream section.
 Count the number of packets per stream and report it in the
 corresponding stream section.
 
-@item -read_intervals @var{read_intervals}
-
-Read only the specified intervals. @var{read_intervals} must be a
-sequence of interval specifications separated by ",".
-@command{ffprobe} will seek to the interval starting point, and will
-continue reading from that.
-
-Each interval is specified by two optional parts, separated by "%".
-
-The first part specifies the interval start position. It is
-interpreted as an abolute position, or as a relative offset from the
-current position if it is preceded by the "+" character. If this first
-part is not specified, no seeking will be performed when reading this
-interval.
-
-The second part specifies the interval end position. It is interpreted
-as an absolute position, or as a relative offset from the current
-position if it is preceded by the "+" character. If the offset
-specification starts with "#", it is interpreted as the number of
-packets to read (not including the flushing packets) from the interval
-start. If no second part is specified, the program will read until the
-end of the input.
-
-Note that seeking is not accurate, thus the actual interval start
-point may be different from the specified position. Also, when an
-interval duration is specified, the absolute end time will be computed
-by adding the duration to the interval start point found by seeking
-the file, rather than to the specified start value.
-
-The formal syntax is given by:
-@example
-@var{INTERVAL}  ::= [@var{START}|+@var{START_OFFSET}][%[@var{END}|+@var{END_OFFSET}]]
-@var{INTERVALS} ::= @var{INTERVAL}[,@var{INTERVALS}]
-@end example
-
-A few examples follow.
-@itemize
-@item
-Seek to time 10, read packets until 20 seconds after the found seek
-point, then seek to position @code{01:30} (1 minute and thirty
-seconds) and read packets until position @code{01:45}.
-@example
-10%+20,01:30%01:45
-@end example
-
-@item
-Read only 42 packets after seeking to position @code{01:23}:
-@example
-01:23%+#42
-@end example
-
-@item
-Read only the first 20 seconds from the start:
-@example
-%+20
-@end example
-
-@item
-Read from the start until position @code{02:30}:
-@example
-%02:30
-@end example
-@end itemize
-
 @item -show_private_data, -private
 Show private data, that is data depending on the format of the
 particular shown element.
@@ -351,8 +279,8 @@ keyN=valN
 [/SECTION]
 @end example
 
-Metadata tags are printed as a line in the corresponding FORMAT, STREAM or
-PROGRAM_STREAM section, and are prefixed by the string "TAG:".
+Metadata tags are printed as a line in the corresponding FORMAT or
+STREAM section, and are prefixed by the string "TAG:".
 
 A description of the accepted options follows.
 

doc/ffprobe.xsd

@@ -11,7 +11,6 @@
             <xsd:element name="packets"  type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" />
             <xsd:element name="frames"   type="ffprobe:framesType"  minOccurs="0" maxOccurs="1" />
             <xsd:element name="streams"  type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
-            <xsd:element name="programs" type="ffprobe:programsType" minOccurs="0" maxOccurs="1" />
             <xsd:element name="chapters" type="ffprobe:chaptersType" minOccurs="0" maxOccurs="1" />
             <xsd:element name="format"   type="ffprobe:formatType"  minOccurs="0" maxOccurs="1" />
             <xsd:element name="error"    type="ffprobe:errorType"   minOccurs="0" maxOccurs="1" />
@@ -88,12 +87,6 @@
         </xsd:sequence>
     </xsd:complexType>
 
-    <xsd:complexType name="programsType">
-        <xsd:sequence>
-            <xsd:element name="program" type="ffprobe:programType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-
     <xsd:complexType name="streamDispositionType">
       <xsd:attribute name="default"          type="xsd:int" use="required" />
       <xsd:attribute name="dub"              type="xsd:int" use="required" />
@@ -110,8 +103,8 @@
 
     <xsd:complexType name="streamType">
       <xsd:sequence>
-        <xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
         <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
+        <xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
       </xsd:sequence>
 
       <xsd:attribute name="index"            type="xsd:int" use="required"/>
@@ -138,7 +131,6 @@
       <xsd:attribute name="sample_fmt"       type="xsd:string"/>
       <xsd:attribute name="sample_rate"      type="xsd:int"/>
       <xsd:attribute name="channels"         type="xsd:int"/>
-      <xsd:attribute name="channel_layout"   type="xsd:string"/>
       <xsd:attribute name="bits_per_sample"  type="xsd:int"/>
 
       <xsd:attribute name="id"               type="xsd:string"/>
@@ -155,23 +147,6 @@
       <xsd:attribute name="nb_read_packets"  type="xsd:int"/>
     </xsd:complexType>
 
-    <xsd:complexType name="programType">
-      <xsd:sequence>
-        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-        <xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1"/>
-      </xsd:sequence>
-
-      <xsd:attribute name="program_id"           type="xsd:int"    use="required"/>
-      <xsd:attribute name="program_num"          type="xsd:int"    use="required"/>
-      <xsd:attribute name="nb_streams"           type="xsd:int"    use="required"/>
-      <xsd:attribute name="start_time"           type="xsd:float"/>
-      <xsd:attribute name="start_pts"            type="xsd:long"/>
-      <xsd:attribute name="end_time"             type="xsd:float"/>
-      <xsd:attribute name="end_pts"              type="xsd:long"/>
-      <xsd:attribute name="pmt_pid"              type="xsd:int"    use="required"/>
-      <xsd:attribute name="pcr_pid"              type="xsd:int"    use="required"/>
-    </xsd:complexType>
-
     <xsd:complexType name="formatType">
       <xsd:sequence>
         <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
@@ -179,14 +154,12 @@
 
       <xsd:attribute name="filename"         type="xsd:string" use="required"/>
       <xsd:attribute name="nb_streams"       type="xsd:int"    use="required"/>
-      <xsd:attribute name="nb_programs"      type="xsd:int"    use="required"/>
       <xsd:attribute name="format_name"      type="xsd:string" use="required"/>
       <xsd:attribute name="format_long_name" type="xsd:string"/>
       <xsd:attribute name="start_time"       type="xsd:float"/>
       <xsd:attribute name="duration"         type="xsd:float"/>
       <xsd:attribute name="size"             type="xsd:long"/>
       <xsd:attribute name="bit_rate"         type="xsd:long"/>
-      <xsd:attribute name="probe_score"      type="xsd:int"/>
     </xsd:complexType>
 
     <xsd:complexType name="tagType">

doc/ffserver.texi

@@ -229,7 +229,7 @@ differing versions of tools. FFM2 is the default.
 @chapter Options
 @c man begin OPTIONS
 
-@include fftools-common-opts.texi
+@include avtools-common-opts.texi
 
 @section Main options
 

doc/filter_design.txt

@@ -166,7 +166,7 @@ Buffer references ownership and permissions
       WRITE permission.
 
     * Filters that read their input to produce a new frame on output (like
-      scale) need the READ permission on input and must request a buffer
+      scale) need the READ permission on input and and must request a buffer
       with the WRITE permission.
 
     * Filters that intend to keep a reference after the filtering process

doc/formats.texi

@@ -57,10 +57,6 @@ Enable RTP MP4A-LATM payload.
 Reduce the latency introduced by optional buffering
 @end table
 
-@item seek2any @var{integer} (@emph{input})
-Allow seeking to non-keyframes on demuxer level when supported if set to 1.
-Default is 0.
-
 @item analyzeduration @var{integer} (@emph{input})
 Specify how many microseconds are analyzed to probe the input. A
 higher value will allow to detect more accurate information, but will
@@ -137,13 +133,6 @@ been without shifting.
 Also note that this affects only leading negative timestamps, and not
 non-monotonic negative timestamps.
 
-@item skip_initial_bytes @var{integer} (@emph{input})
-Set number of bytes to skip before reading header and frames if set to 1.
-Default is 0.
-
-@item correct_ts_overflow @var{integer} (@emph{input})
-Correct single timestamp overflows if set to 1. Default is 1.
-
 @item flush_packets @var{integer} (@emph{output})
 Flush the underlying I/O stream after each packet. Default 1 enables it, and
 has the effect of reducing the latency; 0 disables it and may slightly
@@ -152,37 +141,6 @@ increase performance in some cases.
 
 @c man end FORMAT OPTIONS
 
-@anchor{Format stream specifiers}
-@section Format stream specifiers
-
-Format stream specifiers allow selection of one or more streams that
-match specific properties.
-
-Possible forms of stream specifiers are:
-@table @option
-@item @var{stream_index}
-Matches the stream with this index.
-
-@item @var{stream_type}[:@var{stream_index}]
-@var{stream_type} is one of following: 'v' for video, 'a' for audio,
-'s' for subtitle, 'd' for data, and 't' for attachments. If
-@var{stream_index} is given, then it matches the stream number
-@var{stream_index} of this type. Otherwise, it matches all streams of
-this type.
-
-@item p:@var{program_id}[:@var{stream_index}]
-If @var{stream_index} is given, then it matches the stream with number
-@var{stream_index} in the program with the id
-@var{program_id}. Otherwise, it matches all streams in the program.
-
-@item #@var{stream_id}
-Matches the stream by a format-specific ID.
-@end table
-
-The exact semantics of stream specifiers is defined by the
-@code{avformat_match_stream_specifier()} function declared in the
-@file{libavformat/avformat.h} header.
-
 @include demuxers.texi
 @include muxers.texi
 @include metadata.texi

doc/general.texi

@@ -94,7 +94,7 @@ Then pass @code{--enable-libtwolame} to configure to enable it.
 
 @section libvpx
 
-FFmpeg can make use of the libvpx library for VP8/VP9 encoding.
+FFmpeg can make use of the libvpx library for VP8 encoding.
 
 Go to @url{http://www.webmproject.org/} and follow the instructions for
 installing the library. Then pass @code{--enable-libvpx} to configure to
@@ -133,20 +133,6 @@ Go to @url{https://github.com/dekkers/libilbc} and follow the instructions for
 installing the library. Then pass @code{--enable-libilbc} to configure to
 enable it.
 
-@section libzvbi
-
-libzvbi is a VBI decoding library which can be used by FFmpeg to decode DVB
-teletext pages and DVB teletext subtitles.
-
-Go to @url{http://sourceforge.net/projects/zapping/} and follow the instructions for
-installing the library. Then pass @code{--enable-libzvbi} to configure to
-enable it.
-
-@float NOTE
-libzvbi is licensed under the GNU General Public License Version 2 or later
-(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for details),
-you must upgrade FFmpeg's license to GPL in order to use it.
-@end float
 
 
 @chapter Supported File Formats, Codecs or Features
@@ -182,7 +168,7 @@ library:
 @item AST                       @tab X @tab X
     @tab Audio format used on the Nintendo Wii.
 @item AVI                       @tab X @tab X
-@item AviSynth                  @tab   @tab X
+@item AVISynth                  @tab   @tab X
 @item AVR                       @tab   @tab X
     @tab Audio format used on Mac.
 @item AVS                       @tab   @tab X
@@ -488,8 +474,6 @@ following image formats are supported:
     @tab YUV, JPEG and some extension is not supported yet.
 @item Truevision Targa  @tab X @tab X
     @tab Targa (.TGA) image format
-@item WebP         @tab @tab X
-    @tab WebP image format
 @item XBM  @tab X @tab X
     @tab X BitMap image format
 @item XFace @tab X @tab X
@@ -607,6 +591,7 @@ following image formats are supported:
 @item H.263+ / H.263-1998 / H.263 version 2  @tab  X  @tab  X
 @item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10  @tab  E  @tab  X
     @tab encoding supported through external library libx264
+@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 (VDPAU acceleration)  @tab  E  @tab  X
 @item HuffYUV                @tab  X  @tab  X
 @item HuffYUV FFmpeg variant @tab  X  @tab  X
 @item IBM Ultimotion         @tab     @tab  X
@@ -659,6 +644,7 @@ following image formats are supported:
 @item Motion Pixels video    @tab     @tab  X
 @item MPEG-1 video           @tab  X  @tab  X
 @item MPEG-1/2 video XvMC (X-Video Motion Compensation)  @tab     @tab  X
+@item MPEG-1/2 video (VDPAU acceleration)  @tab     @tab  X
 @item MPEG-2 video           @tab  X  @tab  X
 @item MPEG-4 part 2          @tab  X  @tab  X
     @tab libxvidcore can be used alternatively for encoding.
@@ -676,8 +662,6 @@ following image formats are supported:
     @tab fourcc: VP60,VP61,VP62
 @item VP8                    @tab  E  @tab  X
     @tab fourcc: VP80, encoding supported through external library libvpx
-@item VP9                    @tab  E  @tab  X
-    @tab encoding supported through external library libvpx
 @item Pinnacle TARGA CineWave YUV16 @tab     @tab  X
     @tab fourcc: Y216
 @item Prores                 @tab     @tab  X
@@ -827,8 +811,8 @@ following image formats are supported:
 @item Amazing Studio PAF Audio @tab     @tab  X
 @item Apple lossless audio   @tab  X  @tab  X
     @tab QuickTime fourcc 'alac'
-@item ATRAC1                 @tab     @tab  X
-@item ATRAC3                 @tab     @tab  X
+@item Atrac 1                @tab     @tab  X
+@item Atrac 3                @tab     @tab  X
 @item Bink Audio             @tab     @tab  X
     @tab Used in Bink and Smacker files in many games.
 @item CELT                   @tab     @tab  E
@@ -938,9 +922,8 @@ following image formats are supported:
     @tab Used in LucasArts SMUSH animations.
 @item Vorbis                 @tab  E  @tab  X
     @tab A native but very primitive encoder exists.
-@item Voxware MetaSound      @tab     @tab  X
-    @tab imperfect and incomplete support
-@item WavPack                @tab  X  @tab  X
+@item WavPack                @tab  E  @tab  X
+    @tab supported through external library libwavpack
 @item Westwood Audio (SND1)  @tab     @tab  X
 @item Windows Media Audio 1  @tab  X  @tab  X
 @item Windows Media Audio 2  @tab  X  @tab  X
@@ -963,7 +946,6 @@ performance on systems without hardware floating point support).
 @item 3GPP Timed Text  @tab   @tab   @tab X @tab X
 @item AQTitle          @tab   @tab X @tab   @tab X
 @item DVB              @tab X @tab X @tab X @tab X
-@item DVB teletext     @tab   @tab X @tab   @tab E
 @item DVD              @tab X @tab X @tab X @tab X
 @item JACOsub          @tab X @tab X @tab   @tab X
 @item MicroDVD         @tab X @tab X @tab   @tab X
@@ -986,8 +968,6 @@ performance on systems without hardware floating point support).
 
 @code{X} means that the feature is supported.
 
-@code{E} means that support is provided through an external library.
-
 @section Network Protocols
 
 @multitable @columnfractions .4 .1

doc/indevs.texi

@@ -485,52 +485,87 @@ For more information about OSS see:
 
 @section pulse
 
-PulseAudio input device.
+pulseaudio input device.
 
-To enable this output device you need to configure FFmpeg with @code{--enable-libpulse}.
+To enable this input device during configuration you need libpulse-simple
+installed in your system.
 
 The filename to provide to the input device is a source device or the
 string "default"
 
-To list the PulseAudio source devices and their properties you can invoke
+To list the pulse source devices and their properties you can invoke
 the command @command{pactl list sources}.
 
-More information about PulseAudio can be found on @url{http://www.pulseaudio.org}.
-
-@subsection Options
-@table @option
-@item server
-Connect to a specific PulseAudio server, specified by an IP address.
-Default server is used when not provided.
-
-@item name
-Specify the application name PulseAudio will use when showing active clients,
-by default it is the @code{LIBAVFORMAT_IDENT} string.
-
-@item stream_name
-Specify the stream name PulseAudio will use when showing active streams,
-by default it is "record".
-
-@item sample_rate
-Specify the samplerate in Hz, by default 48kHz is used.
-
-@item channels
-Specify the channels in use, by default 2 (stereo) is set.
-
-@item frame_size
-Specify the number of bytes per frame, by default it is set to 1024.
-
-@item fragment_size
-Specify the minimal buffering fragment in PulseAudio, it will affect the
-audio latency. By default it is unset.
-@end table
-
-@subsection Examples
-Record a stream from default device:
 @example
 ffmpeg -f pulse -i default /tmp/pulse.wav
 @end example
 
+@subsection @var{server} AVOption
+
+The syntax is:
+@example
+-server @var{server name}
+@end example
+
+Connects to a specific server.
+
+@subsection @var{name} AVOption
+
+The syntax is:
+@example
+-name @var{application name}
+@end example
+
+Specify the application name pulse will use when showing active clients,
+by default it is the LIBAVFORMAT_IDENT string
+
+@subsection @var{stream_name} AVOption
+
+The syntax is:
+@example
+-stream_name @var{stream name}
+@end example
+
+Specify the stream name pulse will use when showing active streams,
+by default it is "record"
+
+@subsection @var{sample_rate} AVOption
+
+The syntax is:
+@example
+-sample_rate @var{samplerate}
+@end example
+
+Specify the samplerate in Hz, by default 48kHz is used.
+
+@subsection @var{channels} AVOption
+
+The syntax is:
+@example
+-channels @var{N}
+@end example
+
+Specify the channels in use, by default 2 (stereo) is set.
+
+@subsection @var{frame_size} AVOption
+
+The syntax is:
+@example
+-frame_size @var{bytes}
+@end example
+
+Specify the number of byte per frame, by default it is set to 1024.
+
+@subsection @var{fragment_size} AVOption
+
+The syntax is:
+@example
+-fragment_size @var{bytes}
+@end example
+
+Specify the minimal buffering fragment in pulseaudio, it will affect the
+audio latency. By default it is unset.
+
 @section sndio
 
 sndio input device.
@@ -701,12 +736,12 @@ properties of your X11 display (e.g. grep for "name" or "dimensions").
 
 For example to grab from @file{:0.0} using @command{ffmpeg}:
 @example
-ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0 out.mpg
+ffmpeg -f x11grab -r 25 -s cif -i :0.0 out.mpg
 @end example
 
 Grab at position @code{10,20}:
 @example
-ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
+ffmpeg -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg
 @end example
 
 @subsection Options
@@ -727,12 +762,12 @@ zero) to the edge of region.
 
 For example:
 @example
-ffmpeg -f x11grab -follow_mouse centered -framerate 25 -video_size cif -i :0.0 out.mpg
+ffmpeg -f x11grab -follow_mouse centered -r 25 -s cif -i :0.0 out.mpg
 @end example
 
 To follow only when the mouse pointer reaches within 100 pixels to edge:
 @example
-ffmpeg -f x11grab -follow_mouse 100 -framerate 25 -video_size cif -i :0.0 out.mpg
+ffmpeg -f x11grab -follow_mouse 100 -r 25 -s cif -i :0.0 out.mpg
 @end example
 
 @item framerate
@@ -748,12 +783,12 @@ know what is being grabbed if only a portion of the screen is grabbed.
 
 For example:
 @example
-ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
+ffmpeg -f x11grab -show_region 1 -r 25 -s cif -i :0.0+10,20 out.mpg
 @end example
 
 With @var{follow_mouse}:
 @example
-ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size cif -i :0.0 out.mpg
+ffmpeg -f x11grab -follow_mouse centered -show_region 1 -r 25 -s cif -i :0.0 out.mpg
 @end example
 
 @item video_size

doc/issue_tracker.txt

@@ -1,4 +1,4 @@
-FFmpeg's bug/feature request tracker manual
+FFmpeg's bug/patch/feature request tracker manual
 =================================================
 
 NOTE: This is a draft.
@@ -11,7 +11,7 @@ existing issues can be done through a web interface.
 
 Issues can be different kinds of things we want to keep track of
 but that do not belong into the source tree itself. This includes
-bug reports, feature requests and license violations. We
+bug reports, patches, feature requests and license violations. We
 might add more items to this list in the future, so feel free to
 propose a new `type of issue' on the ffmpeg-devel mailing list if
 you feel it is worth tracking.
@@ -28,9 +28,6 @@ http(s)://trac.ffmpeg.org
 
 Type:
 -----
-art
-    Artwork such as photos, music, banners, and logos.
-
 bug / defect
     An error, flaw, mistake, failure, or fault in FFmpeg or libav* that
     prevents it from behaving as intended.
@@ -44,18 +41,20 @@ feature request / enhancement
 license violation
     ticket to keep track of (L)GPL violations of ffmpeg by others
 
-sponsoring request
-    Developer requests for hardware, software, specifications, money,
-    refunds, etc.
+patch
+    A patch as generated by diff which conforms to the patch submission and
+    development policy.
+
 
 Priority:
 ---------
 critical
-    Bugs about data loss and security issues.
+    Bugs and patches which deal with data loss and security issues.
     No feature request can be critical.
 
 important
-    Bugs which make FFmpeg unusable for a significant number of users.
+    Bugs which make FFmpeg unusable for a significant number of users, and
+    patches fixing them.
     Examples here might be completely broken MPEG-4 decoding or a build issue
     on Linux.
     While broken 4xm decoding or a broken OS/2 build would not be important,
@@ -69,7 +68,7 @@ normal
 
 
 minor
-    Bugs about things like spelling errors, "mp2" instead of
+    Bugs and patches about things like spelling errors, "mp2" instead of
     "mp3" being shown and such.
     Feature requests about things few people want or which do not make a big
     difference.
@@ -104,13 +103,13 @@ This state implicates that the bug either has been reproduced or that
 reproduction is not needed as the bug is already understood.
 
 
-Type/Status:
+Type/Status/Substatus:
 ----------
-*/new
-    Initial state of new bugs and feature requests submitted by
+*/new/new
+    Initial state of new bugs, patches and feature requests submitted by
     users.
 
-*/open
+*/open/open
     Issues which have been briefly looked at and which did not look outright
     invalid.
     This implicates that no real more detailed state applies yet. Conversely,
@@ -118,7 +117,9 @@ Type/Status:
     looked at.
 
 */closed/duplicate
-    Bugs or feature requests which are duplicates.
+    Bugs, patches or feature requests which are duplicates.
+    Note that patches dealing with the same thing in a different way are not
+    duplicates.
     Note, if you mark something as duplicate, do not forget setting the
     superseder so bug reports are properly linked.
 
@@ -133,7 +134,7 @@ Type/Status:
 bug/closed/fixed
     Bugs which have to the best of our knowledge been fixed.
 
-bug/closed/wontfix
+bug/closed/wont_fix
     Bugs which we will not fix. Possible reasons include legality, high
     complexity for the sake of supporting obscure corner cases, speed loss
     for similarly esoteric purposes, et cetera.
@@ -147,15 +148,33 @@ bug/closed/works_for_me
     reproduction failed - that is the code seems to work correctly to the
     best of our knowledge.
 
-feature_request/closed/fixed
+patch/open/approved
+    Patches which have been reviewed and approved by a developer.
+    Such patches can be applied anytime by any other developer after some
+    reasonable testing (compile + regression tests + does the patch do
+    what the author claimed).
+
+patch/open/needs_changes
+    Patches which have been reviewed and need changes to be accepted.
+
+patch/closed/applied
+    Patches which have been applied.
+
+patch/closed/rejected
+    Patches which have been rejected.
+
+feature_request/closed/implemented
     Feature requests which have been implemented.
 
-feature_request/closed/wontfix
+feature_request/closed/wont_implement
     Feature requests which will not be implemented. The reasons here could
     be legal, philosophical or others.
 
+Note, please do not use type-status-substatus combinations other than the
+above without asking on ffmpeg-dev first!
+
 Note2, if you provide the requested info do not forget to remove the
-needs_more_info resolution.
+needs_more_info substatus.
 
 Component:
 ----------

doc/libswresample.texi

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

doc/mips.txt

@@ -47,11 +47,6 @@ Files that have MIPS copyright notice in them:
 * libavutil/mips/
       float_dsp_mips.c
       libm_mips.h
-* libavcodec/
-      fft_fixed_32.c
-      fft_init_table.c
-      fft_table.h
-      mdct_fixed_32.c
 * libavcodec/mips/
       aaccoder_mips.c
       aacpsy_mips.h

doc/muxers.texi

@@ -260,7 +260,7 @@ ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg
 @table @option
 @item start_number @var{number}
 Start the sequence from @var{number}. Default value is 1. Must be a
-non-negative number.
+positive number.
 
 @item -update @var{number}
 If @var{number} is nonzero, the filename will always be interpreted as just a
@@ -451,8 +451,8 @@ pair for each track, making it easier to separate tracks.
 
 This option is implicitly set when writing ismv (Smooth Streaming) files.
 @item -movflags faststart
-Run a second pass moving the index (moov atom) to the beginning of the file.
-This operation can take a while, and will not work in various situations such
+Run a second pass moving the moov atom on top of the file. This
+operation can take a while, and will not work in various situations such
 as fragmented output, thus it is not enabled by default.
 @item -movflags rtphint
 Add RTP hinting tracks to the output file.
@@ -522,40 +522,6 @@ Set the service_id (default 0x0001) also known as program in DVB.
 Set the first PID for PMT (default 0x1000, max 0x1f00).
 @item -mpegts_start_pid @var{number}
 Set the first PID for data packets (default 0x0100, max 0x0f00).
-@item -mpegts_m2ts_mode @var{number}
-Enable m2ts mode if set to 1. Default value is -1 which disables m2ts mode.
-@item -muxrate @var{number}
-Set muxrate.
-@item -pes_payload_size @var{number}
-Set minimum PES packet payload in bytes.
-@item -mpegts_flags @var{flags}
-Set flags (see below).
-@item -mpegts_copyts @var{number}
-Preserve original timestamps, if value is set to 1. Default value is -1, which
-results in shifting timestamps so that they start from 0.
-@item -tables_version @var{number}
-Set PAT, PMT and SDT version (default 0, valid values are from 0 to 31, inclusively).
-This option allows updating stream structure so that standard consumer may
-detect the change. To do so, reopen output AVFormatContext (in case of API
-usage) or restart ffmpeg instance, cyclically changing tables_version value:
-@example
-ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
-ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
-...
-ffmpeg -i source3.ts -codec copy -f mpegts -tables_version 31 udp://1.1.1.1:1111
-ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
-ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
-...
-@end example
-@end table
-
-Option mpegts_flags may take a set of such flags:
-
-@table @option
-@item resend_headers
-Reemit PAT/PMT before writing the next packet.
-@item latm
-Use LATM packetization for AAC.
 @end table
 
 The recognized metadata settings in mpegts muxer are @code{service_provider}
@@ -775,10 +741,6 @@ Reset timestamps at the begin of each segment, so that each segment
 will start with near-zero timestamps. It is meant to ease the playback
 of the generated segments. May not work with some combinations of
 muxers/codecs. It is set to @code{0} by default.
-
-@item initial_offset @var{offset}
-Specify timestamp offset to apply to the output packet timestamps. The
-argument must be a time duration specification, and defaults to 0.
 @end table
 
 @subsection Examples
@@ -852,55 +814,19 @@ leading or trailing spaces or any special character, it must be
 escaped (see the ``Quoting and escaping'' section in the ffmpeg-utils
 manual).
 
-Muxer options can be specified for each slave by prepending them as a list of
+Options can be specified for each slave by prepending them as a list of
 @var{key}=@var{value} pairs separated by ':', between square brackets. If
 the options values contain a special character or the ':' separator, they
 must be escaped; note that this is a second level escaping.
 
-The following special options are also recognized:
-@table @option
-@item f
-Specify the format name. Useful if it cannot be guessed from the
-output name suffix.
-
-@item bsfs[/@var{spec}]
-Specify a list of bitstream filters to apply to the specified
-output. It is possible to specify to which streams a given bitstream
-filter applies, by appending a stream specifier to the option
-separated by @code{/}. If the stream specifier is not specified, the
-bistream filters will be applied to all streams in the output.
-
-Several bitstream filters can be specified, separated by ",".
-
-@item select
-Select the streams that should be mapped to the slave output,
-specified by a stream specifier. If not specified, this defaults to
-all the input streams.
-@end table
-
-Some examples follow.
-@itemize
-@item
-Encode something and both archive it in a WebM file and stream it
+Example: encode something and both archive it in a WebM file and stream it
 as MPEG-TS over UDP (the streams need to be explicitly mapped):
+
 @example
 ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
   "archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"
 @end example
 
-@item
-Use @command{ffmpeg} to encode the input, and send the output
-to three different destinations. The @code{dump_extra} bitstream
-filter is used to add extradata information to all the output video
-keyframes packets, as requested by the MPEG-TS format. The select
-option is applied to @file{out.aac} in order to make it contain only
-audio packets.
-@example
-ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental
-       -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
-@end example
-@end itemize
-
 Note: some codecs may need different options depending on the output format;
 the auto-detection of this can not work with the tee muxer. The main example
 is the @option{global_header} flag.

doc/outdevs.texi

@@ -1,7 +1,7 @@
 @chapter Output Devices
 @c man begin OUTPUT DEVICES
 
-Output devices are configured elements in FFmpeg that can write
+Output devices are configured elements in FFmpeg which allow to write
 multimedia data to an output device attached to your system.
 
 When you configure your FFmpeg build, all the supported output devices
@@ -104,74 +104,10 @@ ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors -
 @end example
 @end itemize
 
-@section fbdev
-
-Linux framebuffer output device.
-
-The Linux framebuffer is a graphic hardware-independent abstraction
-layer to show graphics on a computer monitor, typically on the
-console. It is accessed through a file device node, usually
-@file{/dev/fb0}.
-
-For more detailed information read the file
-@file{Documentation/fb/framebuffer.txt} included in the Linux source tree.
-
-@subsection Options
-@table @option
-
-@item xoffset
-@item yoffset
-Set x/y coordinate of top left corner. Default is 0.
-@end table
-
-@subsection Examples
-Play a file on framebuffer device @file{/dev/fb0}.
-Required pixel format depends on current framebuffer settings.
-@example
-ffmpeg -re -i INPUT -vcodec rawvideo -pix_fmt bgra -f fbdev /dev/fb0
-@end example
-
-See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
-
 @section oss
 
 OSS (Open Sound System) output device.
 
-@section pulse
-
-PulseAudio output device.
-
-To enable this output device you need to configure FFmpeg with @code{--enable-libpulse}.
-
-More information about PulseAudio can be found on @url{http://www.pulseaudio.org}
-
-@subsection Options
-@table @option
-
-@item server
-Connect to a specific PulseAudio server, specified by an IP address.
-Default server is used when not provided.
-
-@item name
-Specify the application name PulseAudio will use when showing active clients,
-by default it is the @code{LIBAVFORMAT_IDENT} string.
-
-@item stream_name
-Specify the stream name PulseAudio will use when showing active streams,
-by default it is set to the specified output name.
-
-@item device
-Specify the device to use. Default device is used when not provided.
-List of output devices can be obtained with command @command{pactl list sinks}.
-
-@end table
-
-@subsection Examples
-Play a file on default device on default server:
-@example
-ffmpeg  -i INPUT -f pulse "stream name"
-@end example
-
 @section sdl
 
 SDL (Simple DirectMedia Layer) output device.
@@ -203,10 +139,6 @@ Set the SDL window size, can be a string of the form
 @var{width}x@var{height} or a video size abbreviation.
 If not specified it defaults to the size of the input video,
 downscaled according to the aspect ratio.
-
-@item window_fullscreen
-Set fullscreen mode when non-zero value is provided.
-Zero is a default.
 @end table
 
 @subsection Examples

doc/print_options.c

@@ -1,20 +1,20 @@
 /*
  * Copyright (c) 2012 Anton Khirnov
  *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -27,9 +27,7 @@
 #include <float.h>
 
 #include "libavformat/avformat.h"
-#include "libavformat/options_table.h"
 #include "libavcodec/avcodec.h"
-#include "libavcodec/options_table.h"
 #include "libavutil/opt.h"
 
 static void print_usage(void)
@@ -98,14 +96,18 @@ static void show_opts(const AVOption *opts, int per_stream)
 
 static void show_format_opts(void)
 {
+#include "libavformat/options_table.h"
+
     printf("@section Format AVOptions\n");
-    show_opts(avformat_options, 0);
+    show_opts(options, 0);
 }
 
 static void show_codec_opts(void)
 {
+#include "libavcodec/options_table.h"
+
     printf("@section Codec AVOptions\n");
-    show_opts(avcodec_options, 1);
+    show_opts(options, 1);
 }
 
 int main(int argc, char **argv)

doc/protocols.texi

@@ -1,8 +1,8 @@
 @chapter Protocols
 @c man begin PROTOCOLS
 
-Protocols are configured elements in FFmpeg that enable access to
-resources that require specific protocols.
+Protocols are configured elements in FFmpeg which allow to access
+resources which require the use of a particular protocol.
 
 When you configure your FFmpeg build, all the supported protocols are
 enabled by default. You can list all available ones using the
@@ -129,20 +129,6 @@ The ff* tools default to the file protocol, that is a resource
 specified with the name "FILE.mpeg" is interpreted as the URL
 "file:FILE.mpeg".
 
-This protocol accepts the following options:
-
-@table @option
-@item truncate
-Truncate existing files on write, if set to 1. A value of 0 prevents
-truncating. Default value is 1.
-
-@item blocksize
-Set I/O operation maximum block size, in bytes. Default value is
-@code{INT_MAX}, which results in not limiting the requested block size.
-Setting this value reasonably low improves user termination request reaction
-time, which is valuable for files on slow medium.
-@end table
-
 @section ftp
 
 FTP (File Transfer Protocol).
@@ -339,16 +325,6 @@ ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
 @end example
 
-This protocol accepts the following options:
-
-@table @option
-@item blocksize
-Set I/O operation maximum block size, in bytes. Default value is
-@code{INT_MAX}, which results in not limiting the requested block size.
-Setting this value reasonably low improves user termination request reaction
-time, which is valuable if data transmission is slow.
-@end table
-
 Note that some formats (typically MOV), require the output protocol to
 be seekable, so they will fail with the pipe output protocol.
 
@@ -361,18 +337,12 @@ content across a TCP/IP network.
 
 The required syntax is:
 @example
-rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
+rtmp://@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
 @end example
 
 The accepted parameters are:
 @table @option
 
-@item username
-An optional username (mostly for publishing).
-
-@item password
-An optional password (mostly for publishing).
-
 @item server
 The address of the RTMP server.
 
@@ -423,8 +393,7 @@ times to construct arbitrary AMF sequences.
 
 @item rtmp_flashver
 Version of the Flash plugin used to run the SWF player. The default
-is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
-<libavformat version>).)
+is LNX 9,0,124,2.
 
 @item rtmp_flush_interval
 Number of packets flushed in the same request (RTMPT only). The default
@@ -474,12 +443,6 @@ For example to read with @command{ffplay} a multimedia resource named
 ffplay rtmp://myserver/vod/sample
 @end example
 
-To publish to a password protected server, passing the playpath and
-app names separately:
-@example
-ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
-@end example
-
 @section rtmpe
 
 Encrypted Real-Time Messaging Protocol.
@@ -520,39 +483,7 @@ The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
 for streaming multimedia content within HTTPS requests to traverse
 firewalls.
 
-@section libssh
-
-Secure File Transfer Protocol via libssh
-
-Allow to read from or write to remote resources using SFTP protocol.
-
-Following syntax is required.
-
-@example
-sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
-@end example
-
-This protocol accepts the following options.
-
-@table @option
-@item timeout
-Set timeout of socket I/O operations used by the underlying low level
-operation. By default it is set to -1, which means that the timeout
-is not specified.
-
-@item truncate
-Truncate existing files on write, if set to 1. A value of 0 prevents
-truncating. Default value is 1.
-
-@end table
-
-Example: Play a file stored on remote server.
-
-@example
-ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
-@end example
-
-@section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
+@section rtmp, rtmpe, rtmps, rtmpt, rtmpte
 
 Real-Time Messaging Protocol and its variants supported through
 librtmp.
@@ -594,70 +525,7 @@ ffplay "rtmp://myserver/live/mystream live=1"
 
 @section rtp
 
-Real-time Transport Protocol.
-
-The required syntax for an RTP URL is:
-rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
-
-@var{port} specifies the RTP port to use.
-
-The following URL options are supported:
-
-@table @option
-
-@item ttl=@var{n}
-Set the TTL (Time-To-Live) value (for multicast only).
-
-@item rtcpport=@var{n}
-Set the remote RTCP port to @var{n}.
-
-@item localrtpport=@var{n}
-Set the local RTP port to @var{n}.
-
-@item localrtcpport=@var{n}'
-Set the local RTCP port to @var{n}.
-
-@item pkt_size=@var{n}
-Set max packet size (in bytes) to @var{n}.
-
-@item connect=0|1
-Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
-to 0).
-
-@item sources=@var{ip}[,@var{ip}]
-List allowed source IP addresses.
-
-@item block=@var{ip}[,@var{ip}]
-List disallowed (blocked) source IP addresses.
-
-@item write_to_source=0|1
-Send packets to the source address of the latest received packet (if
-set to 1) or to a default remote address (if set to 0).
-
-@item localport=@var{n}
-Set the local RTP port to @var{n}.
-
-This is a deprecated option. Instead, @option{localrtpport} should be
-used.
-
-@end table
-
-Important notes:
-
-@enumerate
-
-@item
-If @option{rtcpport} is not set the RTCP port will be set to the RTP
-port value plus 1.
-
-@item
-If @option{localrtpport} (the local RTP port) is not set any available
-port will be used for the local RTP and RTCP ports.
-
-@item
-If @option{localrtcpport} (the local RTCP port) is not set it will be
-set to the the local RTP port value plus 1.
-@end enumerate
+Real-Time Protocol.
 
 @section rtsp
 
@@ -916,48 +784,29 @@ ffplay tcp://@var{hostname}:@var{port}
 
 @section tls
 
-Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
+Transport Layer Security/Secure Sockets Layer
 
 The required syntax for a TLS/SSL url is:
 @example
 tls://@var{hostname}:@var{port}[?@var{options}]
 @end example
 
-The following parameters can be set via command line options
-(or in code via @code{AVOption}s):
-
 @table @option
 
-@item ca_file, cafile=@var{filename}
-A file containing certificate authority (CA) root certificates to treat
-as trusted. If the linked TLS library contains a default this might not
-need to be specified for verification to work, but not all libraries and
-setups have defaults built in.
-The file must be in OpenSSL PEM format.
+@item listen
+Act as a server, listening for an incoming connection.
 
-@item tls_verify=@var{1|0}
-If enabled, try to verify the peer that we are communicating with.
-Note, if using OpenSSL, this currently only makes sure that the
-peer certificate is signed by one of the root certificates in the CA
-database, but it does not validate that the certificate actually
-matches the host name we are trying to connect to. (With GnuTLS,
-the host name is validated as well.)
+@item cafile=@var{filename}
+Certificate authority file. The file must be in OpenSSL PEM format.
 
-This is disabled by default since it requires a CA database to be
-provided by the caller in many cases.
+@item cert=@var{filename}
+Certificate file. The file must be in OpenSSL PEM format.
 
-@item cert_file, cert=@var{filename}
-A file containing a certificate to use in the handshake with the peer.
-(When operating as server, in listen mode, this is more often required
-by the peer, while client certificates only are mandated in certain
-setups.)
+@item key=@var{filename}
+Private key file.
 
-@item key_file, key=@var{filename}
-A file containing the private key for the certificate.
-
-@item listen=@var{1|0}
-If enabled, listen for connections on the provided port, and assume
-the server role in the handshake instead of the client role.
+@item verify=@var{0|1}
+Verify the peer's certificate.
 
 @end table
 
@@ -1064,24 +913,4 @@ To receive over UDP from a remote endpoint:
 ffmpeg -i udp://[@var{multicast-address}]:@var{port}
 @end example
 
-@section unix
-
-Unix local socket
-
-The required syntax for a Unix socket URL is:
-
-@example
-unix://@var{filepath}
-@end example
-
-The following parameters can be set via command line options
-(or in code via @code{AVOption}s):
-
-@table @option
-@item timeout
-Timeout in ms.
-@item listen
-Create the Unix socket in listening mode.
-@end table
-
 @c man end PROTOCOLS

doc/resampler.texi

@@ -42,11 +42,10 @@ Set the internal sample format. Default value is @code{none}.
 This will automatically be chosen when it is not explicitly set.
 
 @item icl, in_channel_layout
-@item ocl, out_channel_layout
-Set the input/output channel layout.
+Set the input channel layout.
 
-See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils}
-for the required syntax.
+@item ocl, out_channel_layout
+Set the output channel layout.
 
 @item clev, center_mix_level
 Set the center mix level. It is a value expressed in deciBel, and must be
@@ -64,11 +63,6 @@ be in the interval [-32,32].
 @item rmvol, rematrix_volume
 Set rematrix volume. Default value is 1.0.
 
-@item rematrix_maxval
-Set maximum output value for rematrixing.
-This can be used to prevent clipping vs. preventing volumn reduction
-A value of 1.0 prevents cliping.
-
 @item flags, swr_flags
 Set flags used by the converter. Default value is 0.
 

doc/scaler.texi

@@ -1,4 +1,3 @@
-@anchor{scaler_options}
 @chapter Scaler Options
 @c man begin SCALER OPTIONS
 
@@ -10,7 +9,6 @@ FFmpeg tools. For programmatic use, they can be set explicitly in the
 
 @table @option
 
-@anchor{sws_flags}
 @item sws_flags
 Set the scaler flags. This is also used to set the scaling
 algorithm. Only a single algorithm should be selected.
@@ -96,24 +94,6 @@ Set scaling algorithm parameters. The specified values are specific of
 some scaling algorithms and ignored by others. The specified values
 are floating point number values.
 
-@item sws_dither
-Set the dithering algorithm. Accepts one of the following
-values. Default value is @samp{auto}.
-
-@table @samp
-@item auto
-automatic choice
-
-@item none
-no dithering
-
-@item bayer
-bayer dither
-
-@item ed
-error diffusion dither
-@end table
-
 @end table
 
 @c man end SCALER OPTIONS

doc/snow.txt

@@ -50,10 +50,8 @@ header:
         temporal_decomposition_count    u   header_state
         spatial_decomposition_count     u   header_state
         colorspace_type                 u   header_state
-        if (nb_planes > 2) {
-            chroma_h_shift              u   header_state
-            chroma_v_shift              u   header_state
-        }
+        chroma_h_shift                  u   header_state
+        chroma_v_shift                  u   header_state
         spatial_scalability             b   header_state
         max_ref_frames-1                u   header_state
         qlogs
@@ -61,7 +59,7 @@ header:
     if(!keyframe){
         update_mc                       b   header_state
         if(update_mc){
-            for(plane=0; plane<nb_plane_types; plane++){
+            for(plane=0; plane<2; plane++){
                 diag_mc                 b   header_state
                 htaps/2-1               u   header_state
                 for(i= p->htaps/2; i; i--)
@@ -82,7 +80,7 @@ header:
     block_max_depth                     s   header_state
 
 qlogs:
-    for(plane=0; plane<nb_plane_types; plane++){
+    for(plane=0; plane<2; plane++){
         quant_table[plane][0][0]        s   header_state
         for(level=0; level < spatial_decomposition_count; level++){
             quant_table[plane][level][1]s   header_state
@@ -133,10 +131,8 @@ block(level):
 
 residual:
     residual2(luma)
-    if (nb_planes > 2) {
-        residual2(chroma_cr)
-        residual2(chroma_cb)
-    }
+    residual2(chroma_cr)
+    residual2(chroma_cb)
 
 residual2:
     for(level=0; level<spatial_decomposition_count; level++){
@@ -150,7 +146,7 @@ residual2:
 subband:
     FIXME
 
-nb_plane_types = gray ? 1 : 2;
+
 
 Tag description:
 ----------------
@@ -172,11 +168,7 @@ spatial_decomposition_count
     FIXME
 
 colorspace_type
-    0   unspecified YcbCr
-    1   Gray
-    2   Gray + Alpha
-    3   GBR
-    4   GBRA
+    0
     this MUST NOT change within a bitstream
 
 chroma_h_shift

doc/t2h.init

@@ -17,7 +17,6 @@ my $TEMPLATE_HEADER = $ENV{"FFMPEG_HEADER"} || <<EOT;
 </head>
 <body>
 <div id="container">
-<div id="body">
 EOT
 
 $PRE_BODY_CLOSE = '</div></div>';
@@ -33,7 +32,7 @@ sub FFmpeg_print_page_foot($$)
         T2H_DEFAULT_program_string() : program_string();
     print $fh '<footer class="footer pagination-right">' . "\n";
     print $fh '<span class="label label-info">' . $program_string;
-    print $fh "</span></footer></div></div></body>\n";
+    print $fh "</span></footer></div>\n";
 }
 
 $float = \&FFmpeg_float;
@@ -93,6 +92,8 @@ $Texi2HTML::THISDOC{program_authors}
 
 $description
 <meta name="keywords" content="$longtitle">
+<meta name="resource-type" content="document">
+<meta name="distribution" content="global">
 <meta name="Generator" content="$Texi2HTML::THISDOC{program}">
 $encoding
 $CSS_LINES

doc/utils.texi

@@ -96,42 +96,14 @@ year-month-day.
 @anchor{time duration syntax}
 @section Time duration
 
-There are two accepted syntaxes for expressing time duration.
-
+The accepted syntax is:
 @example
-[-][@var{HH}:]@var{MM}:@var{SS}[.@var{m}...]
+[-][HH:]MM:SS[.m...]
+[-]S+[.m...]
 @end example
 
-@var{HH} expresses the number of hours, @var{MM} the number of minutes
-for a maximum of 2 digits, and @var{SS} the number of seconds for a
-maximum of 2 digits. The @var{m} at the end expresses decimal value for
-@var{SS}.
-
-@emph{or}
-
-@example
-[-]@var{S}+[.@var{m}...]
-@end example
-
-@var{S} expresses the number of seconds, with the optional decimal part
-@var{m}.
-
-In both expressions, the optional @samp{-} indicates negative duration.
-
-@subsection Examples
-
-The following examples are all valid time duration:
-
-@table @samp
-@item 55
-55 seconds
-
-@item 12:03:45
-12 hours, 03 minutes and 45 seconds
-
-@item 23.189
-23.189 seconds
-@end table
+@var{HH} expresses the number of hours, @var{MM} the number a of minutes
+and @var{SS} the number of seconds.
 
 @anchor{video size syntax}
 @section Video size
@@ -226,18 +198,6 @@ The following abbreviations are recognized:
 3996x2160
 @item 4kscope
 4096x1716
-@item nhd
-640x360
-@item hqvga
-240x160
-@item wqvga
-400x240
-@item fwqvga
-432x240
-@item hvga
-480x320
-@item qhd
-960x540
 @end table
 
 @anchor{video rate syntax}
@@ -283,450 +243,18 @@ The undefined value can be expressed using the "0:0" string.
 @anchor{color syntax}
 @section Color
 
-It can be the name of a color as defined below (case insensitive match) or a
-@code{[0x|#]RRGGBB[AA]} sequence, possibly followed by @@ and a string
+It can be the name of a color (case insensitive match) or a
+[0x|#]RRGGBB[AA] sequence, possibly followed by "@@" and a string
 representing the alpha component.
 
 The alpha component may be a string composed by "0x" followed by an
 hexadecimal number or a decimal number between 0.0 and 1.0, which
-represents the opacity value (@samp{0x00} or @samp{0.0} means completely
-transparent, @samp{0xff} or @samp{1.0} completely opaque). If the alpha
-component is not specified then @samp{0xff} is assumed.
+represents the opacity value (0x00/0.0 means completely transparent,
+0xff/1.0 completely opaque).
+If the alpha component is not specified then 0xff is assumed.
 
-The string @samp{random} will result in a random color.
+The string "random" will result in a random color.
 
-The following names of colors are recognized:
-@table @samp
-@item AliceBlue
-0xF0F8FF
-@item AntiqueWhite
-0xFAEBD7
-@item Aqua
-0x00FFFF
-@item Aquamarine
-0x7FFFD4
-@item Azure
-0xF0FFFF
-@item Beige
-0xF5F5DC
-@item Bisque
-0xFFE4C4
-@item Black
-0x000000
-@item BlanchedAlmond
-0xFFEBCD
-@item Blue
-0x0000FF
-@item BlueViolet
-0x8A2BE2
-@item Brown
-0xA52A2A
-@item BurlyWood
-0xDEB887
-@item CadetBlue
-0x5F9EA0
-@item Chartreuse
-0x7FFF00
-@item Chocolate
-0xD2691E
-@item Coral
-0xFF7F50
-@item CornflowerBlue
-0x6495ED
-@item Cornsilk
-0xFFF8DC
-@item Crimson
-0xDC143C
-@item Cyan
-0x00FFFF
-@item DarkBlue
-0x00008B
-@item DarkCyan
-0x008B8B
-@item DarkGoldenRod
-0xB8860B
-@item DarkGray
-0xA9A9A9
-@item DarkGreen
-0x006400
-@item DarkKhaki
-0xBDB76B
-@item DarkMagenta
-0x8B008B
-@item DarkOliveGreen
-0x556B2F
-@item Darkorange
-0xFF8C00
-@item DarkOrchid
-0x9932CC
-@item DarkRed
-0x8B0000
-@item DarkSalmon
-0xE9967A
-@item DarkSeaGreen
-0x8FBC8F
-@item DarkSlateBlue
-0x483D8B
-@item DarkSlateGray
-0x2F4F4F
-@item DarkTurquoise
-0x00CED1
-@item DarkViolet
-0x9400D3
-@item DeepPink
-0xFF1493
-@item DeepSkyBlue
-0x00BFFF
-@item DimGray
-0x696969
-@item DodgerBlue
-0x1E90FF
-@item FireBrick
-0xB22222
-@item FloralWhite
-0xFFFAF0
-@item ForestGreen
-0x228B22
-@item Fuchsia
-0xFF00FF
-@item Gainsboro
-0xDCDCDC
-@item GhostWhite
-0xF8F8FF
-@item Gold
-0xFFD700
-@item GoldenRod
-0xDAA520
-@item Gray
-0x808080
-@item Green
-0x008000
-@item GreenYellow
-0xADFF2F
-@item HoneyDew
-0xF0FFF0
-@item HotPink
-0xFF69B4
-@item IndianRed
-0xCD5C5C
-@item Indigo
-0x4B0082
-@item Ivory
-0xFFFFF0
-@item Khaki
-0xF0E68C
-@item Lavender
-0xE6E6FA
-@item LavenderBlush
-0xFFF0F5
-@item LawnGreen
-0x7CFC00
-@item LemonChiffon
-0xFFFACD
-@item LightBlue
-0xADD8E6
-@item LightCoral
-0xF08080
-@item LightCyan
-0xE0FFFF
-@item LightGoldenRodYellow
-0xFAFAD2
-@item LightGreen
-0x90EE90
-@item LightGrey
-0xD3D3D3
-@item LightPink
-0xFFB6C1
-@item LightSalmon
-0xFFA07A
-@item LightSeaGreen
-0x20B2AA
-@item LightSkyBlue
-0x87CEFA
-@item LightSlateGray
-0x778899
-@item LightSteelBlue
-0xB0C4DE
-@item LightYellow
-0xFFFFE0
-@item Lime
-0x00FF00
-@item LimeGreen
-0x32CD32
-@item Linen
-0xFAF0E6
-@item Magenta
-0xFF00FF
-@item Maroon
-0x800000
-@item MediumAquaMarine
-0x66CDAA
-@item MediumBlue
-0x0000CD
-@item MediumOrchid
-0xBA55D3
-@item MediumPurple
-0x9370D8
-@item MediumSeaGreen
-0x3CB371
-@item MediumSlateBlue
-0x7B68EE
-@item MediumSpringGreen
-0x00FA9A
-@item MediumTurquoise
-0x48D1CC
-@item MediumVioletRed
-0xC71585
-@item MidnightBlue
-0x191970
-@item MintCream
-0xF5FFFA
-@item MistyRose
-0xFFE4E1
-@item Moccasin
-0xFFE4B5
-@item NavajoWhite
-0xFFDEAD
-@item Navy
-0x000080
-@item OldLace
-0xFDF5E6
-@item Olive
-0x808000
-@item OliveDrab
-0x6B8E23
-@item Orange
-0xFFA500
-@item OrangeRed
-0xFF4500
-@item Orchid
-0xDA70D6
-@item PaleGoldenRod
-0xEEE8AA
-@item PaleGreen
-0x98FB98
-@item PaleTurquoise
-0xAFEEEE
-@item PaleVioletRed
-0xD87093
-@item PapayaWhip
-0xFFEFD5
-@item PeachPuff
-0xFFDAB9
-@item Peru
-0xCD853F
-@item Pink
-0xFFC0CB
-@item Plum
-0xDDA0DD
-@item PowderBlue
-0xB0E0E6
-@item Purple
-0x800080
-@item Red
-0xFF0000
-@item RosyBrown
-0xBC8F8F
-@item RoyalBlue
-0x4169E1
-@item SaddleBrown
-0x8B4513
-@item Salmon
-0xFA8072
-@item SandyBrown
-0xF4A460
-@item SeaGreen
-0x2E8B57
-@item SeaShell
-0xFFF5EE
-@item Sienna
-0xA0522D
-@item Silver
-0xC0C0C0
-@item SkyBlue
-0x87CEEB
-@item SlateBlue
-0x6A5ACD
-@item SlateGray
-0x708090
-@item Snow
-0xFFFAFA
-@item SpringGreen
-0x00FF7F
-@item SteelBlue
-0x4682B4
-@item Tan
-0xD2B48C
-@item Teal
-0x008080
-@item Thistle
-0xD8BFD8
-@item Tomato
-0xFF6347
-@item Turquoise
-0x40E0D0
-@item Violet
-0xEE82EE
-@item Wheat
-0xF5DEB3
-@item White
-0xFFFFFF
-@item WhiteSmoke
-0xF5F5F5
-@item Yellow
-0xFFFF00
-@item YellowGreen
-0x9ACD32
-@end table
-
-@anchor{channel layout syntax}
-@section Channel Layout
-
-A channel layout specifies the spatial disposition of the channels in
-a multi-channel audio stream. To specify a channel layout, FFmpeg
-makes use of a special syntax.
-
-Individual channels are identified by an id, as given by the table
-below:
-@table @samp
-@item FL
-front left
-@item FR
-front right
-@item FC
-front center
-@item LFE
-low frequency
-@item BL
-back left
-@item BR
-back right
-@item FLC
-front left-of-center
-@item FRC
-front right-of-center
-@item BC
-back center
-@item SL
-side left
-@item SR
-side right
-@item TC
-top center
-@item TFL
-top front left
-@item TFC
-top front center
-@item TFR
-top front right
-@item TBL
-top back left
-@item TBC
-top back center
-@item TBR
-top back right
-@item DL
-downmix left
-@item DR
-downmix right
-@item WL
-wide left
-@item WR
-wide right
-@item SDL
-surround direct left
-@item SDR
-surround direct right
-@item LFE2
-low frequency 2
-@end table
-
-Standard channel layout compositions can be specified by using the
-following identifiers:
-@table @samp
-@item mono
-FC
-@item stereo
-FL+FR
-@item 2.1
-FL+FR+LFE
-@item 3.0
-FL+FR+FC
-@item 3.0(back)
-FL+FR+BC
-@item 4.0
-FL+FR+FC+BC
-@item quad
-FL+FR+BL+BR
-@item quad(side)
-FL+FR+SL+SR
-@item 3.1
-FL+FR+FC+LFE
-@item 5.0
-FL+FR+FC+BL+BR
-@item 5.0(side)
-FL+FR+FC+SL+SR
-@item 4.1
-FL+FR+FC+LFE+BC
-@item 5.1
-FL+FR+FC+LFE+BL+BR
-@item 5.1(side)
-FL+FR+FC+LFE+SL+SR
-@item 6.0
-FL+FR+FC+BC+SL+SR
-@item 6.0(front)
-FL+FR+FLC+FRC+SL+SR
-@item hexagonal
-FL+FR+FC+BL+BR+BC
-@item 6.1
-FL+FR+FC+LFE+BC+SL+SR
-@item 6.1
-FL+FR+FC+LFE+BL+BR+BC
-@item 6.1(front)
-FL+FR+LFE+FLC+FRC+SL+SR
-@item 7.0
-FL+FR+FC+BL+BR+SL+SR
-@item 7.0(front)
-FL+FR+FC+FLC+FRC+SL+SR
-@item 7.1
-FL+FR+FC+LFE+BL+BR+SL+SR
-@item 7.1(wide)
-FL+FR+FC+LFE+BL+BR+FLC+FRC
-@item 7.1(wide-side)
-FL+FR+FC+LFE+FLC+FRC+SL+SR
-@item octagonal
-FL+FR+FC+BL+BR+BC+SL+SR
-@item downmix
-DL+DR
-@end table
-
-A custom channel layout can be specified as a sequence of terms, separated by
-'+' or '|'. Each term can be:
-@itemize
-@item
-the name of a standard channel layout (e.g. @samp{mono},
-@samp{stereo}, @samp{4.0}, @samp{quad}, @samp{5.0}, etc.)
-
-@item
-the name of a single channel (e.g. @samp{FL}, @samp{FR}, @samp{FC}, @samp{LFE}, etc.)
-
-@item
-a number of channels, in decimal, optionally followed by 'c', yielding
-the default channel layout for that number of channels (see the
-function @code{av_get_default_channel_layout})
-
-@item
-a channel layout mask, in hexadecimal starting with "0x" (see the
-@code{AV_CH_*} macros in @file{libavutil/channel_layout.h}.
-@end itemize
-
-Starting from libavutil version 53 the trailing character "c" to
-specify a number of channels will be required, while a channel layout
-mask could also be specified as a decimal number (if and only if not
-followed by "c").
-
-See also the function @code{av_get_channel_layout} defined in
-@file{libavutil/channel_layout.h}.
 @c man end SYNTAX
 
 @chapter Expression Evaluation

ffmpeg.c

@@ -751,9 +751,7 @@ static void do_subtitle_out(AVFormatContext *s,
         nb = 1;
 
     /* shift timestamp to honor -ss and make check_recording_time() work with -t */
-    pts = sub->pts;
-    if (output_files[ost->file_index]->start_time != AV_NOPTS_VALUE)
-        pts -= output_files[ost->file_index]->start_time;
+    pts = sub->pts - output_files[ost->file_index]->start_time;
     for (i = 0; i < nb; i++) {
         ost->sync_opts = av_rescale_q(pts, AV_TIME_BASE_Q, enc->time_base);
         if (!check_recording_time(ost))
@@ -817,26 +815,10 @@ static void do_video_out(AVFormatContext *s,
     nb_frames = 1;
 
     format_video_sync = video_sync_method;
-    if (format_video_sync == VSYNC_AUTO) {
+    if (format_video_sync == VSYNC_AUTO)
         format_video_sync = (s->oformat->flags & AVFMT_VARIABLE_FPS) ? ((s->oformat->flags & AVFMT_NOTIMESTAMPS) ? VSYNC_PASSTHROUGH : VSYNC_VFR) : VSYNC_CFR;
-        if (   ist
-            && format_video_sync == VSYNC_CFR
-            && input_files[ist->file_index]->ctx->nb_streams == 1
-            && input_files[ist->file_index]->input_ts_offset == 0) {
-            format_video_sync = VSYNC_VSCFR;
-        }
-        if (format_video_sync == VSYNC_CFR && copy_ts) {
-            format_video_sync = VSYNC_VSCFR;
-        }
-    }
 
     switch (format_video_sync) {
-    case VSYNC_VSCFR:
-        if (ost->frame_number == 0 && delta - duration >= 0.5) {
-            av_log(NULL, AV_LOG_DEBUG, "Not duplicating %d initial frames\n", (int)lrintf(delta - duration));
-            delta = duration;
-            ost->sync_opts = lrint(sync_ipts);
-        }
     case VSYNC_CFR:
         // FIXME set to 0.5 after we fix some dts/pts bugs like in avidec.c
         if (delta < -1.1)
@@ -1088,11 +1070,10 @@ static int reap_filters(void)
             }
             frame_pts = AV_NOPTS_VALUE;
             if (filtered_frame->pts != AV_NOPTS_VALUE) {
-                int64_t start_time = (of->start_time == AV_NOPTS_VALUE) ? 0 : of->start_time;
                 filtered_frame->pts = frame_pts = av_rescale_q(filtered_frame->pts,
                                                 ost->filter->filter->inputs[0]->time_base,
                                                 ost->st->codec->time_base) -
-                                    av_rescale_q(start_time,
+                                    av_rescale_q(of->start_time,
                                                 AV_TIME_BASE_Q,
                                                 ost->st->codec->time_base);
             }
@@ -1396,7 +1377,7 @@ static int check_output_constraints(InputStream *ist, OutputStream *ost)
     if (ost->source_index != ist_index)
         return 0;
 
-    if (of->start_time != AV_NOPTS_VALUE && ist->pts < of->start_time)
+    if (of->start_time && ist->pts < of->start_time)
         return 0;
 
     return 1;
@@ -1405,10 +1386,8 @@ static int check_output_constraints(InputStream *ist, OutputStream *ost)
 static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *pkt)
 {
     OutputFile *of = output_files[ost->file_index];
-    InputFile   *f = input_files [ist->file_index];
-    int64_t start_time = (of->start_time == AV_NOPTS_VALUE) ? 0 : of->start_time;
-    int64_t ost_tb_start_time = av_rescale_q(start_time, AV_TIME_BASE_Q, ost->st->time_base);
-    int64_t ist_tb_start_time = av_rescale_q(start_time, AV_TIME_BASE_Q, ist->st->time_base);
+    int64_t ost_tb_start_time = av_rescale_q(of->start_time, AV_TIME_BASE_Q, ost->st->time_base);
+    int64_t ist_tb_start_time = av_rescale_q(of->start_time, AV_TIME_BASE_Q, ist->st->time_base);
     AVPicture pict;
     AVPacket opkt;
 
@@ -1419,7 +1398,7 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
         return;
 
     if (pkt->pts == AV_NOPTS_VALUE) {
-        if (!ost->frame_number && ist->pts < start_time &&
+        if (!ost->frame_number && ist->pts < of->start_time &&
             !ost->copy_prior_start)
             return;
     } else {
@@ -1429,21 +1408,11 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
     }
 
     if (of->recording_time != INT64_MAX &&
-        ist->pts >= of->recording_time + start_time) {
+        ist->pts >= of->recording_time + of->start_time) {
         close_output_stream(ost);
         return;
     }
 
-    if (f->recording_time != INT64_MAX) {
-        start_time = f->ctx->start_time;
-        if (f->start_time != AV_NOPTS_VALUE)
-            start_time += f->start_time;
-        if (ist->pts >= f->recording_time + start_time) {
-            close_output_stream(ost);
-            return;
-        }
-    }
-
     /* force the input stream PTS */
     if (ost->st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
         audio_size += pkt->size;
@@ -2120,21 +2089,6 @@ static int transcode_init(void)
     char error[1024];
     int want_sdp = 1;
 
-    for (i = 0; i < nb_filtergraphs; i++) {
-        FilterGraph *fg = filtergraphs[i];
-        for (j = 0; j < fg->nb_outputs; j++) {
-            OutputFilter *ofilter = fg->outputs[j];
-            if (!ofilter->ost || ofilter->ost->source_index >= 0)
-                continue;
-            if (fg->nb_inputs != 1)
-                continue;
-            for (k = nb_input_streams-1; k >= 0 ; k--)
-                if (fg->inputs[0]->ist == input_streams[k])
-                    break;
-            ofilter->ost->source_index = k;
-        }
-    }
-
     /* init framerate emulation */
     for (i = 0; i < nb_input_files; i++) {
         InputFile *ifile = input_files[i];
@@ -2176,15 +2130,6 @@ static int transcode_init(void)
             ost->st->disposition          = ist->st->disposition;
             codec->bits_per_raw_sample    = icodec->bits_per_raw_sample;
             codec->chroma_sample_location = icodec->chroma_sample_location;
-        } else {
-            for (j=0; j<oc->nb_streams; j++) {
-                AVStream *st = oc->streams[j];
-                if (st != ost->st && st->codec->codec_type == codec->codec_type)
-                    break;
-            }
-            if (j == oc->nb_streams)
-                if (codec->codec_type == AVMEDIA_TYPE_AUDIO || codec->codec_type == AVMEDIA_TYPE_VIDEO)
-                    ost->st->disposition = AV_DISPOSITION_DEFAULT;
         }
 
         if (ost->stream_copy) {
@@ -2373,7 +2318,7 @@ static int transcode_init(void)
                 if (ost->filter && !(codec->time_base.num && codec->time_base.den))
                     codec->time_base = ost->filter->filter->inputs[0]->time_base;
                 if (   av_q2d(codec->time_base) < 0.001 && video_sync_method != VSYNC_PASSTHROUGH
-                   && (video_sync_method == VSYNC_CFR || video_sync_method == VSYNC_VSCFR || (video_sync_method == VSYNC_AUTO && !(oc->oformat->flags & AVFMT_VARIABLE_FPS)))){
+                   && (video_sync_method == VSYNC_CFR || (video_sync_method == VSYNC_AUTO && !(oc->oformat->flags & AVFMT_VARIABLE_FPS)))){
                     av_log(oc, AV_LOG_WARNING, "Frame rate very high for a muxer not efficiently supporting it.\n"
                                                "Please consider specifying a lower framerate, a different muxer or -vsync 2\n");
                 }
@@ -2391,17 +2336,10 @@ static int transcode_init(void)
                 if (!strncmp(ost->enc->name, "libx264", 7) &&
                     codec->pix_fmt == AV_PIX_FMT_NONE &&
                     ost->filter->filter->inputs[0]->format != AV_PIX_FMT_YUV420P)
-                    av_log(NULL, AV_LOG_WARNING,
+                    av_log(NULL, AV_LOG_INFO,
                            "No pixel format specified, %s for H.264 encoding chosen.\n"
                            "Use -pix_fmt yuv420p for compatibility with outdated media players.\n",
                            av_get_pix_fmt_name(ost->filter->filter->inputs[0]->format));
-                if (!strncmp(ost->enc->name, "mpeg2video", 10) &&
-                    codec->pix_fmt == AV_PIX_FMT_NONE &&
-                    ost->filter->filter->inputs[0]->format != AV_PIX_FMT_YUV420P)
-                    av_log(NULL, AV_LOG_WARNING,
-                           "No pixel format specified, %s for MPEG-2 encoding chosen.\n"
-                           "Use -pix_fmt yuv420p for compatibility with outdated media players.\n",
-                           av_get_pix_fmt_name(ost->filter->filter->inputs[0]->format));
                 codec->pix_fmt = ost->filter->filter->inputs[0]->format;
 
                 if (!icodec ||
@@ -2721,7 +2659,7 @@ static int check_keyboard_interaction(int64_t cur_time)
         char buf[4096], target[64], command[256], arg[256] = {0};
         double time;
         int k, n = 0;
-        fprintf(stderr, "\nEnter command: <target>|all <time>|-1 <command>[ <argument>]\n");
+        fprintf(stderr, "\nEnter command: <target> <time> <command>[ <argument>]\n");
         i = 0;
         while ((k = read_key()) != '\n' && k != '\r' && i < sizeof(buf)-1)
             if (k > 0)
@@ -2737,10 +2675,7 @@ static int check_keyboard_interaction(int64_t cur_time)
                     if (time < 0) {
                         ret = avfilter_graph_send_command(fg->graph, target, command, arg, buf, sizeof(buf),
                                                           key == 'c' ? AVFILTER_CMD_FLAG_ONE : 0);
-                        fprintf(stderr, "Command reply for stream %d: ret:%d res:\n%s", i, ret, buf);
-                    } else if (key == 'c') {
-                        fprintf(stderr, "Queing commands only on filters supporting the specific command is unsupported\n");
-                        ret = AVERROR_PATCHWELCOME;
+                        fprintf(stderr, "Command reply for stream %d: ret:%d res:%s\n", i, ret, buf);
                     } else {
                         ret = avfilter_graph_queue_command(fg->graph, target, command, arg, 0, time);
                     }
@@ -2777,8 +2712,7 @@ static int check_keyboard_interaction(int64_t cur_time)
                         "?      show this help\n"
                         "+      increase verbosity\n"
                         "-      decrease verbosity\n"
-                        "c      Send command to first matching filter supporting it\n"
-                        "C      Send/Que command to all matching filters\n"
+                        "c      Send command to filtergraph\n"
                         "D      cycle through available debug modes\n"
                         "h      dump packets/hex press to cycle through the 3 states\n"
                         "q      quit\n"
@@ -3083,7 +3017,7 @@ static int process_input(int file_index)
         if(delta < -1LL*dts_delta_threshold*AV_TIME_BASE ||
             (delta > 1LL*dts_delta_threshold*AV_TIME_BASE &&
                 ist->st->codec->codec_type != AVMEDIA_TYPE_SUBTITLE) ||
-            pkt_dts + AV_TIME_BASE/10 < ist->pts){
+            pkt_dts+1<ist->pts){
             ifile->ts_offset -= delta;
             av_log(NULL, AV_LOG_DEBUG,
                    "timestamp discontinuity %"PRId64", new offset= %"PRId64"\n",
@@ -3460,8 +3394,8 @@ int main(int argc, char **argv)
     }
     av_log(NULL, AV_LOG_DEBUG, "%"PRIu64" frames successfully decoded, %"PRIu64" decoding errors\n",
            decode_error_stat[0], decode_error_stat[1]);
-    if ((decode_error_stat[0] + decode_error_stat[1]) * max_error_rate < decode_error_stat[1])
-        exit_program(69);
+    if (2*decode_error_stat[0] < decode_error_stat[1])
+        exit_program(254);
 
     exit_program(received_nb_signals ? 255 : 0);
     return 0;

ffmpeg.h

@@ -51,7 +51,6 @@
 #define VSYNC_PASSTHROUGH 0
 #define VSYNC_CFR         1
 #define VSYNC_VFR         2
-#define VSYNC_VSCFR       0xfe
 #define VSYNC_DROP        0xff
 
 #define MAX_STREAMS 1024    /* arbitrary sanity check value */
@@ -94,7 +93,6 @@ typedef struct OptionsContext {
     /* input options */
     int64_t input_ts_offset;
     int rate_emu;
-    int accurate_seek;
 
     SpecifierOpt *ts_scale;
     int        nb_ts_scale;
@@ -282,16 +280,12 @@ typedef struct InputFile {
     int eof_reached;      /* true if eof reached */
     int eagain;           /* true if last read attempt returned EAGAIN */
     int ist_index;        /* index of first stream in input_streams */
-    int64_t input_ts_offset;
     int64_t ts_offset;
     int64_t last_ts;
-    int64_t start_time;   /* user-specified start time in AV_TIME_BASE or AV_NOPTS_VALUE */
-    int64_t recording_time;
     int nb_streams;       /* number of stream that ffmpeg is aware of; may be different
                              from ctx.nb_streams if new streams appear during av_read_frame() */
     int nb_streams_warn;  /* number of streams that the user was warned of */
     int rate_emu;
-    int accurate_seek;
 
 #if HAVE_PTHREADS
     pthread_t thread;           /* thread reading from this file */
@@ -422,7 +416,6 @@ extern int qp_hist;
 extern int stdin_interaction;
 extern int frame_bits_per_raw_sample;
 extern AVIOContext *progress_avio;
-extern float max_error_rate;
 
 extern const AVIOInterruptCB int_cb;
 

ffmpeg_filter.c

@@ -92,11 +92,6 @@ void choose_sample_fmt(AVStream *st, AVCodec *codec)
 
 static char *choose_pix_fmts(OutputStream *ost)
 {
-    AVDictionaryEntry *strict_dict = av_dict_get(ost->opts, "strict", NULL, 0);
-    if (strict_dict)
-        // used by choose_pixel_fmt() and below
-        av_opt_set(ost->st->codec, "strict", strict_dict->value, 0);
-
      if (ost->keep_pix_fmt) {
         if (ost->filter)
             avfilter_graph_set_auto_convert(ost->filter->graph->graph,
@@ -281,18 +276,21 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
     ist->filters[ist->nb_filters - 1] = fg->inputs[fg->nb_inputs - 1];
 }
 
-static int insert_trim(int64_t start_time, int64_t duration,
-                       AVFilterContext **last_filter, int *pad_idx,
-                       const char *filter_name)
+static int insert_trim(OutputStream *ost, AVFilterContext **last_filter, int *pad_idx)
 {
+    OutputFile *of = output_files[ost->file_index];
     AVFilterGraph *graph = (*last_filter)->graph;
     AVFilterContext *ctx;
     const AVFilter *trim;
-    enum AVMediaType type = avfilter_pad_get_type((*last_filter)->output_pads, *pad_idx);
-    const char *name = (type == AVMEDIA_TYPE_VIDEO) ? "trim" : "atrim";
+    const char *name = ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO ? "trim" : "atrim";
+    char filter_name[128];
     int ret = 0;
 
-    if (duration == INT64_MAX && start_time == AV_NOPTS_VALUE)
+    if (of->recording_time == INT64_MAX && !of->start_time)
+        return 0;
+
+    // Use with duration and without output starttime is buggy with trim filters
+    if (!of->start_time)
         return 0;
 
     trim = avfilter_get_by_name(name);
@@ -302,16 +300,18 @@ static int insert_trim(int64_t start_time, int64_t duration,
         return AVERROR_FILTER_NOT_FOUND;
     }
 
+    snprintf(filter_name, sizeof(filter_name), "%s for output stream %d:%d",
+             name, ost->file_index, ost->index);
     ctx = avfilter_graph_alloc_filter(graph, trim, filter_name);
     if (!ctx)
         return AVERROR(ENOMEM);
 
-    if (duration != INT64_MAX) {
-        ret = av_opt_set_int(ctx, "durationi", duration,
+    if (of->recording_time != INT64_MAX) {
+        ret = av_opt_set_double(ctx, "duration", (double)of->recording_time / 1e6,
                                 AV_OPT_SEARCH_CHILDREN);
     }
-    if (ret >= 0 && start_time != AV_NOPTS_VALUE) {
-        ret = av_opt_set_int(ctx, "starti", start_time,
+    if (ret >= 0 && of->start_time) {
+        ret = av_opt_set_double(ctx, "start", (double)of->start_time / 1e6,
                                 AV_OPT_SEARCH_CHILDREN);
     }
     if (ret < 0) {
@@ -336,7 +336,6 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter,
 {
     char *pix_fmts;
     OutputStream *ost = ofilter->ost;
-    OutputFile    *of = output_files[ost->file_index];
     AVCodecContext *codec = ost->st->codec;
     AVFilterContext *last_filter = out->filter_ctx;
     int pad_idx = out->pad_idx;
@@ -409,10 +408,7 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter,
         pad_idx = 0;
     }
 
-    snprintf(name, sizeof(name), "trim for output stream %d:%d",
-             ost->file_index, ost->index);
-    ret = insert_trim(of->start_time, of->recording_time,
-                      &last_filter, &pad_idx, name);
+    ret = insert_trim(ost, &last_filter, &pad_idx);
     if (ret < 0)
         return ret;
 
@@ -426,9 +422,9 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter,
 static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out)
 {
     OutputStream *ost = ofilter->ost;
-    OutputFile    *of = output_files[ost->file_index];
     AVCodecContext *codec  = ost->st->codec;
     AVFilterContext *last_filter = out->filter_ctx;
+    OutputFile *of = output_files[ost->file_index];
     int pad_idx = out->pad_idx;
     char *sample_fmts, *sample_rates, *channel_layouts;
     char name[255];
@@ -538,10 +534,7 @@ static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter,
         }
     }
 
-    snprintf(name, sizeof(name), "trim for output stream %d:%d",
-             ost->file_index, ost->index);
-    ret = insert_trim(of->start_time, of->recording_time,
-                      &last_filter, &pad_idx, name);
+    ret = insert_trim(ost, &last_filter, &pad_idx);
     if (ret < 0)
         return ret;
 
@@ -555,7 +548,7 @@ static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter,
 {                                                                  \
     AVFilterContext *ctx = inout->filter_ctx;                      \
     AVFilterPad *pads = in ? ctx->input_pads  : ctx->output_pads;  \
-    int       nb_pads = in ? ctx->nb_inputs   : ctx->nb_outputs;   \
+    int       nb_pads = in ? ctx->input_count : ctx->output_count; \
     AVIOContext *pb;                                               \
                                                                    \
     if (avio_open_dyn_buf(&pb) < 0)                                \
@@ -619,22 +612,17 @@ static int sub2video_prepare(InputStream *ist)
 static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
                                         AVFilterInOut *in)
 {
-    AVFilterContext *last_filter;
-    const AVFilter *buffer_filt = avfilter_get_by_name("buffer");
+    AVFilterContext *first_filter = in->filter_ctx;
+    AVFilter *filter = avfilter_get_by_name("buffer");
     InputStream *ist = ifilter->ist;
-    InputFile     *f = input_files[ist->file_index];
     AVRational tb = ist->framerate.num ? av_inv_q(ist->framerate) :
                                          ist->st->time_base;
     AVRational fr = ist->framerate;
     AVRational sar;
     AVBPrint args;
     char name[255];
-    int ret, pad_idx = 0;
-
-    if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot connect video filter to audio input\n");
-        return AVERROR(EINVAL);
-    }
+    int pad_idx = in->pad_idx;
+    int ret;
 
     if (!fr.num)
         fr = av_guess_frame_rate(input_files[ist->file_index]->ctx, ist->st, NULL);
@@ -662,10 +650,9 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
     snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index,
              ist->file_index, ist->st->index);
 
-    if ((ret = avfilter_graph_create_filter(&ifilter->filter, buffer_filt, name,
+    if ((ret = avfilter_graph_create_filter(&ifilter->filter, filter, name,
                                             args.str, NULL, fg->graph)) < 0)
         return ret;
-    last_filter = ifilter->filter;
 
     if (ist->framerate.num) {
         AVFilterContext *setpts;
@@ -678,10 +665,11 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
                                                 fg->graph)) < 0)
             return ret;
 
-        if ((ret = avfilter_link(last_filter, 0, setpts, 0)) < 0)
+        if ((ret = avfilter_link(setpts, 0, first_filter, pad_idx)) < 0)
             return ret;
 
-        last_filter = setpts;
+        first_filter = setpts;
+        pad_idx = 0;
     }
 
     if (do_deinterlace) {
@@ -695,20 +683,14 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
                                                 fg->graph)) < 0)
             return ret;
 
-        if ((ret = avfilter_link(last_filter, 0, yadif, 0)) < 0)
+        if ((ret = avfilter_link(yadif, 0, first_filter, pad_idx)) < 0)
             return ret;
 
-        last_filter = yadif;
+        first_filter = yadif;
+        pad_idx = 0;
     }
 
-    snprintf(name, sizeof(name), "trim for input stream %d:%d",
-             ist->file_index, ist->st->index);
-    ret = insert_trim(((f->start_time == AV_NOPTS_VALUE) || !f->accurate_seek) ?
-                      AV_NOPTS_VALUE : 0, f->recording_time, &last_filter, &pad_idx, name);
-    if (ret < 0)
-        return ret;
-
-    if ((ret = avfilter_link(last_filter, 0, in->filter_ctx, in->pad_idx)) < 0)
+    if ((ret = avfilter_link(ifilter->filter, 0, first_filter, pad_idx)) < 0)
         return ret;
     return 0;
 }
@@ -716,18 +698,13 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
 static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
                                         AVFilterInOut *in)
 {
-    AVFilterContext *last_filter;
-    const AVFilter *abuffer_filt = avfilter_get_by_name("abuffer");
+    AVFilterContext *first_filter = in->filter_ctx;
+    AVFilter *filter = avfilter_get_by_name("abuffer");
     InputStream *ist = ifilter->ist;
-    InputFile     *f = input_files[ist->file_index];
+    int pad_idx = in->pad_idx;
     AVBPrint args;
     char name[255];
-    int ret, pad_idx = 0;
-
-    if (ist->st->codec->codec_type != AVMEDIA_TYPE_AUDIO) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot connect audio filter to non audio input\n");
-        return AVERROR(EINVAL);
-    }
+    int ret;
 
     av_bprint_init(&args, 0, AV_BPRINT_SIZE_AUTOMATIC);
     av_bprintf(&args, "time_base=%d/%d:sample_rate=%d:sample_fmt=%s",
@@ -742,11 +719,10 @@ static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
     snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index,
              ist->file_index, ist->st->index);
 
-    if ((ret = avfilter_graph_create_filter(&ifilter->filter, abuffer_filt,
+    if ((ret = avfilter_graph_create_filter(&ifilter->filter, filter,
                                             name, args.str, NULL,
                                             fg->graph)) < 0)
         return ret;
-    last_filter = ifilter->filter;
 
 #define AUTO_INSERT_FILTER_INPUT(opt_name, filter_name, arg) do {                 \
     AVFilterContext *filt_ctx;                                              \
@@ -762,11 +738,11 @@ static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
     if (ret < 0)                                                            \
         return ret;                                                         \
                                                                             \
-    ret = avfilter_link(last_filter, 0, filt_ctx, 0);                       \
+    ret = avfilter_link(filt_ctx, 0, first_filter, pad_idx);                \
     if (ret < 0)                                                            \
         return ret;                                                         \
                                                                             \
-    last_filter = filt_ctx;                                                 \
+    first_filter = filt_ctx;                                                  \
 } while (0)
 
     if (audio_sync_method > 0) {
@@ -802,15 +778,7 @@ static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
         snprintf(args, sizeof(args), "%f", audio_volume / 256.);
         AUTO_INSERT_FILTER_INPUT("-vol", "volume", args);
     }
-
-    snprintf(name, sizeof(name), "trim for input stream %d:%d",
-             ist->file_index, ist->st->index);
-    ret = insert_trim(((f->start_time == AV_NOPTS_VALUE) || !f->accurate_seek) ?
-                      AV_NOPTS_VALUE : 0, f->recording_time, &last_filter, &pad_idx, name);
-    if (ret < 0)
-        return ret;
-
-    if ((ret = avfilter_link(last_filter, 0, in->filter_ctx, in->pad_idx)) < 0)
+    if ((ret = avfilter_link(ifilter->filter, 0, first_filter, pad_idx)) < 0)
         return ret;
 
     return 0;
@@ -865,10 +833,6 @@ int configure_filtergraph(FilterGraph *fg)
         if (strlen(args))
             args[strlen(args) - 1] = '\0';
         fg->graph->resample_lavr_opts = av_strdup(args);
-
-        e = av_dict_get(ost->opts, "threads", NULL, 0);
-        if (e)
-            av_opt_set(fg->graph, "threads", e->value, 0);
     }
 
     if ((ret = avfilter_graph_parse2(fg->graph, graph_desc, &inputs, &outputs)) < 0)

ffmpeg_opt.c

@@ -84,7 +84,6 @@ int print_stats       = -1;
 int qp_hist           = 0;
 int stdin_interaction = 1;
 int frame_bits_per_raw_sample = 0;
-float max_error_rate  = 2.0/3;
 
 
 static int intra_only         = 0;
@@ -96,7 +95,9 @@ static int do_psnr            = 0;
 static int input_sync;
 static int override_ffserver  = 0;
 
-static void uninit_options(OptionsContext *o)
+static int64_t recording_time = INT64_MAX;
+
+static void uninit_options(OptionsContext *o, int is_input)
 {
     const OptionDef *po = options;
     int i;
@@ -126,19 +127,28 @@ static void uninit_options(OptionsContext *o)
     av_freep(&o->audio_channel_maps);
     av_freep(&o->streamid_map);
     av_freep(&o->attachments);
+
+    if (is_input)
+        recording_time = o->recording_time;
+    else
+        recording_time = INT64_MAX;
 }
 
-static void init_options(OptionsContext *o)
+static void init_options(OptionsContext *o, int is_input)
 {
     memset(o, 0, sizeof(*o));
 
+    if (!is_input && recording_time != INT64_MAX) {
+        o->recording_time = recording_time;
+        av_log(NULL, AV_LOG_WARNING,
+                "-t is not an input option, keeping it for the next output;"
+                " consider fixing your command line.\n");
+    } else
+        o->recording_time = INT64_MAX;
     o->stop_time = INT64_MAX;
     o->mux_max_delay  = 0.7;
-    o->start_time     = AV_NOPTS_VALUE;
-    o->recording_time = INT64_MAX;
     o->limit_filesize = UINT64_MAX;
     o->chapters_input_file = INT_MAX;
-    o->accurate_seek  = 1;
 }
 
 /* return a copy of the input with the stream specifiers removed from the keys */
@@ -593,7 +603,7 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
         case AVMEDIA_TYPE_VIDEO:
             if(!ist->dec)
                 ist->dec = avcodec_find_decoder(dec->codec_id);
-            if (av_codec_get_lowres(dec)) {
+            if (dec->lowres) {
                 dec->flags |= CODEC_FLAG_EMU_EDGE;
             }
 
@@ -649,16 +659,11 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
 
 static void assert_file_overwrite(const char *filename)
 {
-    if (file_overwrite && no_file_overwrite) {
-        fprintf(stderr, "Error, both -y and -n supplied. Exiting.\n");
-        exit_program(1);
-    }
-
-    if (!file_overwrite &&
+    if ((!file_overwrite || no_file_overwrite) &&
         (strchr(filename, ':') == NULL || filename[1] == ':' ||
          av_strstart(filename, "file:", NULL))) {
         if (avio_check(filename, 0) == 0) {
-            if (stdin_interaction && !no_file_overwrite) {
+            if (stdin_interaction && (!no_file_overwrite || file_overwrite)) {
                 fprintf(stderr,"File '%s' already exists. Overwrite ? [y/N] ", filename);
                 fflush(stderr);
                 term_exit();
@@ -786,14 +791,6 @@ static int open_input_file(OptionsContext *o, const char *filename)
         find_codec_or_die(audio_codec_name   , AVMEDIA_TYPE_AUDIO   , 0)->id : AV_CODEC_ID_NONE;
     ic->subtitle_codec_id= subtitle_codec_name ?
         find_codec_or_die(subtitle_codec_name, AVMEDIA_TYPE_SUBTITLE, 0)->id : AV_CODEC_ID_NONE;
-
-    if (video_codec_name)
-        av_format_set_video_codec   (ic, find_codec_or_die(video_codec_name   , AVMEDIA_TYPE_VIDEO   , 0));
-    if (audio_codec_name)
-        av_format_set_audio_codec   (ic, find_codec_or_die(audio_codec_name   , AVMEDIA_TYPE_AUDIO   , 0));
-    if (subtitle_codec_name)
-        av_format_set_subtitle_codec(ic, find_codec_or_die(subtitle_codec_name, AVMEDIA_TYPE_SUBTITLE, 0));
-
     ic->flags |= AVFMT_FLAG_NONBLOCK;
     ic->interrupt_callback = int_cb;
 
@@ -822,13 +819,13 @@ static int open_input_file(OptionsContext *o, const char *filename)
         exit_program(1);
     }
 
-    timestamp = (o->start_time == AV_NOPTS_VALUE) ? 0 : o->start_time;
+    timestamp = o->start_time;
     /* add the stream start time */
     if (ic->start_time != AV_NOPTS_VALUE)
         timestamp += ic->start_time;
 
     /* if seeking requested, we execute it */
-    if (o->start_time != AV_NOPTS_VALUE) {
+    if (o->start_time != 0) {
         ret = avformat_seek_file(ic, -1, INT64_MIN, timestamp, timestamp, 0);
         if (ret < 0) {
             av_log(NULL, AV_LOG_WARNING, "%s: could not seek to position %0.3f\n",
@@ -850,13 +847,9 @@ static int open_input_file(OptionsContext *o, const char *filename)
 
     f->ctx        = ic;
     f->ist_index  = nb_input_streams - ic->nb_streams;
-    f->start_time = o->start_time;
-    f->recording_time = o->recording_time;
-    f->input_ts_offset = o->input_ts_offset;
     f->ts_offset  = o->input_ts_offset - (copy_ts ? 0 : timestamp);
     f->nb_streams = ic->nb_streams;
     f->rate_emu   = o->rate_emu;
-    f->accurate_seek = o->accurate_seek;
 
     /* check if all codec options have been used */
     unused_opts = strip_specifiers(o->g->codec_opts);
@@ -1453,8 +1446,7 @@ static int copy_chapters(InputFile *ifile, OutputFile *ofile, int copy_metadata)
 
     for (i = 0; i < is->nb_chapters; i++) {
         AVChapter *in_ch = is->chapters[i], *out_ch;
-        int64_t start_time = (ofile->start_time == AV_NOPTS_VALUE) ? 0 : ofile->start_time;
-        int64_t ts_off   = av_rescale_q(start_time - ifile->ts_offset,
+        int64_t ts_off   = av_rescale_q(ofile->start_time - ifile->ts_offset,
                                        AV_TIME_BASE_Q, in_ch->time_base);
         int64_t rt       = (ofile->recording_time == INT64_MAX) ? INT64_MAX :
                            av_rescale_q(ofile->recording_time, AV_TIME_BASE_Q, in_ch->time_base);
@@ -1589,12 +1581,11 @@ static int open_output_file(OptionsContext *o, const char *filename)
     }
 
     if (o->stop_time != INT64_MAX && o->recording_time == INT64_MAX) {
-        int64_t start_time = o->start_time == AV_NOPTS_VALUE ? 0 : o->start_time;
-        if (o->stop_time <= start_time) {
+        if (o->stop_time <= o->start_time) {
             av_log(NULL, AV_LOG_WARNING, "-to value smaller than -ss; ignoring -to.\n");
             o->stop_time = INT64_MAX;
         } else {
-            o->recording_time = o->stop_time - start_time;
+            o->recording_time = o->stop_time - o->start_time;
         }
     }
 
@@ -2458,9 +2449,7 @@ void show_help_default(const char *opt, const char *arg)
         int flags = AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_ENCODING_PARAM;
         show_help_children(avcodec_get_class(), flags);
         show_help_children(avformat_get_class(), flags);
-#if CONFIG_SWSCALE
         show_help_children(sws_get_class(), flags);
-#endif
         show_help_children(swr_get_class(), AV_OPT_FLAG_AUDIO_PARAM);
         show_help_children(avfilter_get_class(), AV_OPT_FLAG_VIDEO_PARAM | AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM);
     }
@@ -2492,7 +2481,7 @@ static int open_files(OptionGroupList *l, const char *inout,
         OptionGroup *g = &l->groups[i];
         OptionsContext o;
 
-        init_options(&o);
+        init_options(&o, !strcmp(inout, "input"));
         o.g = g;
 
         ret = parse_optgroup(&o, g);
@@ -2504,7 +2493,7 @@ static int open_files(OptionGroupList *l, const char *inout,
 
         av_log(NULL, AV_LOG_DEBUG, "Opening an %s file: %s.\n", inout, g->arg);
         ret = open_file(&o, g->arg);
-        uninit_options(&o);
+        uninit_options(&o, !strcmp(inout, "input"));
         if (ret < 0) {
             av_log(NULL, AV_LOG_ERROR, "Error opening %s file %s.\n",
                    inout, g->arg);
@@ -2589,7 +2578,7 @@ const OptionDef options[] = {
     { "y",              OPT_BOOL,                                    {              &file_overwrite },
         "overwrite output files" },
     { "n",              OPT_BOOL,                                    {              &no_file_overwrite },
-        "never overwrite output files" },
+        "do not overwrite output files" },
     { "c",              HAS_ARG | OPT_STRING | OPT_SPEC |
                         OPT_INPUT | OPT_OUTPUT,                      { .off       = OFFSET(codec_names) },
         "codec name", "codec" },
@@ -2612,8 +2601,7 @@ const OptionDef options[] = {
     { "map_chapters",   HAS_ARG | OPT_INT | OPT_EXPERT | OPT_OFFSET |
                         OPT_OUTPUT,                                  { .off = OFFSET(chapters_input_file) },
         "set chapters mapping", "input_file_index" },
-    { "t",              HAS_ARG | OPT_TIME | OPT_OFFSET |
-                        OPT_INPUT | OPT_OUTPUT,                      { .off = OFFSET(recording_time) },
+    { "t",              HAS_ARG | OPT_TIME | OPT_OFFSET | OPT_INPUT | OPT_OUTPUT, { .off = OFFSET(recording_time) },
         "record or transcode \"duration\" seconds of audio/video",
         "duration" },
     { "to",             HAS_ARG | OPT_TIME | OPT_OFFSET | OPT_OUTPUT,  { .off = OFFSET(stop_time) },
@@ -2623,9 +2611,6 @@ const OptionDef options[] = {
     { "ss",             HAS_ARG | OPT_TIME | OPT_OFFSET |
                         OPT_INPUT | OPT_OUTPUT,                      { .off = OFFSET(start_time) },
         "set the start time offset", "time_off" },
-    { "accurate_seek",  OPT_BOOL | OPT_OFFSET | OPT_EXPERT |
-                        OPT_INPUT,                                   { .off = OFFSET(accurate_seek) },
-        "enable/disable accurate seeking with -ss" },
     { "itsoffset",      HAS_ARG | OPT_TIME | OPT_OFFSET |
                         OPT_EXPERT | OPT_INPUT,                      { .off = OFFSET(input_ts_offset) },
         "set the input ts offset", "time_off" },
@@ -2721,8 +2706,6 @@ const OptionDef options[] = {
         "extract an attachment into a file", "filename" },
     { "debug_ts",       OPT_BOOL | OPT_EXPERT,                       { &debug_ts },
         "print timestamp debugging info" },
-    { "max_error_rate",  HAS_ARG | OPT_FLOAT,                        { &max_error_rate },
-        "maximum error rate", "ratio of errors (0.0: no errors, 1.0: 100% errors) above which ffmpeg returns an error instead of success." },
 
     /* video options */
     { "vframes",      OPT_VIDEO | HAS_ARG  | OPT_PERFILE | OPT_OUTPUT,           { .func_arg = opt_video_frames },

ffplay.c

@@ -177,8 +177,6 @@ typedef struct VideoState {
     int read_pause_return;
     AVFormatContext *ic;
     int realtime;
-    int audio_finished;
-    int video_finished;
 
     Clock audclk;
     Clock vidclk;
@@ -218,7 +216,6 @@ typedef struct VideoState {
     int frame_drops_early;
     int frame_drops_late;
     AVFrame *frame;
-    int64_t audio_frame_next_pts;
 
     enum ShowMode {
         SHOW_MODE_NONE = -1, SHOW_MODE_VIDEO = 0, SHOW_MODE_WAVES, SHOW_MODE_RDFT, SHOW_MODE_NB
@@ -308,6 +305,7 @@ static int workaround_bugs = 1;
 static int fast = 0;
 static int genpts = 0;
 static int lowres = 0;
+static int idct = FF_IDCT_AUTO;
 static int error_concealment = 3;
 static int decoder_reorder_pts = -1;
 static int autoexit;
@@ -407,16 +405,6 @@ static int packet_queue_put(PacketQueue *q, AVPacket *pkt)
     return ret;
 }
 
-static int packet_queue_put_nullpacket(PacketQueue *q, int stream_index)
-{
-    AVPacket pkt1, *pkt = &pkt1;
-    av_init_packet(pkt);
-    pkt->data = NULL;
-    pkt->size = 0;
-    pkt->stream_index = stream_index;
-    return packet_queue_put(q, pkt);
-}
-
 /* packet queue handling */
 static void packet_queue_init(PacketQueue *q)
 {
@@ -785,14 +773,6 @@ static void blend_subrect(AVPicture *dst, const AVSubtitleRect *rect, int imgw,
     }
 }
 
-static void free_picture(VideoPicture *vp)
-{
-     if (vp->bmp) {
-         SDL_FreeYUVOverlay(vp->bmp);
-         vp->bmp = NULL;
-     }
-}
-
 static void free_subpicture(SubPicture *sp)
 {
     avsubtitle_free(&sp->sub);
@@ -995,8 +975,7 @@ static void video_audio_display(VideoState *s)
                 }
                 av_rdft_calc(s->rdft, data[ch]);
             }
-            /* Least efficient way to do this, we should of course
-             * directly access it but it is more than fast enough. */
+            // least efficient way to do this, we should of course directly access it but its more than fast enough
             for (y = 0; y < s->height; y++) {
                 double w = 1 / sqrt(nb_freq);
                 int a = sqrt(w * sqrt(data[0][2 * y + 0] * data[0][2 * y + 0] + data[0][2 * y + 1] * data[0][2 * y + 1]));
@@ -1021,6 +1000,7 @@ static void video_audio_display(VideoState *s)
 
 static void stream_close(VideoState *is)
 {
+    VideoPicture *vp;
     int i;
     /* XXX: use a special url_shutdown call to abort parse cleanly */
     is->abort_request = 1;
@@ -1030,10 +1010,13 @@ static void stream_close(VideoState *is)
     packet_queue_destroy(&is->subtitleq);
 
     /* free all pictures */
-    for (i = 0; i < VIDEO_PICTURE_QUEUE_SIZE; i++)
-        free_picture(&is->pictq[i]);
-    for (i = 0; i < SUBPICTURE_QUEUE_SIZE; i++)
-        free_subpicture(&is->subpq[i]);
+    for (i = 0; i < VIDEO_PICTURE_QUEUE_SIZE; i++) {
+        vp = &is->pictq[i];
+        if (vp->bmp) {
+            SDL_FreeYUVOverlay(vp->bmp);
+            vp->bmp = NULL;
+        }
+    }
     SDL_DestroyMutex(is->pictq_mutex);
     SDL_DestroyCond(is->pictq_cond);
     SDL_DestroyMutex(is->subpq_mutex);
@@ -1507,7 +1490,8 @@ static void alloc_picture(VideoState *is)
 
     vp = &is->pictq[is->pictq_windex];
 
-    free_picture(vp);
+    if (vp->bmp)
+        SDL_FreeYUVOverlay(vp->bmp);
 
     video_open(is, 0, vp);
 
@@ -1686,9 +1670,6 @@ static int get_video_frame(VideoState *is, AVFrame *frame, AVPacket *pkt, int *s
     if(avcodec_decode_video2(is->video_st->codec, frame, &got_picture, pkt) < 0)
         return 0;
 
-    if (!got_picture && !pkt->data)
-        is->video_finished = *serial;
-
     if (got_picture) {
         int ret = 1;
         double dpts = NAN;
@@ -1715,7 +1696,7 @@ static int get_video_frame(VideoState *is, AVFrame *frame, AVPacket *pkt, int *s
                     !isnan(ptsdiff) && ptsdiff > 0 && ptsdiff < AV_NOSYNC_THRESHOLD &&
                     clockdiff + ptsdiff - is->frame_last_filter_delay < 0 &&
                     is->videoq.nb_packets) {
-                    is->frame_last_dropped_pos = av_frame_get_pkt_pos(frame);
+                    is->frame_last_dropped_pos = pkt->pos;
                     is->frame_last_dropped_pts = dpts;
                     is->frame_last_dropped_serial = *serial;
                     is->frame_drops_early++;
@@ -1833,8 +1814,6 @@ static int configure_audio_filters(VideoState *is, const char *afilters, int for
     int64_t channel_layouts[2] = { 0, -1 };
     int channels[2] = { 0, -1 };
     AVFilterContext *filt_asrc = NULL, *filt_asink = NULL;
-    char aresample_swr_opts[512] = "";
-    AVDictionaryEntry *e = NULL;
     char asrc_args[256];
     int ret;
 
@@ -1842,12 +1821,6 @@ static int configure_audio_filters(VideoState *is, const char *afilters, int for
     if (!(is->agraph = avfilter_graph_alloc()))
         return AVERROR(ENOMEM);
 
-    while ((e = av_dict_get(swr_opts, "", e, AV_DICT_IGNORE_SUFFIX)))
-        av_strlcatf(aresample_swr_opts, sizeof(aresample_swr_opts), "%s=%s:", e->key, e->value);
-    if (strlen(aresample_swr_opts))
-        aresample_swr_opts[strlen(aresample_swr_opts)-1] = '\0';
-    av_opt_set(is->agraph, "aresample_swr_opts", aresample_swr_opts, 0);
-
     ret = snprintf(asrc_args, sizeof(asrc_args),
                    "sample_rate=%d:sample_fmt=%s:channels=%d:time_base=%d/%d",
                    is->audio_filter_src.freq, av_get_sample_fmt_name(is->audio_filter_src.fmt),
@@ -1975,8 +1948,6 @@ static int video_thread(void *arg)
 
             ret = av_buffersink_get_frame_flags(filt_out, frame, 0);
             if (ret < 0) {
-                if (ret == AVERROR_EOF)
-                    is->video_finished = serial;
                 ret = 0;
                 break;
             }
@@ -1991,7 +1962,7 @@ static int video_thread(void *arg)
         }
 #else
         pts = (frame->pts == AV_NOPTS_VALUE) ? NAN : frame->pts * av_q2d(is->video_st->time_base);
-        ret = queue_picture(is, frame, pts, av_frame_get_pkt_pos(frame), serial);
+        ret = queue_picture(is, frame, pts, pkt.pos, serial);
         av_frame_unref(frame);
 #endif
 
@@ -2074,8 +2045,6 @@ static int subtitle_thread(void *arg)
             SDL_LockMutex(is->subpq_mutex);
             is->subpq_size++;
             SDL_UnlockMutex(is->subpq_mutex);
-        } else if (got_subtitle) {
-            avsubtitle_free(&sp->sub);
         }
         av_free_packet(pkt);
     }
@@ -2160,6 +2129,8 @@ static int audio_decode_frame(VideoState *is)
     int64_t dec_channel_layout;
     int got_frame;
     av_unused double audio_clock0;
+    int new_packet = 0;
+    int flush_complete = 0;
     int wanted_nb_samples;
     AVRational tb;
     int ret;
@@ -2167,7 +2138,7 @@ static int audio_decode_frame(VideoState *is)
 
     for (;;) {
         /* NOTE: the audio packet can contain several frames */
-        while (pkt_temp->stream_index != -1 || is->audio_buf_frames_pending) {
+        while (pkt_temp->size > 0 || (!pkt_temp->data && new_packet) || is->audio_buf_frames_pending) {
             if (!is->frame) {
                 if (!(is->frame = avcodec_alloc_frame()))
                     return AVERROR(ENOMEM);
@@ -2183,6 +2154,9 @@ static int audio_decode_frame(VideoState *is)
                 return -1;
 
             if (!is->audio_buf_frames_pending) {
+                if (flush_complete)
+                    break;
+                new_packet = 0;
                 len1 = avcodec_decode_audio4(dec, is->frame, &got_frame, pkt_temp);
                 if (len1 < 0) {
                     /* if error, we skip the frame */
@@ -2190,32 +2164,23 @@ static int audio_decode_frame(VideoState *is)
                     break;
                 }
 
-                pkt_temp->dts =
-                pkt_temp->pts = AV_NOPTS_VALUE;
                 pkt_temp->data += len1;
                 pkt_temp->size -= len1;
-                if (pkt_temp->data && pkt_temp->size <= 0 || !pkt_temp->data && !got_frame)
-                    pkt_temp->stream_index = -1;
-                if (!pkt_temp->data && !got_frame)
-                    is->audio_finished = is->audio_pkt_temp_serial;
 
-                if (!got_frame)
+                if (!got_frame) {
+                    /* stop sending empty packets if the decoder is finished */
+                    if (!pkt_temp->data && dec->codec->capabilities & CODEC_CAP_DELAY)
+                        flush_complete = 1;
                     continue;
+                }
 
                 tb = (AVRational){1, is->frame->sample_rate};
                 if (is->frame->pts != AV_NOPTS_VALUE)
                     is->frame->pts = av_rescale_q(is->frame->pts, dec->time_base, tb);
                 else if (is->frame->pkt_pts != AV_NOPTS_VALUE)
                     is->frame->pts = av_rescale_q(is->frame->pkt_pts, is->audio_st->time_base, tb);
-                else if (is->audio_frame_next_pts != AV_NOPTS_VALUE)
-#if CONFIG_AVFILTER
-                    is->frame->pts = av_rescale_q(is->audio_frame_next_pts, (AVRational){1, is->audio_filter_src.freq}, tb);
-#else
-                    is->frame->pts = av_rescale_q(is->audio_frame_next_pts, (AVRational){1, is->audio_src.freq}, tb);
-#endif
-
-                if (is->frame->pts != AV_NOPTS_VALUE)
-                    is->audio_frame_next_pts = is->frame->pts + is->frame->nb_samples;
+                if (pkt_temp->pts != AV_NOPTS_VALUE)
+                    pkt_temp->pts += (double) is->frame->nb_samples / is->frame->sample_rate / av_q2d(is->audio_st->time_base);
 
 #if CONFIG_AVFILTER
                 dec_channel_layout = get_valid_channel_layout(is->frame->channel_layout, av_frame_get_channels(is->frame));
@@ -2257,8 +2222,6 @@ static int audio_decode_frame(VideoState *is)
                     is->audio_buf_frames_pending = 0;
                     continue;
                 }
-                if (ret == AVERROR_EOF)
-                    is->audio_finished = is->audio_pkt_temp_serial;
                 return ret;
             }
             is->audio_buf_frames_pending = 1;
@@ -2355,7 +2318,6 @@ static int audio_decode_frame(VideoState *is)
         if (pkt->data)
             av_free_packet(pkt);
         memset(pkt_temp, 0, sizeof(*pkt_temp));
-        pkt_temp->stream_index = -1;
 
         if (is->audioq.abort_request) {
             return -1;
@@ -2365,15 +2327,13 @@ static int audio_decode_frame(VideoState *is)
             SDL_CondSignal(is->continue_read_thread);
 
         /* read next packet */
-        if ((packet_queue_get(&is->audioq, pkt, 1, &is->audio_pkt_temp_serial)) < 0)
+        if ((new_packet = packet_queue_get(&is->audioq, pkt, 1, &is->audio_pkt_temp_serial)) < 0)
             return -1;
 
         if (pkt->data == flush_pkt.data) {
             avcodec_flush_buffers(dec);
+            flush_complete = 0;
             is->audio_buf_frames_pending = 0;
-            is->audio_frame_next_pts = AV_NOPTS_VALUE;
-            if ((is->ic->iformat->flags & (AVFMT_NOBINSEARCH | AVFMT_NOGENSEARCH | AVFMT_NO_BYTE_SEEK)) && !is->ic->iformat->read_seek)
-                is->audio_frame_next_pts = is->audio_st->start_time;
         }
 
         *pkt_temp = *pkt;
@@ -2425,7 +2385,7 @@ static int audio_open(void *opaque, int64_t wanted_channel_layout, int wanted_nb
 {
     SDL_AudioSpec wanted_spec, spec;
     const char *env;
-    static const int next_nb_channels[] = {0, 0, 1, 6, 2, 6, 4, 6};
+    const int next_nb_channels[] = {0, 0, 1, 6, 2, 6, 4, 6};
 
     env = SDL_getenv("SDL_AUDIO_CHANNELS");
     if (env) {
@@ -2490,7 +2450,6 @@ static int stream_component_open(VideoState *is, int stream_index)
     int sample_rate, nb_channels;
     int64_t channel_layout;
     int ret;
-    int stream_lowres = lowres;
 
     if (stream_index < 0 || stream_index >= ic->nb_streams)
         return -1;
@@ -2515,15 +2474,16 @@ static int stream_component_open(VideoState *is, int stream_index)
 
     avctx->codec_id = codec->id;
     avctx->workaround_bugs   = workaround_bugs;
-    if(stream_lowres > av_codec_get_max_lowres(codec)){
+    avctx->lowres            = lowres;
+    if(avctx->lowres > codec->max_lowres){
         av_log(avctx, AV_LOG_WARNING, "The maximum value for lowres supported by the decoder is %d\n",
-                av_codec_get_max_lowres(codec));
-        stream_lowres = av_codec_get_max_lowres(codec);
+                codec->max_lowres);
+        avctx->lowres= codec->max_lowres;
     }
-    av_codec_set_lowres(avctx, stream_lowres);
+    avctx->idct_algo         = idct;
     avctx->error_concealment = error_concealment;
 
-    if(stream_lowres) avctx->flags |= CODEC_FLAG_EMU_EDGE;
+    if(avctx->lowres) avctx->flags |= CODEC_FLAG_EMU_EDGE;
     if (fast)   avctx->flags2 |= CODEC_FLAG2_FAST;
     if(codec->capabilities & CODEC_CAP_DR1)
         avctx->flags |= CODEC_FLAG_EMU_EDGE;
@@ -2531,8 +2491,8 @@ static int stream_component_open(VideoState *is, int stream_index)
     opts = filter_codec_opts(codec_opts, avctx->codec_id, ic, ic->streams[stream_index], codec);
     if (!av_dict_get(opts, "threads", NULL, 0))
         av_dict_set(&opts, "threads", "auto", 0);
-    if (stream_lowres)
-        av_dict_set(&opts, "lowres", av_asprintf("%d", stream_lowres), AV_DICT_DONT_STRDUP_VAL);
+    if (avctx->lowres)
+        av_dict_set(&opts, "lowres", av_asprintf("%d", avctx->lowres), AV_DICT_DONT_STRDUP_VAL);
     if (avctx->codec_type == AVMEDIA_TYPE_VIDEO || avctx->codec_type == AVMEDIA_TYPE_AUDIO)
         av_dict_set(&opts, "refcounted_frames", "1", 0);
     if (avcodec_open2(avctx, codec, &opts) < 0)
@@ -2583,7 +2543,6 @@ static int stream_component_open(VideoState *is, int stream_index)
 
         memset(&is->audio_pkt, 0, sizeof(is->audio_pkt));
         memset(&is->audio_pkt_temp, 0, sizeof(is->audio_pkt_temp));
-        is->audio_pkt_temp.stream_index = -1;
 
         is->audio_stream = stream_index;
         is->audio_st = ic->streams[stream_index];
@@ -2842,8 +2801,7 @@ static int read_thread(void *arg)
     }
 
     if (is->video_stream < 0 && is->audio_stream < 0) {
-        av_log(NULL, AV_LOG_FATAL, "Failed to open file '%s' or configure filtergraph\n",
-               is->filename);
+        av_log(NULL, AV_LOG_FATAL, "%s: could not open codecs\n", is->filename);
         ret = -1;
         goto fail;
     }
@@ -2913,7 +2871,6 @@ static int read_thread(void *arg)
                 if ((ret = av_copy_packet(&copy, &is->video_st->attached_pic)) < 0)
                     goto fail;
                 packet_queue_put(&is->videoq, &copy);
-                packet_queue_put_nullpacket(&is->videoq, is->video_stream);
             }
             is->queue_attachments_req = 0;
         }
@@ -2931,22 +2888,30 @@ static int read_thread(void *arg)
             SDL_UnlockMutex(wait_mutex);
             continue;
         }
-        if (!is->paused &&
-            (!is->audio_st || is->audio_finished == is->audioq.serial) &&
-            (!is->video_st || (is->video_finished == is->videoq.serial && is->pictq_size == 0))) {
-            if (loop != 1 && (!loop || --loop)) {
-                stream_seek(is, start_time != AV_NOPTS_VALUE ? start_time : 0, 0, 0);
-            } else if (autoexit) {
-                ret = AVERROR_EOF;
-                goto fail;
-            }
-        }
         if (eof) {
-            if (is->video_stream >= 0)
-                packet_queue_put_nullpacket(&is->videoq, is->video_stream);
-            if (is->audio_stream >= 0)
-                packet_queue_put_nullpacket(&is->audioq, is->audio_stream);
+            if (is->video_stream >= 0) {
+                av_init_packet(pkt);
+                pkt->data = NULL;
+                pkt->size = 0;
+                pkt->stream_index = is->video_stream;
+                packet_queue_put(&is->videoq, pkt);
+            }
+            if (is->audio_stream >= 0) {
+                av_init_packet(pkt);
+                pkt->data = NULL;
+                pkt->size = 0;
+                pkt->stream_index = is->audio_stream;
+                packet_queue_put(&is->audioq, pkt);
+            }
             SDL_Delay(10);
+            if (is->audioq.size + is->videoq.size + is->subtitleq.size == 0) {
+                if (loop != 1 && (!loop || --loop)) {
+                    stream_seek(is, start_time != AV_NOPTS_VALUE ? start_time : 0, 0, 0);
+                } else if (autoexit) {
+                    ret = AVERROR_EOF;
+                    goto fail;
+                }
+            }
             eof=0;
             continue;
         }
@@ -3053,8 +3018,6 @@ static void stream_cycle_channel(VideoState *is, int codec_type)
     int start_index, stream_index;
     int old_index;
     AVStream *st;
-    AVProgram *p = NULL;
-    int nb_streams = is->ic->nb_streams;
 
     if (codec_type == AVMEDIA_TYPE_VIDEO) {
         start_index = is->last_video_stream;
@@ -3067,22 +3030,8 @@ static void stream_cycle_channel(VideoState *is, int codec_type)
         old_index = is->subtitle_stream;
     }
     stream_index = start_index;
-
-    if (codec_type != AVMEDIA_TYPE_VIDEO && is->video_stream != -1) {
-        p = av_find_program_from_stream(ic, NULL, is->video_stream);
-        if (p) {
-            nb_streams = p->nb_stream_indexes;
-            for (start_index = 0; start_index < nb_streams; start_index++)
-                if (p->stream_index[start_index] == stream_index)
-                    break;
-            if (start_index == nb_streams)
-                start_index = -1;
-            stream_index = start_index;
-        }
-    }
-
     for (;;) {
-        if (++stream_index >= nb_streams)
+        if (++stream_index >= is->ic->nb_streams)
         {
             if (codec_type == AVMEDIA_TYPE_SUBTITLE)
             {
@@ -3096,7 +3045,7 @@ static void stream_cycle_channel(VideoState *is, int codec_type)
         }
         if (stream_index == start_index)
             return;
-        st = is->ic->streams[p ? p->stream_index[stream_index] : stream_index];
+        st = ic->streams[stream_index];
         if (st->codec->codec_type == codec_type) {
             /* check that parameters are OK */
             switch (codec_type) {
@@ -3114,8 +3063,6 @@ static void stream_cycle_channel(VideoState *is, int codec_type)
         }
     }
  the_end:
-    if (p && stream_index != -1)
-        stream_index = p->stream_index[stream_index];
     stream_component_close(is, old_index);
     stream_component_open(is, stream_index);
 }
@@ -3203,11 +3150,6 @@ static void event_loop(VideoState *cur_stream)
             case SDLK_v:
                 stream_cycle_channel(cur_stream, AVMEDIA_TYPE_VIDEO);
                 break;
-            case SDLK_c:
-                stream_cycle_channel(cur_stream, AVMEDIA_TYPE_VIDEO);
-                stream_cycle_channel(cur_stream, AVMEDIA_TYPE_AUDIO);
-                stream_cycle_channel(cur_stream, AVMEDIA_TYPE_SUBTITLE);
-                break;
             case SDLK_t:
                 stream_cycle_channel(cur_stream, AVMEDIA_TYPE_SUBTITLE);
                 break;
@@ -3460,6 +3402,7 @@ static const OptionDef options[] = {
     { "genpts", OPT_BOOL | OPT_EXPERT, { &genpts }, "generate pts", "" },
     { "drp", OPT_INT | HAS_ARG | OPT_EXPERT, { &decoder_reorder_pts }, "let decoder reorder pts 0=off 1=on -1=auto", ""},
     { "lowres", OPT_INT | HAS_ARG | OPT_EXPERT, { &lowres }, "", "" },
+    { "idct", OPT_INT | HAS_ARG | OPT_EXPERT, { &idct }, "set idct algo",  "algo" },
     { "ec", OPT_INT | HAS_ARG | OPT_EXPERT, { &error_concealment }, "set error concealment options",  "bit_mask" },
     { "sync", HAS_ARG | OPT_EXPERT, { .func_arg = opt_sync }, "set audio-video sync. type (type=audio/video/ext)", "type" },
     { "autoexit", OPT_BOOL | OPT_EXPERT, { &autoexit }, "exit at the end", "" },
@@ -3509,10 +3452,9 @@ void show_help_default(const char *opt, const char *arg)
            "q, ESC              quit\n"
            "f                   toggle full screen\n"
            "p, SPC              pause\n"
-           "a                   cycle audio channel in the current program\n"
+           "a                   cycle audio channel\n"
            "v                   cycle video channel\n"
-           "t                   cycle subtitle channel in the current program\n"
-           "c                   cycle program\n"
+           "t                   cycle subtitle channel\n"
            "w                   show audio waves\n"
            "s                   activate frame-step mode\n"
            "left/right          seek backward/forward 10 seconds\n"

ffprobe.c

@@ -37,9 +37,7 @@
 #include "libavutil/pixdesc.h"
 #include "libavutil/dict.h"
 #include "libavutil/libm.h"
-#include "libavutil/parseutils.h"
 #include "libavutil/timecode.h"
-#include "libavutil/timestamp.h"
 #include "libavdevice/avdevice.h"
 #include "libswscale/swscale.h"
 #include "libswresample/swresample.h"
@@ -59,7 +57,6 @@ static int do_show_error   = 0;
 static int do_show_format  = 0;
 static int do_show_frames  = 0;
 static int do_show_packets = 0;
-static int do_show_programs = 0;
 static int do_show_streams = 0;
 static int do_show_stream_disposition = 0;
 static int do_show_data    = 0;
@@ -75,17 +72,6 @@ static int show_private_data            = 1;
 static char *print_format;
 static char *stream_specifier;
 
-typedef struct {
-    int id;             ///< identifier
-    int64_t start, end; ///< start, end in second/AV_TIME_BASE units
-    int has_start, has_end;
-    int start_is_offset, end_is_offset;
-    int duration_frames;
-} ReadInterval;
-
-static ReadInterval *read_intervals;
-static int read_intervals_nb = 0;
-
 /* section structure definition */
 
 #define SECTION_MAX_NB_CHILDREN 10
@@ -122,14 +108,7 @@ typedef enum {
     SECTION_ID_PACKET,
     SECTION_ID_PACKETS,
     SECTION_ID_PACKETS_AND_FRAMES,
-    SECTION_ID_PROGRAM_STREAM_DISPOSITION,
-    SECTION_ID_PROGRAM_STREAM_TAGS,
-    SECTION_ID_PROGRAM,
-    SECTION_ID_PROGRAM_STREAMS,
-    SECTION_ID_PROGRAM_STREAM,
-    SECTION_ID_PROGRAM_TAGS,
     SECTION_ID_PROGRAM_VERSION,
-    SECTION_ID_PROGRAMS,
     SECTION_ID_ROOT,
     SECTION_ID_STREAM,
     SECTION_ID_STREAM_DISPOSITION,
@@ -152,17 +131,10 @@ static struct section sections[] = {
     [SECTION_ID_PACKETS] =            { SECTION_ID_PACKETS, "packets", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PACKET, -1} },
     [SECTION_ID_PACKETS_AND_FRAMES] = { SECTION_ID_PACKETS_AND_FRAMES, "packets_and_frames", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PACKET, -1} },
     [SECTION_ID_PACKET] =             { SECTION_ID_PACKET, "packet", 0, { -1 } },
-    [SECTION_ID_PROGRAM_STREAM_DISPOSITION] = { SECTION_ID_PROGRAM_STREAM_DISPOSITION, "disposition", 0, { -1 }, .unique_name = "program_stream_disposition" },
-    [SECTION_ID_PROGRAM_STREAM_TAGS] =        { SECTION_ID_PROGRAM_STREAM_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "program_stream_tags" },
-    [SECTION_ID_PROGRAM] =                    { SECTION_ID_PROGRAM, "program", 0, { SECTION_ID_PROGRAM_TAGS, SECTION_ID_PROGRAM_STREAMS, -1 } },
-    [SECTION_ID_PROGRAM_STREAMS] =            { SECTION_ID_PROGRAM_STREAMS, "streams", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PROGRAM_STREAM, -1 }, .unique_name = "program_streams" },
-    [SECTION_ID_PROGRAM_STREAM] =             { SECTION_ID_PROGRAM_STREAM, "stream", 0, { SECTION_ID_PROGRAM_STREAM_DISPOSITION, SECTION_ID_PROGRAM_STREAM_TAGS, -1 }, .unique_name = "program_stream" },
-    [SECTION_ID_PROGRAM_TAGS] =               { SECTION_ID_PROGRAM_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "program_tags" },
     [SECTION_ID_PROGRAM_VERSION] =    { SECTION_ID_PROGRAM_VERSION, "program_version", 0, { -1 } },
-    [SECTION_ID_PROGRAMS] =                   { SECTION_ID_PROGRAMS, "programs", SECTION_FLAG_IS_ARRAY, { SECTION_ID_PROGRAM, -1 } },
     [SECTION_ID_ROOT] =               { SECTION_ID_ROOT, "root", SECTION_FLAG_IS_WRAPPER,
-                                        { SECTION_ID_CHAPTERS, SECTION_ID_FORMAT, SECTION_ID_FRAMES, SECTION_ID_PROGRAMS, SECTION_ID_STREAMS,
-                                          SECTION_ID_PACKETS, SECTION_ID_ERROR, SECTION_ID_PROGRAM_VERSION, SECTION_ID_LIBRARY_VERSIONS, -1} },
+                                        { SECTION_ID_CHAPTERS, SECTION_ID_FORMAT, SECTION_ID_FRAMES, SECTION_ID_STREAMS, SECTION_ID_PACKETS,
+                                          SECTION_ID_ERROR, SECTION_ID_PROGRAM_VERSION, SECTION_ID_LIBRARY_VERSIONS, -1} },
     [SECTION_ID_STREAMS] =            { SECTION_ID_STREAMS, "streams", SECTION_FLAG_IS_ARRAY, { SECTION_ID_STREAM, -1 } },
     [SECTION_ID_STREAM] =             { SECTION_ID_STREAM, "stream", 0, { SECTION_ID_STREAM_DISPOSITION, SECTION_ID_STREAM_TAGS, -1 } },
     [SECTION_ID_STREAM_DISPOSITION] = { SECTION_ID_STREAM_DISPOSITION, "disposition", 0, { -1 }, .unique_name = "stream_disposition" },
@@ -703,8 +675,6 @@ typedef struct CompactContext {
     char *escape_mode_str;
     const char * (*escape_str)(AVBPrint *dst, const char *src, const char sep, void *log_ctx);
     int nested_section[SECTION_MAX_NB_LEVELS];
-    int has_nested_elems[SECTION_MAX_NB_LEVELS];
-    int terminate_line[SECTION_MAX_NB_LEVELS];
 } CompactContext;
 
 #undef OFFSET
@@ -752,28 +722,18 @@ static void compact_print_section_header(WriterContext *wctx)
     const struct section *section = wctx->section[wctx->level];
     const struct section *parent_section = wctx->level ?
         wctx->section[wctx->level-1] : NULL;
-    compact->terminate_line[wctx->level] = 1;
-    compact->has_nested_elems[wctx->level] = 0;
 
     av_bprint_clear(&wctx->section_pbuf[wctx->level]);
-    if (!(section->flags & SECTION_FLAG_IS_ARRAY) && parent_section &&
+    if (parent_section &&
         !(parent_section->flags & (SECTION_FLAG_IS_WRAPPER|SECTION_FLAG_IS_ARRAY))) {
         compact->nested_section[wctx->level] = 1;
-        compact->has_nested_elems[wctx->level-1] = 1;
         av_bprintf(&wctx->section_pbuf[wctx->level], "%s%s:",
                    wctx->section_pbuf[wctx->level-1].str,
                    (char *)av_x_if_null(section->element_name, section->name));
         wctx->nb_item[wctx->level] = wctx->nb_item[wctx->level-1];
-    } else {
-        if (parent_section && compact->has_nested_elems[wctx->level-1] &&
-            (section->flags & SECTION_FLAG_IS_ARRAY)) {
-            compact->terminate_line[wctx->level-1] = 0;
-            printf("\n");
-        }
-        if (compact->print_section &&
-            !(section->flags & (SECTION_FLAG_IS_WRAPPER|SECTION_FLAG_IS_ARRAY)))
-            printf("%s%c", section->name, compact->item_sep);
-    }
+    } else if (compact->print_section &&
+        !(section->flags & (SECTION_FLAG_IS_WRAPPER|SECTION_FLAG_IS_ARRAY)))
+       printf("%s%c", section->name, compact->item_sep);
 }
 
 static void compact_print_section_footer(WriterContext *wctx)
@@ -781,7 +741,6 @@ static void compact_print_section_footer(WriterContext *wctx)
     CompactContext *compact = wctx->priv;
 
     if (!compact->nested_section[wctx->level] &&
-        compact->terminate_line[wctx->level] &&
         !(wctx->section[wctx->level]->flags & (SECTION_FLAG_IS_WRAPPER|SECTION_FLAG_IS_ARRAY)))
         printf("\n");
 }
@@ -1580,6 +1539,7 @@ static av_always_inline int process_frame(WriterContext *w,
     AVCodecContext *dec_ctx = fmt_ctx->streams[pkt->stream_index]->codec;
     int ret = 0, got_frame = 0;
 
+    avcodec_get_frame_defaults(frame);
     if (dec_ctx->codec) {
         switch (dec_ctx->codec_type) {
         case AVMEDIA_TYPE_VIDEO:
@@ -1605,98 +1565,16 @@ static av_always_inline int process_frame(WriterContext *w,
     return got_frame;
 }
 
-static void log_read_interval(const ReadInterval *interval, void *log_ctx, int log_level)
-{
-    av_log(log_ctx, log_level, "id:%d", interval->id);
-
-    if (interval->has_start) {
-        av_log(log_ctx, log_level, " start:%s%s", interval->start_is_offset ? "+" : "",
-               av_ts2timestr(interval->start, &AV_TIME_BASE_Q));
-    } else {
-        av_log(log_ctx, log_level, " start:N/A");
-    }
-
-    if (interval->has_end) {
-        av_log(log_ctx, log_level, " end:%s", interval->end_is_offset ? "+" : "");
-        if (interval->duration_frames)
-            av_log(log_ctx, log_level, "#%"PRId64, interval->end);
-        else
-            av_log(log_ctx, log_level, "%s", av_ts2timestr(interval->end, &AV_TIME_BASE_Q));
-    } else {
-        av_log(log_ctx, log_level, " end:N/A");
-    }
-
-    av_log(log_ctx, log_level, "\n");
-}
-
-static int read_interval_packets(WriterContext *w, AVFormatContext *fmt_ctx,
-                                 const ReadInterval *interval, int64_t *cur_ts)
+static void read_packets(WriterContext *w, AVFormatContext *fmt_ctx)
 {
     AVPacket pkt, pkt1;
-    AVFrame *frame = NULL;
-    int ret = 0, i = 0, frame_count = 0;
-    int64_t start = -INT64_MAX, end = interval->end;
-    int has_start = 0, has_end = interval->has_end && !interval->end_is_offset;
+    AVFrame frame;
+    int i = 0;
 
     av_init_packet(&pkt);
 
-    av_log(NULL, AV_LOG_VERBOSE, "Processing read interval ");
-    log_read_interval(interval, NULL, AV_LOG_VERBOSE);
-
-    if (interval->has_start) {
-        int64_t target;
-        if (interval->start_is_offset) {
-            if (*cur_ts == AV_NOPTS_VALUE) {
-                av_log(NULL, AV_LOG_ERROR,
-                       "Could not seek to relative position since current "
-                       "timestamp is not defined\n");
-                ret = AVERROR(EINVAL);
-                goto end;
-            }
-            target = *cur_ts + interval->start;
-        } else {
-            target = interval->start;
-        }
-
-        av_log(NULL, AV_LOG_VERBOSE, "Seeking to read interval start point %s\n",
-               av_ts2timestr(target, &AV_TIME_BASE_Q));
-        if ((ret = avformat_seek_file(fmt_ctx, -1, -INT64_MAX, target, INT64_MAX, 0)) < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Could not seek to position %"PRId64": %s\n",
-                   interval->start, av_err2str(ret));
-            goto end;
-        }
-    }
-
-    frame = av_frame_alloc();
-    if (!frame) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
     while (!av_read_frame(fmt_ctx, &pkt)) {
         if (selected_streams[pkt.stream_index]) {
-            AVRational tb = fmt_ctx->streams[pkt.stream_index]->time_base;
-
-            if (pkt.pts != AV_NOPTS_VALUE)
-                *cur_ts = av_rescale_q(pkt.pts, tb, AV_TIME_BASE_Q);
-
-            if (!has_start && *cur_ts != AV_NOPTS_VALUE) {
-                start = *cur_ts;
-                has_start = 1;
-            }
-
-            if (has_start && !has_end && interval->end_is_offset) {
-                end = start + interval->end;
-                has_end = 1;
-            }
-
-            if (interval->end_is_offset && interval->duration_frames) {
-                if (frame_count >= interval->end)
-                    break;
-            } else if (has_end && *cur_ts != AV_NOPTS_VALUE && *cur_ts >= end) {
-                break;
-            }
-
-            frame_count++;
             if (do_read_packets) {
                 if (do_show_packets)
                     show_packet(w, fmt_ctx, &pkt, i++);
@@ -1704,7 +1582,7 @@ static int read_interval_packets(WriterContext *w, AVFormatContext *fmt_ctx,
             }
             if (do_read_frames) {
                 pkt1 = pkt;
-                while (pkt1.size && process_frame(w, fmt_ctx, frame, &pkt1) > 0);
+                while (pkt1.size && process_frame(w, fmt_ctx, &frame, &pkt1) > 0);
             }
         }
         av_free_packet(&pkt);
@@ -1716,36 +1594,11 @@ static int read_interval_packets(WriterContext *w, AVFormatContext *fmt_ctx,
     for (i = 0; i < fmt_ctx->nb_streams; i++) {
         pkt.stream_index = i;
         if (do_read_frames)
-            while (process_frame(w, fmt_ctx, frame, &pkt) > 0);
-    }
-
-end:
-    av_frame_free(&frame);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Could not read packets in interval ");
-        log_read_interval(interval, NULL, AV_LOG_ERROR);
-    }
-    return ret;
-}
-
-static void read_packets(WriterContext *w, AVFormatContext *fmt_ctx)
-{
-    int i, ret = 0;
-    int64_t cur_ts = fmt_ctx->start_time;
-
-    if (read_intervals_nb == 0) {
-        ReadInterval interval = (ReadInterval) { .has_start = 0, .has_end = 0 };
-        ret = read_interval_packets(w, fmt_ctx, &interval, &cur_ts);
-    } else {
-        for (i = 0; i < read_intervals_nb; i++) {
-            ret = read_interval_packets(w, fmt_ctx, &read_intervals[i], &cur_ts);
-            if (ret < 0)
-                break;
-        }
+            while (process_frame(w, fmt_ctx, &frame, &pkt) > 0);
     }
 }
 
-static void show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_idx, int in_program)
+static void show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_idx)
 {
     AVStream *stream = fmt_ctx->streams[stream_idx];
     AVCodecContext *dec_ctx;
@@ -1757,7 +1610,7 @@ static void show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_i
 
     av_bprint_init(&pbuf, 1, AV_BPRINT_SIZE_UNLIMITED);
 
-    writer_print_section_header(w, in_program ? SECTION_ID_PROGRAM_STREAM : SECTION_ID_STREAM);
+    writer_print_section_header(w, SECTION_ID_STREAM);
 
     print_int("index", stream->index);
 
@@ -1828,28 +1681,8 @@ static void show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_i
             else   print_str_opt("sample_fmt", "unknown");
             print_val("sample_rate",     dec_ctx->sample_rate, unit_hertz_str);
             print_int("channels",        dec_ctx->channels);
-
-            if (dec_ctx->channel_layout) {
-                av_bprint_clear(&pbuf);
-                av_bprint_channel_layout(&pbuf, dec_ctx->channels, dec_ctx->channel_layout);
-                print_str    ("channel_layout", pbuf.str);
-            } else {
-                print_str_opt("channel_layout", "unknown");
-            }
-
             print_int("bits_per_sample", av_get_bits_per_sample(dec_ctx->codec_id));
             break;
-
-        case AVMEDIA_TYPE_SUBTITLE:
-            if (dec_ctx->width)
-                print_int("width",       dec_ctx->width);
-            else
-                print_str_opt("width",   "N/A");
-            if (dec_ctx->height)
-                print_int("height",      dec_ctx->height);
-            else
-                print_str_opt("height",  "N/A");
-            break;
         }
     } else {
         print_str_opt("codec_type", "unknown");
@@ -1893,7 +1726,7 @@ static void show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_i
     } while (0)
 
     if (do_show_stream_disposition) {
-    writer_print_section_header(w, in_program ? SECTION_ID_PROGRAM_STREAM_DISPOSITION : SECTION_ID_STREAM_DISPOSITION);
+    writer_print_section_header(w, SECTION_ID_STREAM_DISPOSITION);
     PRINT_DISPOSITION(DEFAULT,          "default");
     PRINT_DISPOSITION(DUB,              "dub");
     PRINT_DISPOSITION(ORIGINAL,         "original");
@@ -1908,7 +1741,7 @@ static void show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_i
     writer_print_section_footer(w);
     }
 
-    show_tags(w, stream->metadata, in_program ? SECTION_ID_PROGRAM_STREAM_TAGS : SECTION_ID_STREAM_TAGS);
+    show_tags(w, stream->metadata, SECTION_ID_STREAM_TAGS);
 
     writer_print_section_footer(w);
     av_bprint_finalize(&pbuf, NULL);
@@ -1921,47 +1754,7 @@ static void show_streams(WriterContext *w, AVFormatContext *fmt_ctx)
     writer_print_section_header(w, SECTION_ID_STREAMS);
     for (i = 0; i < fmt_ctx->nb_streams; i++)
         if (selected_streams[i])
-            show_stream(w, fmt_ctx, i, 0);
-    writer_print_section_footer(w);
-}
-
-static void show_program(WriterContext *w, AVFormatContext *fmt_ctx, AVProgram *program)
-{
-    int i;
-
-    writer_print_section_header(w, SECTION_ID_PROGRAM);
-    print_int("program_id", program->id);
-    print_int("program_num", program->program_num);
-    print_int("nb_streams", program->nb_stream_indexes);
-    print_int("pmt_pid", program->pmt_pid);
-    print_int("pcr_pid", program->pcr_pid);
-    print_ts("start_pts", program->start_time);
-    print_time("start_time", program->start_time, &AV_TIME_BASE_Q);
-    print_ts("end_pts", program->end_time);
-    print_time("end_time", program->end_time, &AV_TIME_BASE_Q);
-    show_tags(w, program->metadata, SECTION_ID_PROGRAM_TAGS);
-
-    writer_print_section_header(w, SECTION_ID_PROGRAM_STREAMS);
-    for (i = 0; i < program->nb_stream_indexes; i++) {
-        if (selected_streams[program->stream_index[i]])
-            show_stream(w, fmt_ctx, program->stream_index[i], 1);
-    }
-    writer_print_section_footer(w);
-
-    writer_print_section_footer(w);
-}
-
-static void show_programs(WriterContext *w, AVFormatContext *fmt_ctx)
-{
-    int i;
-
-    writer_print_section_header(w, SECTION_ID_PROGRAMS);
-    for (i = 0; i < fmt_ctx->nb_programs; i++) {
-        AVProgram *program = fmt_ctx->programs[i];
-        if (!program)
-            continue;
-        show_program(w, fmt_ctx, program);
-    }
+            show_stream(w, fmt_ctx, i);
     writer_print_section_footer(w);
 }
 
@@ -1994,7 +1787,6 @@ static void show_format(WriterContext *w, AVFormatContext *fmt_ctx)
     writer_print_section_header(w, SECTION_ID_FORMAT);
     print_str("filename",         fmt_ctx->filename);
     print_int("nb_streams",       fmt_ctx->nb_streams);
-    print_int("nb_programs",      fmt_ctx->nb_programs);
     print_str("format_name",      fmt_ctx->iformat->name);
     if (!do_bitexact) {
         if (fmt_ctx->iformat->long_name) print_str    ("format_long_name", fmt_ctx->iformat->long_name);
@@ -2006,7 +1798,6 @@ static void show_format(WriterContext *w, AVFormatContext *fmt_ctx)
     else           print_str_opt("size", "N/A");
     if (fmt_ctx->bit_rate > 0) print_val    ("bit_rate", fmt_ctx->bit_rate, unit_bit_per_second_str);
     else                       print_str_opt("bit_rate", "N/A");
-    print_int("probe_score", av_format_get_probe_score(fmt_ctx));
     show_tags(w, fmt_ctx->metadata, SECTION_ID_FORMAT_TAGS);
 
     writer_print_section_footer(w);
@@ -2064,18 +1855,18 @@ static int open_input_file(AVFormatContext **fmt_ctx_ptr, const char *filename)
         AVCodec *codec;
 
         if (stream->codec->codec_id == AV_CODEC_ID_PROBE) {
-            av_log(NULL, AV_LOG_WARNING,
+            av_log(NULL, AV_LOG_ERROR,
                    "Failed to probe codec for input stream %d\n",
                     stream->index);
         } else if (!(codec = avcodec_find_decoder(stream->codec->codec_id))) {
-            av_log(NULL, AV_LOG_WARNING,
+            av_log(NULL, AV_LOG_ERROR,
                     "Unsupported codec with id %d for input stream %d\n",
                     stream->codec->codec_id, stream->index);
         } else {
             AVDictionary *opts = filter_codec_opts(codec_opts, stream->codec->codec_id,
                                                    fmt_ctx, stream, codec);
             if (avcodec_open2(stream->codec, codec, &opts) < 0) {
-                av_log(NULL, AV_LOG_WARNING, "Could not open codec for input stream %d\n",
+                av_log(NULL, AV_LOG_ERROR, "Error while opening codec for input stream %d\n",
                        stream->index);
             }
             if ((t = av_dict_get(opts, "", NULL, AV_DICT_IGNORE_SUFFIX))) {
@@ -2149,8 +1940,6 @@ static int probe_file(WriterContext *wctx, const char *filename)
         if (do_show_frames || do_show_packets)
             writer_print_section_footer(wctx);
     }
-    if (do_show_programs)
-        show_programs(wctx, fmt_ctx);
     if (do_show_streams)
         show_streams(wctx, fmt_ctx);
     if (do_show_chapters)
@@ -2357,145 +2146,6 @@ void show_help_default(const char *opt, const char *arg)
     show_help_children(avformat_get_class(), AV_OPT_FLAG_DECODING_PARAM);
 }
 
-/**
- * Parse interval specification, according to the format:
- * INTERVAL ::= [START|+START_OFFSET][%[END|+END_OFFSET]]
- * INTERVALS ::= INTERVAL[,INTERVALS]
-*/
-static int parse_read_interval(const char *interval_spec,
-                               ReadInterval *interval)
-{
-    int ret = 0;
-    char *next, *p, *spec = av_strdup(interval_spec);
-    if (!spec)
-        return AVERROR(ENOMEM);
-
-    if (!*spec) {
-        av_log(NULL, AV_LOG_ERROR, "Invalid empty interval specification\n");
-        ret = AVERROR(EINVAL);
-        goto end;
-    }
-
-    p = spec;
-    next = strchr(spec, '%');
-    if (next)
-        *next++ = 0;
-
-    /* parse first part */
-    if (*p) {
-        interval->has_start = 1;
-
-        if (*p == '+') {
-            interval->start_is_offset = 1;
-            p++;
-        } else {
-            interval->start_is_offset = 0;
-        }
-
-        ret = av_parse_time(&interval->start, p, 1);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Invalid interval start specification '%s'\n", p);
-            goto end;
-        }
-    } else {
-        interval->has_start = 0;
-    }
-
-    /* parse second part */
-    p = next;
-    if (p && *p) {
-        int64_t us;
-        interval->has_end = 1;
-
-        if (*p == '+') {
-            interval->end_is_offset = 1;
-            p++;
-        } else {
-            interval->end_is_offset = 0;
-        }
-
-        if (interval->end_is_offset && *p == '#') {
-            long long int lli;
-            char *tail;
-            interval->duration_frames = 1;
-            p++;
-            lli = strtoll(p, &tail, 10);
-            if (*tail || lli < 0) {
-                av_log(NULL, AV_LOG_ERROR,
-                       "Invalid or negative value '%s' for duration number of frames\n", p);
-                goto end;
-            }
-            interval->end = lli;
-        } else {
-            ret = av_parse_time(&us, p, 1);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Invalid interval end/duration specification '%s'\n", p);
-                goto end;
-            }
-            interval->end = us;
-        }
-    } else {
-        interval->has_end = 0;
-    }
-
-end:
-    av_free(spec);
-    return ret;
-}
-
-static int parse_read_intervals(const char *intervals_spec)
-{
-    int ret, n, i;
-    char *p, *spec = av_strdup(intervals_spec);
-    if (!spec)
-        return AVERROR(ENOMEM);
-
-    /* preparse specification, get number of intervals */
-    for (n = 0, p = spec; *p; p++)
-        if (*p == ',')
-            n++;
-    n++;
-
-    read_intervals = av_malloc(n * sizeof(*read_intervals));
-    if (!read_intervals) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-    read_intervals_nb = n;
-
-    /* parse intervals */
-    p = spec;
-    for (i = 0; p; i++) {
-        char *next;
-
-        av_assert0(i < read_intervals_nb);
-        next = strchr(p, ',');
-        if (next)
-            *next++ = 0;
-
-        read_intervals[i].id = i;
-        ret = parse_read_interval(p, &read_intervals[i]);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Error parsing read interval #%d '%s'\n",
-                   i, p);
-            goto end;
-        }
-        av_log(NULL, AV_LOG_VERBOSE, "Parsed log interval ");
-        log_read_interval(&read_intervals[i], NULL, AV_LOG_VERBOSE);
-        p = next;
-    }
-    av_assert0(i == read_intervals_nb);
-
-end:
-    av_free(spec);
-    return ret;
-}
-
-static int opt_read_intervals(void *optctx, const char *opt, const char *arg)
-{
-    return parse_read_intervals(arg);
-}
-
 static int opt_pretty(void *optctx, const char *opt, const char *arg)
 {
     show_value_unit              = 1;
@@ -2556,7 +2206,6 @@ DEFINE_OPT_SHOW_SECTION(library_versions, LIBRARY_VERSIONS);
 DEFINE_OPT_SHOW_SECTION(packets,          PACKETS);
 DEFINE_OPT_SHOW_SECTION(program_version,  PROGRAM_VERSION);
 DEFINE_OPT_SHOW_SECTION(streams,          STREAMS);
-DEFINE_OPT_SHOW_SECTION(programs,         PROGRAMS);
 
 static const OptionDef real_options[] = {
 #include "cmdutils_common_opts.h"
@@ -2583,7 +2232,6 @@ static const OptionDef real_options[] = {
     { "show_entries", HAS_ARG, {.func_arg = opt_show_entries},
       "show a set of specified entries", "entry_list" },
     { "show_packets", 0, {(void*)&opt_show_packets}, "show packets info" },
-    { "show_programs", 0, {(void*)&opt_show_programs}, "show programs info" },
     { "show_streams", 0, {(void*)&opt_show_streams}, "show streams info" },
     { "show_chapters", 0, {(void*)&opt_show_chapters}, "show chapters info" },
     { "count_frames", OPT_BOOL, {(void*)&do_count_frames}, "count the number of frames per stream" },
@@ -2594,7 +2242,6 @@ static const OptionDef real_options[] = {
     { "show_private_data", OPT_BOOL, {(void*)&show_private_data}, "show private data" },
     { "private",           OPT_BOOL, {(void*)&show_private_data}, "same as show_private_data" },
     { "bitexact", OPT_BOOL, {&do_bitexact}, "force bitexact output" },
-    { "read_intervals", HAS_ARG, {.func_arg = opt_read_intervals}, "set read intervals", "read_intervals" },
     { "default", HAS_ARG | OPT_AUDIO | OPT_VIDEO | OPT_EXPERT, {.func_arg = opt_default}, "generic catch all option", "" },
     { "i", HAS_ARG, {.func_arg = opt_input_file_i}, "read specified file", "input_file"},
     { NULL, },
@@ -2648,10 +2295,8 @@ int main(int argc, char **argv)
     SET_DO_SHOW(LIBRARY_VERSIONS, library_versions);
     SET_DO_SHOW(PACKETS, packets);
     SET_DO_SHOW(PROGRAM_VERSION, program_version);
-    SET_DO_SHOW(PROGRAMS, programs);
     SET_DO_SHOW(STREAMS, streams);
     SET_DO_SHOW(STREAM_DISPOSITION, stream_disposition);
-    SET_DO_SHOW(PROGRAM_STREAM_DISPOSITION, stream_disposition);
 
     if (do_bitexact && (do_show_program_version || do_show_library_versions)) {
         av_log(NULL, AV_LOG_ERROR,
@@ -2689,7 +2334,7 @@ int main(int argc, char **argv)
             ffprobe_show_library_versions(wctx);
 
         if (!input_filename &&
-            ((do_show_format || do_show_programs || do_show_streams || do_show_chapters || do_show_packets || do_show_error) ||
+            ((do_show_format || do_show_streams || do_show_chapters || do_show_packets || do_show_error) ||
              (!do_show_program_version && !do_show_library_versions))) {
             show_usage();
             av_log(NULL, AV_LOG_ERROR, "You have to specify one input file.\n");
@@ -2707,7 +2352,6 @@ int main(int argc, char **argv)
 
 end:
     av_freep(&print_format);
-    av_freep(&read_intervals);
 
     uninit_opts();
     for (i = 0; i < FF_ARRAY_ELEMS(sections); i++)

ffserver.c

@@ -36,7 +36,6 @@
 #include "libavformat/network.h"
 #include "libavformat/os_support.h"
 #include "libavformat/rtpdec.h"
-#include "libavformat/rtpproto.h"
 #include "libavformat/rtsp.h"
 #include "libavformat/avio_internal.h"
 #include "libavformat/internal.h"
@@ -48,7 +47,6 @@
 #include "libavutil/dict.h"
 #include "libavutil/intreadwrite.h"
 #include "libavutil/mathematics.h"
-#include "libavutil/pixdesc.h"
 #include "libavutil/random_seed.h"
 #include "libavutil/parseutils.h"
 #include "libavutil/opt.h"
@@ -65,6 +63,9 @@
 #include <time.h>
 #include <sys/wait.h>
 #include <signal.h>
+#if HAVE_DLFCN_H
+#include <dlfcn.h>
+#endif
 
 #include "cmdutils.h"
 
@@ -327,14 +328,6 @@ static AVLFG random_state;
 
 static FILE *logfile = NULL;
 
-static void htmlstrip(char *s) {
-    while (s && *s) {
-        s += strspn(s, "0123456789abcdefghijklmnopqrstuvwxyzABCDEFGHIJKLMNOPQRSTUVWXYZ,. ");
-        if (*s)
-            *s++ = '?';
-    }
-}
-
 static int64_t ffm_read_write_index(int fd)
 {
     uint8_t buf[8];
@@ -1894,7 +1887,6 @@ static int http_parse_request(HTTPContext *c)
  send_error:
     c->http_error = 404;
     q = c->buffer;
-    htmlstrip(msg);
     snprintf(q, c->buffer_size,
                   "HTTP/1.0 404 Not Found\r\n"
                   "Content-type: text/html\r\n"
@@ -3914,7 +3906,7 @@ static void add_codec(FFStream *stream, AVCodecContext *av)
         av->rc_buffer_aggressivity = 1.0;
 
         if (!av->rc_eq)
-            av->rc_eq = av_strdup("tex^qComp");
+            av->rc_eq = "tex^qComp";
         if (!av->i_quant_factor)
             av->i_quant_factor = -0.8;
         if (!av->b_quant_factor)
@@ -3962,6 +3954,33 @@ static enum AVCodecID opt_video_codec(const char *arg)
     return p->id;
 }
 
+/* simplistic plugin support */
+
+#if HAVE_DLOPEN
+static void load_module(const char *filename)
+{
+    void *dll;
+    void (*init_func)(void);
+    dll = dlopen(filename, RTLD_NOW);
+    if (!dll) {
+        fprintf(stderr, "Could not load module '%s' - %s\n",
+                filename, dlerror());
+        return;
+    }
+
+    init_func = dlsym(dll, "ffserver_module_init");
+    if (!init_func) {
+        fprintf(stderr,
+                "%s: init function 'ffserver_module_init()' not found\n",
+                filename);
+        dlclose(dll);
+        return;
+    }
+
+    init_func();
+}
+#endif
+
 static int ffserver_opt_default(const char *opt, const char *arg,
                        AVCodecContext *avctx, int type)
 {
@@ -4435,14 +4454,6 @@ static int parse_ffconfig(const char *filename)
                     video_enc.time_base.den = frame_rate.num;
                 }
             }
-        } else if (!av_strcasecmp(cmd, "PixelFormat")) {
-            get_arg(arg, sizeof(arg), &p);
-            if (stream) {
-                video_enc.pix_fmt = av_get_pix_fmt(arg);
-                if (video_enc.pix_fmt == AV_PIX_FMT_NONE) {
-                    ERROR("Unknown pixel format: %s\n", arg);
-                }
-            }
         } else if (!av_strcasecmp(cmd, "VideoGopSize")) {
             get_arg(arg, sizeof(arg), &p);
             if (stream)
@@ -4626,7 +4637,12 @@ static int parse_ffconfig(const char *filename)
                 redirect = NULL;
             }
         } else if (!av_strcasecmp(cmd, "LoadModule")) {
-            ERROR("Loadable modules no longer supported\n");
+            get_arg(arg, sizeof(arg), &p);
+#if HAVE_DLOPEN
+            load_module(arg);
+#else
+            ERROR("Module support not compiled into this version: '%s'\n", arg);
+#endif
         } else {
             ERROR("Incorrect keyword: '%s'\n", cmd);
         }
@@ -4691,8 +4707,6 @@ int main(int argc, char **argv)
 {
     struct sigaction sigact = { { 0 } };
 
-    config_filename = av_strdup("/etc/ffserver.conf");
-
     parse_loglevel(argc, argv, options);
     av_register_all();
     avformat_network_init();
@@ -4703,6 +4717,9 @@ int main(int argc, char **argv)
 
     parse_options(NULL, argc, argv, options, NULL);
 
+    if (!config_filename)
+        config_filename = av_strdup("/etc/ffserver.conf");
+
     unsetenv("http_proxy");             /* Kill the http_proxy */
 
     av_lfg_init(&random_state, av_get_random_seed());

libavcodec/012v.c

@@ -49,12 +49,6 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
         av_log(avctx, AV_LOG_ERROR, "Width 1 not supported.\n");
         return AVERROR_INVALIDDATA;
     }
-
-    if (   avctx->codec_tag == MKTAG('0', '1', '2', 'v')
-        && avpkt->size % avctx->height == 0
-        && avpkt->size / avctx->height * 3 >= width * 8)
-        stride = avpkt->size / avctx->height;
-
     if (avpkt->size < avctx->height * stride) {
         av_log(avctx, AV_LOG_ERROR, "Packet too small: %d instead of %d\n",
                avpkt->size, avctx->height * stride);
@@ -150,10 +144,10 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
 
 AVCodec ff_zero12v_decoder = {
     .name           = "012v",
-    .long_name      = NULL_IF_CONFIG_SMALL("Uncompressed 4:2:2 10-bit"),
     .type           = AVMEDIA_TYPE_VIDEO,
     .id             = AV_CODEC_ID_012V,
     .init           = zero12v_decode_init,
     .decode         = zero12v_decode_frame,
     .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("Uncompressed 4:2:2 10-bit"),
 };

libavcodec/4xm.c

@@ -842,8 +842,6 @@ static int decode_frame(AVCodecContext *avctx, void *data,
     if (buf_size < 20)
         return AVERROR_INVALIDDATA;
 
-    av_assert0(avctx->width % 16 == 0 && avctx->height % 16 == 0);
-
     if (buf_size < AV_RL32(buf + 4) + 8) {
         av_log(f->avctx, AV_LOG_ERROR, "size mismatch %d %d\n",
                buf_size, AV_RL32(buf + 4));
@@ -1020,7 +1018,6 @@ static av_cold int decode_end(AVCodecContext *avctx)
 
 AVCodec ff_fourxm_decoder = {
     .name           = "4xm",
-    .long_name      = NULL_IF_CONFIG_SMALL("4X Movie"),
     .type           = AVMEDIA_TYPE_VIDEO,
     .id             = AV_CODEC_ID_4XM,
     .priv_data_size = sizeof(FourXContext),
@@ -1028,4 +1025,5 @@ AVCodec ff_fourxm_decoder = {
     .close          = decode_end,
     .decode         = decode_frame,
     .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("4X Movie"),
 };

libavcodec/8bps.c

@@ -64,7 +64,7 @@ static int decode_frame(AVCodecContext *avctx, void *data,
     unsigned char *pixptr, *pixptr_end;
     unsigned int height = avctx->height; // Real image height
     unsigned int dlen, p, row;
-    const unsigned char *lp, *dp, *ep;
+    const unsigned char *lp, *dp;
     unsigned char count;
     unsigned int planes     = c->planes;
     unsigned char *planemap = c->planemap;
@@ -73,8 +73,6 @@ static int decode_frame(AVCodecContext *avctx, void *data,
     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
         return ret;
 
-    ep = encoded + buf_size;
-
     /* Set data pointer after line lengths */
     dp = encoded + planes * (height << 1);
 
@@ -86,19 +84,19 @@ static int decode_frame(AVCodecContext *avctx, void *data,
         for (row = 0; row < height; row++) {
             pixptr = frame->data[0] + row * frame->linesize[0] + planemap[p];
             pixptr_end = pixptr + frame->linesize[0];
-            if (ep - lp < row * 2 + 2)
-                return AVERROR_INVALIDDATA;
+            if(lp - encoded + row*2 + 1 >= buf_size)
+                return -1;
             dlen = av_be2ne16(*(const unsigned short *)(lp + row * 2));
             /* Decode a row of this plane */
             while (dlen > 0) {
-                if (ep - dp <= 1)
+                if (dp + 1 >= buf + buf_size)
                     return AVERROR_INVALIDDATA;
                 if ((count = *dp++) <= 127) {
                     count++;
                     dlen -= count + 1;
-                    if (pixptr_end - pixptr < count * planes)
+                    if (pixptr + count * planes > pixptr_end)
                         break;
-                    if (ep - dp < count)
+                    if (dp + count > buf + buf_size)
                         return AVERROR_INVALIDDATA;
                     while (count--) {
                         *pixptr = *dp++;
@@ -106,7 +104,7 @@ static int decode_frame(AVCodecContext *avctx, void *data,
                     }
                 } else {
                     count = 257 - count;
-                    if (pixptr_end - pixptr < count * planes)
+                    if (pixptr + count * planes > pixptr_end)
                         break;
                     while (count--) {
                         *pixptr = *dp;
@@ -159,7 +157,17 @@ static av_cold int decode_init(AVCodecContext *avctx)
     case 32:
         avctx->pix_fmt = AV_PIX_FMT_RGB32;
         c->planes      = 4;
-        /* handle planemap setup later for decoding rgb24 data as rbg32 */
+#if HAVE_BIGENDIAN
+        c->planemap[0] = 1; // 1st plane is red
+        c->planemap[1] = 2; // 2nd plane is green
+        c->planemap[2] = 3; // 3rd plane is blue
+        c->planemap[3] = 0; // 4th plane is alpha
+#else
+        c->planemap[0] = 2; // 1st plane is red
+        c->planemap[1] = 1; // 2nd plane is green
+        c->planemap[2] = 0; // 3rd plane is blue
+        c->planemap[3] = 3; // 4th plane is alpha
+#endif
         break;
     default:
         av_log(avctx, AV_LOG_ERROR, "Error: Unsupported color depth: %u.\n",
@@ -167,22 +175,16 @@ static av_cold int decode_init(AVCodecContext *avctx)
         return AVERROR_INVALIDDATA;
     }
 
-    if (avctx->pix_fmt == AV_PIX_FMT_RGB32) {
-        c->planemap[0] = HAVE_BIGENDIAN ? 1 : 2; // 1st plane is red
-        c->planemap[1] = HAVE_BIGENDIAN ? 2 : 1; // 2nd plane is green
-        c->planemap[2] = HAVE_BIGENDIAN ? 3 : 0; // 3rd plane is blue
-        c->planemap[3] = HAVE_BIGENDIAN ? 0 : 3; // 4th plane is alpha
-    }
     return 0;
 }
 
 AVCodec ff_eightbps_decoder = {
     .name           = "8bps",
-    .long_name      = NULL_IF_CONFIG_SMALL("QuickTime 8BPS video"),
     .type           = AVMEDIA_TYPE_VIDEO,
     .id             = AV_CODEC_ID_8BPS,
     .priv_data_size = sizeof(EightBpsContext),
     .init           = decode_init,
     .decode         = decode_frame,
     .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("QuickTime 8BPS video"),
 };

libavcodec/8svx.c

@@ -187,7 +187,6 @@ static av_cold int eightsvx_decode_close(AVCodecContext *avctx)
 #if CONFIG_EIGHTSVX_FIB_DECODER
 AVCodec ff_eightsvx_fib_decoder = {
   .name           = "8svx_fib",
-  .long_name      = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
   .type           = AVMEDIA_TYPE_AUDIO,
   .id             = AV_CODEC_ID_8SVX_FIB,
   .priv_data_size = sizeof (EightSvxContext),
@@ -195,6 +194,7 @@ AVCodec ff_eightsvx_fib_decoder = {
   .decode         = eightsvx_decode_frame,
   .close          = eightsvx_decode_close,
   .capabilities   = CODEC_CAP_DR1,
+  .long_name      = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
   .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
                                                     AV_SAMPLE_FMT_NONE },
 };
@@ -202,7 +202,6 @@ AVCodec ff_eightsvx_fib_decoder = {
 #if CONFIG_EIGHTSVX_EXP_DECODER
 AVCodec ff_eightsvx_exp_decoder = {
   .name           = "8svx_exp",
-  .long_name      = NULL_IF_CONFIG_SMALL("8SVX exponential"),
   .type           = AVMEDIA_TYPE_AUDIO,
   .id             = AV_CODEC_ID_8SVX_EXP,
   .priv_data_size = sizeof (EightSvxContext),
@@ -210,6 +209,7 @@ AVCodec ff_eightsvx_exp_decoder = {
   .decode         = eightsvx_decode_frame,
   .close          = eightsvx_decode_close,
   .capabilities   = CODEC_CAP_DR1,
+  .long_name      = NULL_IF_CONFIG_SMALL("8SVX exponential"),
   .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
                                                     AV_SAMPLE_FMT_NONE },
 };

libavcodec/Makefile

@@ -31,8 +31,6 @@ OBJS = allcodecs.o                                                      \
        resample2.o                                                      \
        utils.o                                                          \
 
-OBJS-$(HAVE_MSVCRT) += file_open.o
-
 # parts needed for many different codecs
 OBJS-$(CONFIG_AANDCTTABLES)            += aandcttab.o
 OBJS-$(CONFIG_AC3DSP)                  += ac3dsp.o
@@ -44,10 +42,8 @@ OBJS-$(CONFIG_DSPUTIL)                 += dsputil.o faanidct.o          \
                                           simple_idct.o jrevdct.o
 OBJS-$(CONFIG_ENCODERS)                += faandct.o jfdctfst.o jfdctint.o
 OBJS-$(CONFIG_ERROR_RESILIENCE)        += error_resilience.o
-OBJS-$(CONFIG_EXIF)                    += exif.o tiff_common.o
 FFT-OBJS-$(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_GOLOMB)                  += golomb.o
 OBJS-$(CONFIG_H264CHROMA)              += h264chroma.o
@@ -59,7 +55,7 @@ OBJS-$(CONFIG_HUFFMAN)                 += huffman.o
 OBJS-$(CONFIG_LIBXVID)                 += libxvid_rc.o
 OBJS-$(CONFIG_LPC)                     += lpc.o
 OBJS-$(CONFIG_LSP)                     += lsp.o
-OBJS-$(CONFIG_MDCT)                    += mdct_fixed.o mdct_float.o mdct_fixed_32.o
+OBJS-$(CONFIG_MDCT)                    += mdct_fixed.o mdct_float.o
 OBJS-$(CONFIG_MPEGAUDIO)               += mpegaudio.o mpegaudiodata.o   \
                                           mpegaudiodecheader.o
 OBJS-$(CONFIG_MPEGAUDIODSP)            += mpegaudiodsp.o                \
@@ -233,10 +229,6 @@ OBJS-$(CONFIG_H264_DECODER)            += h264.o                               \
                                           cabac.o h264_sei.o h264_ps.o         \
                                           h264_refs.o h264_cavlc.o h264_cabac.o
 OBJS-$(CONFIG_H264_VDA_DECODER)        += vda_h264_dec.o
-OBJS-$(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 cabac.o
-
 OBJS-$(CONFIG_HUFFYUV_DECODER)         += huffyuv.o huffyuvdec.o
 OBJS-$(CONFIG_HUFFYUV_ENCODER)         += huffyuv.o huffyuvenc.o
 OBJS-$(CONFIG_IAC_DECODER)             += imc.o
@@ -268,8 +260,6 @@ OBJS-$(CONFIG_LOCO_DECODER)            += loco.o
 OBJS-$(CONFIG_MACE3_DECODER)           += mace.o
 OBJS-$(CONFIG_MACE6_DECODER)           += mace.o
 OBJS-$(CONFIG_MDEC_DECODER)            += mdec.o mpeg12.o mpeg12data.o
-OBJS-$(CONFIG_METASOUND_DECODER)       += metasound.o metasound_data.o \
-                                          twinvq.o
 OBJS-$(CONFIG_MICRODVD_DECODER)        += microdvddec.o ass.o
 OBJS-$(CONFIG_MIMIC_DECODER)           += mimic.o
 OBJS-$(CONFIG_MJPEG_DECODER)           += mjpegdec.o mjpeg.o
@@ -331,23 +321,23 @@ OBJS-$(CONFIG_NUV_DECODER)             += nuv.o rtjpeg.o
 OBJS-$(CONFIG_PAF_VIDEO_DECODER)       += paf.o
 OBJS-$(CONFIG_PAF_AUDIO_DECODER)       += paf.o
 OBJS-$(CONFIG_PAM_DECODER)             += pnmdec.o pnm.o
-OBJS-$(CONFIG_PAM_ENCODER)             += pamenc.o
+OBJS-$(CONFIG_PAM_ENCODER)             += pamenc.o pnm.o
 OBJS-$(CONFIG_PBM_DECODER)             += pnmdec.o pnm.o
-OBJS-$(CONFIG_PBM_ENCODER)             += pnmenc.o
+OBJS-$(CONFIG_PBM_ENCODER)             += pnmenc.o pnm.o
 OBJS-$(CONFIG_PCX_DECODER)             += pcx.o
 OBJS-$(CONFIG_PCX_ENCODER)             += pcxenc.o
 OBJS-$(CONFIG_PGM_DECODER)             += pnmdec.o pnm.o
-OBJS-$(CONFIG_PGM_ENCODER)             += pnmenc.o
+OBJS-$(CONFIG_PGM_ENCODER)             += pnmenc.o pnm.o
 OBJS-$(CONFIG_PGMYUV_DECODER)          += pnmdec.o pnm.o
-OBJS-$(CONFIG_PGMYUV_ENCODER)          += pnmenc.o
+OBJS-$(CONFIG_PGMYUV_ENCODER)          += pnmenc.o pnm.o
 OBJS-$(CONFIG_PGSSUB_DECODER)          += pgssubdec.o
 OBJS-$(CONFIG_PICTOR_DECODER)          += pictordec.o cga_data.o
 OBJS-$(CONFIG_PJS_DECODER)             += textdec.o ass.o
 OBJS-$(CONFIG_PNG_DECODER)             += png.o pngdec.o pngdsp.o
 OBJS-$(CONFIG_PNG_ENCODER)             += png.o pngenc.o
 OBJS-$(CONFIG_PPM_DECODER)             += pnmdec.o pnm.o
-OBJS-$(CONFIG_PPM_ENCODER)             += pnmenc.o
-OBJS-$(CONFIG_PRORES_DECODER)          += proresdec2.o proresdsp.o proresdata.o
+OBJS-$(CONFIG_PPM_ENCODER)             += pnmenc.o pnm.o
+OBJS-$(CONFIG_PRORES_DECODER)          += proresdec2.o proresdsp.o
 OBJS-$(CONFIG_PRORES_LGPL_DECODER)     += proresdec_lgpl.o proresdsp.o proresdata.o
 OBJS-$(CONFIG_PRORES_ENCODER)          += proresenc_anatoliy.o
 OBJS-$(CONFIG_PRORES_AW_ENCODER)       += proresenc_anatoliy.o
@@ -431,7 +421,7 @@ OBJS-$(CONFIG_TARGA_Y216_DECODER)      += targa_y216dec.o
 OBJS-$(CONFIG_THEORA_DECODER)          += xiph.o
 OBJS-$(CONFIG_THP_DECODER)             += mjpegdec.o mjpeg.o
 OBJS-$(CONFIG_TIERTEXSEQVIDEO_DECODER) += tiertexseqv.o
-OBJS-$(CONFIG_TIFF_DECODER)            += tiff.o lzw.o faxcompr.o tiff_data.o tiff_common.o
+OBJS-$(CONFIG_TIFF_DECODER)            += tiff.o lzw.o faxcompr.o tiff_data.o
 OBJS-$(CONFIG_TIFF_ENCODER)            += tiffenc.o rle.o lzwenc.o tiff_data.o
 OBJS-$(CONFIG_TMV_DECODER)             += tmv.o cga_data.o
 OBJS-$(CONFIG_TRUEHD_DECODER)          += mlpdec.o mlpdsp.o
@@ -442,7 +432,7 @@ OBJS-$(CONFIG_TSCC_DECODER)            += tscc.o msrledec.o
 OBJS-$(CONFIG_TSCC2_DECODER)           += tscc2.o
 OBJS-$(CONFIG_TTA_DECODER)             += tta.o ttadata.o
 OBJS-$(CONFIG_TTA_ENCODER)             += ttaenc.o ttadata.o
-OBJS-$(CONFIG_TWINVQ_DECODER)          += twinvqdec.o twinvq.o
+OBJS-$(CONFIG_TWINVQ_DECODER)          += twinvq.o
 OBJS-$(CONFIG_TXD_DECODER)             += txd.o s3tc.o
 OBJS-$(CONFIG_ULTI_DECODER)            += ulti.o
 OBJS-$(CONFIG_UTVIDEO_DECODER)         += utvideodec.o utvideo.o
@@ -475,13 +465,10 @@ OBJS-$(CONFIG_VP5_DECODER)             += vp5.o vp56.o vp56data.o vp56dsp.o \
 OBJS-$(CONFIG_VP6_DECODER)             += vp6.o vp56.o vp56data.o vp56dsp.o \
                                           vp6dsp.o vp56rac.o
 OBJS-$(CONFIG_VP8_DECODER)             += vp8.o vp8dsp.o vp56rac.o
-OBJS-$(CONFIG_VP9_DECODER)             += vp9.o vp9dsp.o vp56rac.o
 OBJS-$(CONFIG_VPLAYER_DECODER)         += textdec.o ass.o
 OBJS-$(CONFIG_VQA_DECODER)             += vqavideo.o
 OBJS-$(CONFIG_WAVPACK_DECODER)         += wavpack.o
-OBJS-$(CONFIG_WAVPACK_ENCODER)         += wavpackenc.o
 OBJS-$(CONFIG_WEBP_DECODER)            += vp8.o vp8dsp.o vp56rac.o
-OBJS-$(CONFIG_WEBP_DECODER)            += webp.o
 OBJS-$(CONFIG_WEBVTT_DECODER)          += webvttdec.o
 OBJS-$(CONFIG_WMALOSSLESS_DECODER)     += wmalosslessdec.o wma_common.o
 OBJS-$(CONFIG_WMAPRO_DECODER)          += wmaprodec.o wma.o wma_common.o
@@ -527,8 +514,8 @@ OBJS-$(CONFIG_ZMBV_ENCODER)            += zmbvenc.o
 # (AD)PCM decoders/encoders
 OBJS-$(CONFIG_PCM_ALAW_DECODER)           += pcm.o
 OBJS-$(CONFIG_PCM_ALAW_ENCODER)           += pcm.o
-OBJS-$(CONFIG_PCM_BLURAY_DECODER)         += pcm-bluray.o
-OBJS-$(CONFIG_PCM_DVD_DECODER)            += pcm-dvd.o
+OBJS-$(CONFIG_PCM_BLURAY_DECODER)         += pcm-mpeg.o
+OBJS-$(CONFIG_PCM_DVD_DECODER)            += pcm.o
 OBJS-$(CONFIG_PCM_F32BE_DECODER)          += pcm.o
 OBJS-$(CONFIG_PCM_F32BE_ENCODER)          += pcm.o
 OBJS-$(CONFIG_PCM_F32LE_DECODER)          += pcm.o
@@ -627,7 +614,7 @@ OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER)       += adpcmenc.o adpcm_data.o
 OBJS-$(CONFIG_VIMA_DECODER)               += vima.o adpcm_data.o
 
 # hardware accelerators
-OBJS-$(CONFIG_H263_VAAPI_HWACCEL)         += vaapi_mpeg4.o vaapi_mpeg.o
+OBJS-$(CONFIG_H263_VAAPI_HWACCEL)         += vaapi_mpeg4.o
 OBJS-$(CONFIG_H263_VDPAU_HWACCEL)         += vdpau_mpeg4.o
 OBJS-$(CONFIG_H264_DXVA2_HWACCEL)         += dxva2_h264.o
 OBJS-$(CONFIG_H264_VAAPI_HWACCEL)         += vaapi_h264.o
@@ -635,12 +622,12 @@ OBJS-$(CONFIG_H264_VDA_HWACCEL)           += vda_h264.o
 OBJS-$(CONFIG_H264_VDPAU_HWACCEL)         += vdpau_h264.o
 OBJS-$(CONFIG_MPEG1_VDPAU_HWACCEL)        += vdpau_mpeg12.o
 OBJS-$(CONFIG_MPEG2_DXVA2_HWACCEL)        += dxva2_mpeg2.o
-OBJS-$(CONFIG_MPEG2_VAAPI_HWACCEL)        += vaapi_mpeg2.o vaapi_mpeg.o
+OBJS-$(CONFIG_MPEG2_VAAPI_HWACCEL)        += vaapi_mpeg2.o
 OBJS-$(CONFIG_MPEG2_VDPAU_HWACCEL)        += vdpau_mpeg12.o
-OBJS-$(CONFIG_MPEG4_VAAPI_HWACCEL)        += vaapi_mpeg4.o vaapi_mpeg.o
+OBJS-$(CONFIG_MPEG4_VAAPI_HWACCEL)        += vaapi_mpeg4.o
 OBJS-$(CONFIG_MPEG4_VDPAU_HWACCEL)        += vdpau_mpeg4.o
 OBJS-$(CONFIG_VC1_DXVA2_HWACCEL)          += dxva2_vc1.o
-OBJS-$(CONFIG_VC1_VAAPI_HWACCEL)          += vaapi_vc1.o vaapi_mpeg.o
+OBJS-$(CONFIG_VC1_VAAPI_HWACCEL)          += vaapi_vc1.o
 OBJS-$(CONFIG_VC1_VDPAU_HWACCEL)          += vdpau_vc1.o
 
 # libavformat dependencies
@@ -690,7 +677,6 @@ OBJS-$(CONFIG_WTV_DEMUXER)             += mpeg4audio.o mpegaudiodata.o
 OBJS-$(CONFIG_LIBAACPLUS_ENCODER)         += libaacplus.o
 OBJS-$(CONFIG_LIBCELT_DECODER)            += libcelt_dec.o
 OBJS-$(CONFIG_LIBFAAC_ENCODER)            += libfaac.o
-OBJS-$(CONFIG_LIBFDK_AAC_DECODER)         += libfdk-aacdec.o
 OBJS-$(CONFIG_LIBFDK_AAC_ENCODER)         += libfdk-aacenc.o
 OBJS-$(CONFIG_LIBGSM_DECODER)             += libgsm.o
 OBJS-$(CONFIG_LIBGSM_ENCODER)             += libgsm.o
@@ -733,7 +719,6 @@ OBJS-$(CONFIG_LIBWAVPACK_ENCODER)         += libwavpackenc.o
 OBJS-$(CONFIG_LIBX264_ENCODER)            += libx264.o
 OBJS-$(CONFIG_LIBXAVS_ENCODER)            += libxavs.o
 OBJS-$(CONFIG_LIBXVID_ENCODER)            += libxvid.o
-OBJS-$(CONFIG_LIBZVBI_TELETEXT_DECODER)   += libzvbi-teletextdec.o
 
 # parsers
 OBJS-$(CONFIG_AAC_PARSER)              += aac_parser.o aac_ac3_parser.o \
@@ -748,7 +733,6 @@ OBJS-$(CONFIG_COOK_PARSER)             += cook_parser.o
 OBJS-$(CONFIG_DCA_PARSER)              += dca_parser.o dca.o
 OBJS-$(CONFIG_DIRAC_PARSER)            += dirac_parser.o
 OBJS-$(CONFIG_DNXHD_PARSER)            += dnxhd_parser.o
-OBJS-$(CONFIG_DPX_PARSER)              += dpx_parser.o
 OBJS-$(CONFIG_DVBSUB_PARSER)           += dvbsub_parser.o
 OBJS-$(CONFIG_DVD_NAV_PARSER)          += dvd_nav_parser.o
 OBJS-$(CONFIG_DVDSUB_PARSER)           += dvdsub_parser.o
@@ -762,7 +746,6 @@ OBJS-$(CONFIG_H264_PARSER)             += h264_parser.o h264.o            \
                                           h264_refs.o h264_sei.o h264_direct.o \
                                           h264_loopfilter.o h264_cabac.o \
                                           h264_cavlc.o h264_ps.o
-OBJS-$(CONFIG_HEVC_PARSER)             += hevc_parser.o
 OBJS-$(CONFIG_MJPEG_PARSER)            += mjpeg_parser.o
 OBJS-$(CONFIG_MLP_PARSER)              += mlp_parser.o mlp.o
 OBJS-$(CONFIG_MPEG4VIDEO_PARSER)       += mpeg4video_parser.o h263.o \
@@ -822,19 +805,18 @@ SKIPHEADERS-$(CONFIG_LIBUTVIDEO)       += libutvideo.h
 SKIPHEADERS-$(CONFIG_MPEG_XVMC_DECODER) += xvmc.h
 SKIPHEADERS-$(CONFIG_VAAPI)            += vaapi_internal.h
 SKIPHEADERS-$(CONFIG_VDA)              += vda.h
-SKIPHEADERS-$(CONFIG_VDPAU)            += vdpau.h vdpau_internal.h
+SKIPHEADERS-$(CONFIG_VDPAU)            += vdpau.h
 
 TESTPROGS = cabac                                                       \
+            dct                                                         \
             fft                                                         \
             fft-fixed                                                   \
-            fft-fixed32                                                 \
             golomb                                                      \
             iirfilter                                                   \
             imgconvert                                                  \
             rangecoder                                                  \
             snowenc                                                     \
 
-TESTPROGS-$(CONFIG_DCT) += dct
 TESTPROGS-$(HAVE_MMX) += motion
 TESTOBJS = dctref.o
 

libavcodec/a64multienc.c

@@ -40,6 +40,9 @@
 #define C64YRES 200
 
 typedef struct A64Context {
+    /* general variables */
+    AVFrame picture;
+
     /* variables for multicolor modes */
     AVLFG randctx;
     int mc_lifetime;
@@ -186,7 +189,6 @@ static void render_charset(AVCodecContext *avctx, uint8_t *charset,
 static av_cold int a64multi_close_encoder(AVCodecContext *avctx)
 {
     A64Context *c = avctx->priv_data;
-    av_frame_free(&avctx->coded_frame);
     av_free(c->mc_meta_charset);
     av_free(c->mc_best_cb);
     av_free(c->mc_charset);
@@ -238,12 +240,8 @@ static av_cold int a64multi_init_encoder(AVCodecContext *avctx)
     AV_WB32(avctx->extradata, c->mc_lifetime);
     AV_WB32(avctx->extradata + 16, INTERLACED);
 
-    avctx->coded_frame = av_frame_alloc();
-    if (!avctx->coded_frame) {
-        a64multi_close_encoder(avctx);
-        return AVERROR(ENOMEM);
-    }
-
+    avcodec_get_frame_defaults(&c->picture);
+    avctx->coded_frame            = &c->picture;
     avctx->coded_frame->pict_type = AV_PICTURE_TYPE_I;
     avctx->coded_frame->key_frame = 1;
     if (!avctx->codec_tag)
@@ -273,7 +271,7 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
                                  const AVFrame *pict, int *got_packet)
 {
     A64Context *c = avctx->priv_data;
-    AVFrame *const p = avctx->coded_frame;
+    AVFrame *const p = &c->picture;
 
     int frame;
     int x, y;
@@ -398,7 +396,6 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
 #if CONFIG_A64MULTI_ENCODER
 AVCodec ff_a64multi_encoder = {
     .name           = "a64multi",
-    .long_name      = NULL_IF_CONFIG_SMALL("Multicolor charset for Commodore 64"),
     .type           = AVMEDIA_TYPE_VIDEO,
     .id             = AV_CODEC_ID_A64_MULTI,
     .priv_data_size = sizeof(A64Context),
@@ -406,13 +403,13 @@ 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},
+    .long_name      = NULL_IF_CONFIG_SMALL("Multicolor charset for Commodore 64"),
     .capabilities   = CODEC_CAP_DELAY,
 };
 #endif
 #if CONFIG_A64MULTI5_ENCODER
 AVCodec ff_a64multi5_encoder = {
     .name           = "a64multi5",
-    .long_name      = NULL_IF_CONFIG_SMALL("Multicolor charset for Commodore 64, extended with 5th color (colram)"),
     .type           = AVMEDIA_TYPE_VIDEO,
     .id             = AV_CODEC_ID_A64_MULTI5,
     .priv_data_size = sizeof(A64Context),
@@ -420,6 +417,7 @@ 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},
+    .long_name      = NULL_IF_CONFIG_SMALL("Multicolor charset for Commodore 64, extended with 5th color (colram)"),
     .capabilities   = CODEC_CAP_DELAY,
 };
 #endif

libavcodec/aac.h

@@ -157,7 +157,7 @@ typedef struct LongTermPrediction {
 typedef struct IndividualChannelStream {
     uint8_t max_sfb;            ///< number of scalefactor bands per group
     enum WindowSequence window_sequence[2];
-    uint8_t use_kb_window[2];   ///< If set, use Kaiser-Bessel window, otherwise use a sine window.
+    uint8_t use_kb_window[2];   ///< If set, use Kaiser-Bessel window, otherwise use a sinus window.
     int num_window_groups;
     uint8_t group_len[8];
     LongTermPrediction ltp;
@@ -234,7 +234,7 @@ typedef struct SingleChannelElement {
     int sf_idx[128];                                ///< scalefactor indices (used by encoder)
     uint8_t zeroes[128];                            ///< band is not coded (used by encoder)
     DECLARE_ALIGNED(32, float,   coeffs)[1024];     ///< coefficients for IMDCT
-    DECLARE_ALIGNED(32, float,   saved)[1536];      ///< overlap
+    DECLARE_ALIGNED(32, float,   saved)[1024];      ///< overlap
     DECLARE_ALIGNED(32, float,   ret_buf)[2048];    ///< PCM output buffer
     DECLARE_ALIGNED(16, float,   ltp_state)[3072];  ///< time signal for LTP
     PredictorState predictor_state[MAX_PREDICTORS];
@@ -290,7 +290,6 @@ struct AACContext {
      */
     FFTContext mdct;
     FFTContext mdct_small;
-    FFTContext mdct_ld;
     FFTContext mdct_ltp;
     FmtConvertContext fmt_conv;
     AVFloatDSPContext fdsp;

libavcodec/aac_ac3_parser.h

@@ -28,13 +28,13 @@
 #include "parser.h"
 
 typedef enum {
-    AAC_AC3_PARSE_ERROR_SYNC        = -0x1030c0a,
-    AAC_AC3_PARSE_ERROR_BSID        = -0x2030c0a,
-    AAC_AC3_PARSE_ERROR_SAMPLE_RATE = -0x3030c0a,
-    AAC_AC3_PARSE_ERROR_FRAME_SIZE  = -0x4030c0a,
-    AAC_AC3_PARSE_ERROR_FRAME_TYPE  = -0x5030c0a,
-    AAC_AC3_PARSE_ERROR_CRC         = -0x6030c0a,
-    AAC_AC3_PARSE_ERROR_CHANNEL_CFG = -0x7030c0a,
+    AAC_AC3_PARSE_ERROR_SYNC        = -1,
+    AAC_AC3_PARSE_ERROR_BSID        = -2,
+    AAC_AC3_PARSE_ERROR_SAMPLE_RATE = -3,
+    AAC_AC3_PARSE_ERROR_FRAME_SIZE  = -4,
+    AAC_AC3_PARSE_ERROR_FRAME_TYPE  = -5,
+    AAC_AC3_PARSE_ERROR_CRC         = -6,
+    AAC_AC3_PARSE_ERROR_CHANNEL_CFG = -7,
 } AACAC3ParseError;
 
 typedef struct AACAC3ParseContext {

libavcodec/aac_adtstoasc_bsf.c

@@ -112,7 +112,7 @@ static int aac_adtstoasc_filter(AVBitStreamFilterContext *bsfc,
 }
 
 AVBitStreamFilter ff_aac_adtstoasc_bsf = {
-    .name           = "aac_adtstoasc",
-    .priv_data_size = sizeof(AACBSFContext),
-    .filter         = aac_adtstoasc_filter,
+    "aac_adtstoasc",
+    sizeof(AACBSFContext),
+    aac_adtstoasc_filter,
 };

libavcodec/aac_tablegen.h

@@ -35,7 +35,7 @@ void ff_aac_tableinit(void)
 {
     int i;
     for (i = 0; i < 428; i++)
-        ff_aac_pow2sf_tab[i] = pow(2, (i - POW_SF2_ZERO) / 4.0);
+        ff_aac_pow2sf_tab[i] = pow(2, (i - POW_SF2_ZERO) / 4.);
 }
 #endif /* CONFIG_HARDCODED_TABLES */
 

libavcodec/aaccoder.c

@@ -1113,25 +1113,25 @@ static void search_for_ms(AACEncContext *s, ChannelElement *cpe,
 }
 
 AACCoefficientsEncoder ff_aac_coders[AAC_CODER_NB] = {
-    [AAC_CODER_FAAC] = {
+    {
         search_for_quantizers_faac,
         encode_window_bands_info,
         quantize_and_encode_band,
         search_for_ms,
     },
-    [AAC_CODER_ANMR] = {
+    {
         search_for_quantizers_anmr,
         encode_window_bands_info,
         quantize_and_encode_band,
         search_for_ms,
     },
-    [AAC_CODER_TWOLOOP] = {
+    {
         search_for_quantizers_twoloop,
         codebook_trellis_rate,
         quantize_and_encode_band,
         search_for_ms,
     },
-    [AAC_CODER_FAST] = {
+    {
         search_for_quantizers_fast,
         encode_window_bands_info,
         quantize_and_encode_band,

libavcodec/aacdec.c

@@ -2,7 +2,6 @@
  * AAC decoder
  * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
  * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
- * Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
  *
  * AAC LATM decoder
  * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
@@ -194,9 +193,6 @@ static int frame_configure_elements(AVCodecContext *avctx)
 
     /* get output buffer */
     av_frame_unref(ac->frame);
-    if (!avctx->channels)
-        return 1;
-
     ac->frame->nb_samples = 2048;
     if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0)
         return ret;
@@ -749,7 +745,7 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
                                      MPEG4AudioConfig *m4ac,
                                      int channel_config)
 {
-    int extension_flag, ret, ep_config, res_flags;
+    int extension_flag, ret;
     uint8_t layout_map[MAX_ELEM_ID*4][3];
     int tags = 0;
 
@@ -795,90 +791,14 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
         case AOT_ER_AAC_LTP:
         case AOT_ER_AAC_SCALABLE:
         case AOT_ER_AAC_LD:
-            res_flags = get_bits(gb, 3);
-            if (res_flags) {
-                avpriv_report_missing_feature(avctx,
-                                              "AAC data resilience (flags %x)",
-                                              res_flags);
-                return AVERROR_PATCHWELCOME;
-            }
+            skip_bits(gb, 3);      /* aacSectionDataResilienceFlag
+                                    * aacScalefactorDataResilienceFlag
+                                    * aacSpectralDataResilienceFlag
+                                    */
             break;
         }
         skip_bits1(gb);    // extensionFlag3 (TBD in version 3)
     }
-    switch (m4ac->object_type) {
-    case AOT_ER_AAC_LC:
-    case AOT_ER_AAC_LTP:
-    case AOT_ER_AAC_SCALABLE:
-    case AOT_ER_AAC_LD:
-        ep_config = get_bits(gb, 2);
-        if (ep_config) {
-            avpriv_report_missing_feature(avctx,
-                                          "epConfig %d", ep_config);
-            return AVERROR_PATCHWELCOME;
-        }
-    }
-    return 0;
-}
-
-static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx,
-                                     GetBitContext *gb,
-                                     MPEG4AudioConfig *m4ac,
-                                     int channel_config)
-{
-    int ret, ep_config, res_flags;
-    uint8_t layout_map[MAX_ELEM_ID*4][3];
-    int tags = 0;
-    const int ELDEXT_TERM = 0;
-
-    m4ac->ps  = 0;
-    m4ac->sbr = 0;
-
-    if (get_bits1(gb)) { // frameLengthFlag
-        avpriv_request_sample(avctx, "960/120 MDCT window");
-        return AVERROR_PATCHWELCOME;
-    }
-
-    res_flags = get_bits(gb, 3);
-    if (res_flags) {
-        avpriv_report_missing_feature(avctx,
-                                      "AAC data resilience (flags %x)",
-                                      res_flags);
-        return AVERROR_PATCHWELCOME;
-    }
-
-    if (get_bits1(gb)) { // ldSbrPresentFlag
-        avpriv_report_missing_feature(avctx,
-                                      "Low Delay SBR");
-        return AVERROR_PATCHWELCOME;
-    }
-
-    while (get_bits(gb, 4) != ELDEXT_TERM) {
-        int len = get_bits(gb, 4);
-        if (len == 15)
-            len += get_bits(gb, 8);
-        if (len == 15 + 255)
-            len += get_bits(gb, 16);
-        if (get_bits_left(gb) < len * 8 + 4) {
-            av_log(ac->avctx, AV_LOG_ERROR, overread_err);
-            return AVERROR_INVALIDDATA;
-        }
-        skip_bits_long(gb, 8 * len);
-    }
-
-    if ((ret = set_default_channel_config(avctx, layout_map,
-                                          &tags, channel_config)))
-        return ret;
-
-    if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
-        return ret;
-
-    ep_config = get_bits(gb, 2);
-    if (ep_config) {
-        avpriv_report_missing_feature(avctx,
-                                      "epConfig %d", ep_config);
-        return AVERROR_PATCHWELCOME;
-    }
     return 0;
 }
 
@@ -920,13 +840,6 @@ static int decode_audio_specific_config(AACContext *ac,
                m4ac->sampling_index);
         return AVERROR_INVALIDDATA;
     }
-    if (m4ac->object_type == AOT_ER_AAC_LD &&
-        (m4ac->sampling_index < 3 || m4ac->sampling_index > 7)) {
-        av_log(avctx, AV_LOG_ERROR,
-               "invalid low delay sampling rate index %d\n",
-               m4ac->sampling_index);
-        return AVERROR_INVALIDDATA;
-    }
 
     skip_bits_long(&gb, i);
 
@@ -934,22 +847,15 @@ static int decode_audio_specific_config(AACContext *ac,
     case AOT_AAC_MAIN:
     case AOT_AAC_LC:
     case AOT_AAC_LTP:
-    case AOT_ER_AAC_LC:
-    case AOT_ER_AAC_LD:
         if ((ret = decode_ga_specific_config(ac, avctx, &gb,
                                             m4ac, m4ac->chan_config)) < 0)
             return ret;
         break;
-    case AOT_ER_AAC_ELD:
-        if ((ret = decode_eld_specific_config(ac, avctx, &gb,
-                                              m4ac, m4ac->chan_config)) < 0)
-            return ret;
-        break;
     default:
-        avpriv_report_missing_feature(avctx,
-                                      "Audio object type %s%d",
-                                      m4ac->sbr == 1 ? "SBR+" : "",
-                                      m4ac->object_type);
+        av_log(avctx, AV_LOG_ERROR,
+               "Audio object type %s%d is not supported.\n",
+               m4ac->sbr == 1 ? "SBR+" : "",
+               m4ac->object_type);
         return AVERROR(ENOSYS);
     }
 
@@ -1110,15 +1016,12 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
                     352);
 
     ff_mdct_init(&ac->mdct,       11, 1, 1.0 / (32768.0 * 1024.0));
-    ff_mdct_init(&ac->mdct_ld,    10, 1, 1.0 / (32768.0 * 512.0));
     ff_mdct_init(&ac->mdct_small,  8, 1, 1.0 / (32768.0 * 128.0));
     ff_mdct_init(&ac->mdct_ltp,   11, 0, -2.0 * 32768.0);
     // window initialization
     ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
-    ff_kbd_window_init(ff_aac_kbd_long_512,  4.0, 512);
     ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
     ff_init_ff_sine_windows(10);
-    ff_init_ff_sine_windows( 9);
     ff_init_ff_sine_windows( 7);
 
     cbrt_tableinit();
@@ -1185,25 +1088,14 @@ static void decode_ltp(LongTermPrediction *ltp,
 static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
                            GetBitContext *gb)
 {
-    int aot = ac->oc[1].m4ac.object_type;
-    if (aot != AOT_ER_AAC_ELD) {
-        if (get_bits1(gb)) {
-            av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
-            return AVERROR_INVALIDDATA;
-        }
-        ics->window_sequence[1] = ics->window_sequence[0];
-        ics->window_sequence[0] = get_bits(gb, 2);
-        if (aot == AOT_ER_AAC_LD &&
-            ics->window_sequence[0] != ONLY_LONG_SEQUENCE) {
-            av_log(ac->avctx, AV_LOG_ERROR,
-                   "AAC LD is only defined for ONLY_LONG_SEQUENCE but "
-                   "window sequence %d found.\n", ics->window_sequence[0]);
-            ics->window_sequence[0] = ONLY_LONG_SEQUENCE;
-            return AVERROR_INVALIDDATA;
-        }
-        ics->use_kb_window[1]   = ics->use_kb_window[0];
-        ics->use_kb_window[0]   = get_bits1(gb);
+    if (get_bits1(gb)) {
+        av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
+        return AVERROR_INVALIDDATA;
     }
+    ics->window_sequence[1] = ics->window_sequence[0];
+    ics->window_sequence[0] = get_bits(gb, 2);
+    ics->use_kb_window[1]   = ics->use_kb_window[0];
+    ics->use_kb_window[0]   = get_bits1(gb);
     ics->num_window_groups  = 1;
     ics->group_len[0]       = 1;
     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
@@ -1225,36 +1117,21 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
     } else {
         ics->max_sfb               = get_bits(gb, 6);
         ics->num_windows           = 1;
-        if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
-            ics->swb_offset        =     ff_swb_offset_512[ac->oc[1].m4ac.sampling_index];
-            ics->num_swb           =    ff_aac_num_swb_512[ac->oc[1].m4ac.sampling_index];
-            if (!ics->num_swb || !ics->swb_offset)
-                return AVERROR_BUG;
-        } else {
-            ics->swb_offset        =    ff_swb_offset_1024[ac->oc[1].m4ac.sampling_index];
-            ics->num_swb           =   ff_aac_num_swb_1024[ac->oc[1].m4ac.sampling_index];
-        }
+        ics->swb_offset            =    ff_swb_offset_1024[ac->oc[1].m4ac.sampling_index];
+        ics->num_swb               =   ff_aac_num_swb_1024[ac->oc[1].m4ac.sampling_index];
         ics->tns_max_bands         = ff_tns_max_bands_1024[ac->oc[1].m4ac.sampling_index];
-        if (aot != AOT_ER_AAC_ELD) {
-            ics->predictor_present     = get_bits1(gb);
-            ics->predictor_reset_group = 0;
-        }
+        ics->predictor_present     = get_bits1(gb);
+        ics->predictor_reset_group = 0;
         if (ics->predictor_present) {
-            if (aot == AOT_AAC_MAIN) {
+            if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
                 if (decode_prediction(ac, ics, gb)) {
                     goto fail;
                 }
-            } else if (aot == AOT_AAC_LC ||
-                       aot == AOT_ER_AAC_LC) {
+            } else if (ac->oc[1].m4ac.object_type == AOT_AAC_LC) {
                 av_log(ac->avctx, AV_LOG_ERROR,
                        "Prediction is not allowed in AAC-LC.\n");
                 goto fail;
             } else {
-                if (aot == AOT_ER_AAC_LD) {
-                    av_log(ac->avctx, AV_LOG_ERROR,
-                           "LTP in ER AAC LD not yet implemented.\n");
-                    return AVERROR_PATCHWELCOME;
-                }
                 if ((ics->ltp.present = get_bits(gb, 1)))
                     decode_ltp(&ics->ltp, gb, ics->max_sfb);
             }
@@ -1347,7 +1224,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
             int run_end = band_type_run_end[idx];
             if (band_type[idx] == ZERO_BT) {
                 for (; i < run_end; i++, idx++)
-                    sf[idx] = 0.0;
+                    sf[idx] = 0.;
             } else if ((band_type[idx] == INTENSITY_BT) ||
                        (band_type[idx] == INTENSITY_BT2)) {
                 for (; i < run_end; i++, idx++) {
@@ -1405,12 +1282,12 @@ static int decode_pulses(Pulse *pulse, GetBitContext *gb,
         return -1;
     pulse->pos[0]    = swb_offset[pulse_swb];
     pulse->pos[0]   += get_bits(gb, 5);
-    if (pulse->pos[0] >= swb_offset[num_swb])
+    if (pulse->pos[0] > 1023)
         return -1;
     pulse->amp[0]    = get_bits(gb, 4);
     for (i = 1; i < pulse->num_pulse; i++) {
         pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
-        if (pulse->pos[i] >= swb_offset[num_swb])
+        if (pulse->pos[i] > 1023)
             return -1;
         pulse->amp[i] = get_bits(gb, 4);
     }
@@ -1875,15 +1752,9 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
     TemporalNoiseShaping    *tns = &sce->tns;
     IndividualChannelStream *ics = &sce->ics;
     float *out = sce->coeffs;
-    int global_gain, eld_syntax, er_syntax, pulse_present = 0;
+    int global_gain, pulse_present = 0;
     int ret;
 
-    eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
-    er_syntax  = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC ||
-                 ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP ||
-                 ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD ||
-                 ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
-
     /* This assignment is to silence a GCC warning about the variable being used
      * uninitialized when in fact it always is.
      */
@@ -1905,7 +1776,7 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
 
     pulse_present = 0;
     if (!scale_flag) {
-        if (!eld_syntax && (pulse_present = get_bits1(gb))) {
+        if ((pulse_present = get_bits1(gb))) {
             if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
                 av_log(ac->avctx, AV_LOG_ERROR,
                        "Pulse tool not allowed in eight short sequence.\n");
@@ -1917,19 +1788,12 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
                 return AVERROR_INVALIDDATA;
             }
         }
-        tns->present = get_bits1(gb);
-        if (tns->present && !er_syntax)
-            if (decode_tns(ac, tns, gb, ics) < 0)
-                return AVERROR_INVALIDDATA;
-        if (!eld_syntax && get_bits1(gb)) {
+        if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
+            return AVERROR_INVALIDDATA;
+        if (get_bits1(gb)) {
             avpriv_request_sample(ac->avctx, "SSR");
             return AVERROR_PATCHWELCOME;
         }
-        // I see no textual basis in the spec for this occuring after SSR gain
-        // control, but this is what both reference and real implmentations do
-        if (tns->present && er_syntax)
-            if (decode_tns(ac, tns, gb, ics) < 0)
-                return AVERROR_INVALIDDATA;
     }
 
     if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
@@ -2021,9 +1885,8 @@ static void apply_intensity_stereo(AACContext *ac,
 static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
 {
     int i, ret, common_window, ms_present = 0;
-    int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
 
-    common_window = eld_syntax || get_bits1(gb);
+    common_window = get_bits1(gb);
     if (common_window) {
         if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
             return AVERROR_INVALIDDATA;
@@ -2105,7 +1968,7 @@ static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
         int idx  = 0;
         int cge  = 1;
         int gain = 0;
-        float gain_cache = 1.0;
+        float gain_cache = 1.;
         if (c) {
             cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
             gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
@@ -2502,81 +2365,6 @@ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
     }
 }
 
-static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
-{
-    IndividualChannelStream *ics = &sce->ics;
-    float *in    = sce->coeffs;
-    float *out   = sce->ret;
-    float *saved = sce->saved;
-    const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_512 : ff_sine_512;
-    float *buf  = ac->buf_mdct;
-
-    // imdct
-    ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
-
-    // window overlapping
-    ac->fdsp.vector_fmul_window(out, saved, buf, lwindow_prev, 256);
-
-    // buffer update
-    memcpy(saved, buf + 256, 256 * sizeof(float));
-}
-
-static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
-{
-    float *in    = sce->coeffs;
-    float *out   = sce->ret;
-    float *saved = sce->saved;
-    const float *const window = ff_aac_eld_window;
-    float *buf  = ac->buf_mdct;
-    int i;
-    const int n  = 512;
-    const int n2 = n >> 1;
-    const int n4 = n >> 2;
-
-    // Inverse transform, mapped to the conventional IMDCT by
-    // Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
-    // "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks,"
-    // Audio, Language and Image Processing, 2008. ICALIP 2008. International Conference on
-    // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
-    for (i = 0; i < n2; i+=2) {
-        float temp;
-        temp =  in[i    ]; in[i    ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
-        temp = -in[i + 1]; in[i + 1] =  in[n - 2 - i]; in[n - 2 - i] = temp;
-    }
-    ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
-    for (i = 0; i < n; i+=2) {
-        buf[i] = -buf[i];
-    }
-    // Like with the regular IMDCT at this point we still have the middle half
-    // of a transform but with even symmetry on the left and odd symmetry on
-    // the right
-
-    // window overlapping
-    // The spec says to use samples [0..511] but the reference decoder uses
-    // samples [128..639].
-    for (i = n4; i < n2; i ++) {
-        out[i - n4] =    buf[n2 - 1 - i]       * window[i       - n4] +
-                       saved[      i + n2]     * window[i +   n - n4] +
-                      -saved[  n + n2 - 1 - i] * window[i + 2*n - n4] +
-                      -saved[2*n + n2 + i]     * window[i + 3*n - n4];
-    }
-    for (i = 0; i < n2; i ++) {
-        out[n4 + i] =    buf[i]               * window[i + n2       - n4] +
-                      -saved[      n - 1 - i] * window[i + n2 +   n - n4] +
-                      -saved[  n + i]         * window[i + n2 + 2*n - n4] +
-                       saved[2*n + n - 1 - i] * window[i + n2 + 3*n - n4];
-    }
-    for (i = 0; i < n4; i ++) {
-        out[n2 + n4 + i] =    buf[      i + n2]     * window[i +   n - n4] +
-                           -saved[      n2 - 1 - i] * window[i + 2*n - n4] +
-                           -saved[  n + n2 + i]     * window[i + 3*n - n4];
-    }
-
-    // buffer update
-    memmove(saved + n, saved, 2 * n * sizeof(float));
-    memcpy( saved,       buf,     n * sizeof(float));
-}
-
 /**
  * Apply dependent channel coupling (applied before IMDCT).
  *
@@ -2673,17 +2461,6 @@ static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
 static void spectral_to_sample(AACContext *ac)
 {
     int i, type;
-    void (*imdct_and_window)(AACContext *ac, SingleChannelElement *sce);
-    switch (ac->oc[1].m4ac.object_type) {
-    case AOT_ER_AAC_LD:
-        imdct_and_window = imdct_and_windowing_ld;
-        break;
-    case AOT_ER_AAC_ELD:
-        imdct_and_window = imdct_and_windowing_eld;
-        break;
-    default:
-        imdct_and_window = ac->imdct_and_windowing;
-    }
     for (type = 3; type >= 0; type--) {
         for (i = 0; i < MAX_ELEM_ID; i++) {
             ChannelElement *che = ac->che[type][i];
@@ -2705,11 +2482,11 @@ static void spectral_to_sample(AACContext *ac)
                 if (type <= TYPE_CPE)
                     apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
                 if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
-                    imdct_and_window(ac, &che->ch[0]);
+                    ac->imdct_and_windowing(ac, &che->ch[0]);
                     if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
                         ac->update_ltp(ac, &che->ch[0]);
                     if (type == TYPE_CPE) {
-                        imdct_and_window(ac, &che->ch[1]);
+                        ac->imdct_and_windowing(ac, &che->ch[1]);
                         if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
                             ac->update_ltp(ac, &che->ch[1]);
                     }
@@ -2729,7 +2506,7 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
     int size;
     AACADTSHeaderInfo hdr_info;
     uint8_t layout_map[MAX_ELEM_ID*4][3];
-    int layout_map_tags, ret;
+    int layout_map_tags;
 
     size = avpriv_aac_parse_header(gb, &hdr_info);
     if (size > 0) {
@@ -2743,15 +2520,12 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
         push_output_configuration(ac);
         if (hdr_info.chan_config) {
             ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
-            if ((ret = set_default_channel_config(ac->avctx,
-                                                  layout_map,
-                                                  &layout_map_tags,
-                                                  hdr_info.chan_config)) < 0)
-                return ret;
-            if ((ret = output_configure(ac, layout_map, layout_map_tags,
-                                        FFMAX(ac->oc[1].status,
-                                              OC_TRIAL_FRAME), 0)) < 0)
-                return ret;
+            if (set_default_channel_config(ac->avctx, layout_map,
+                    &layout_map_tags, hdr_info.chan_config))
+                return -7;
+            if (output_configure(ac, layout_map, layout_map_tags,
+                                 FFMAX(ac->oc[1].status, OC_TRIAL_FRAME), 0))
+                return -7;
         } else {
             ac->oc[1].m4ac.chan_config = 0;
             /**
@@ -2785,66 +2559,6 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
     return size;
 }
 
-static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
-                               int *got_frame_ptr, GetBitContext *gb)
-{
-    AACContext *ac = avctx->priv_data;
-    ChannelElement *che;
-    int err, i;
-    int samples = 1024;
-    int chan_config = ac->oc[1].m4ac.chan_config;
-    int aot = ac->oc[1].m4ac.object_type;
-
-    if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD)
-        samples >>= 1;
-
-    ac->frame = data;
-
-    if ((err = frame_configure_elements(avctx)) < 0)
-        return err;
-
-    ac->tags_mapped = 0;
-
-    if (chan_config < 0 || chan_config >= 8) {
-        avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
-                              ac->oc[1].m4ac.chan_config);
-        return AVERROR_INVALIDDATA;
-    }
-    for (i = 0; i < tags_per_config[chan_config]; i++) {
-        const int elem_type = aac_channel_layout_map[chan_config-1][i][0];
-        const int elem_id   = aac_channel_layout_map[chan_config-1][i][1];
-        if (!(che=get_che(ac, elem_type, elem_id))) {
-            av_log(ac->avctx, AV_LOG_ERROR,
-                   "channel element %d.%d is not allocated\n",
-                   elem_type, elem_id);
-            return AVERROR_INVALIDDATA;
-        }
-        if (aot != AOT_ER_AAC_ELD)
-            skip_bits(gb, 4);
-        switch (elem_type) {
-        case TYPE_SCE:
-            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
-            break;
-        case TYPE_CPE:
-            err = decode_cpe(ac, gb, che);
-            break;
-        case TYPE_LFE:
-            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
-            break;
-        }
-        if (err < 0)
-            return err;
-    }
-
-    spectral_to_sample(ac);
-
-    ac->frame->nb_samples = samples;
-    *got_frame_ptr = 1;
-
-    skip_bits_long(gb, get_bits_left(gb));
-    return 0;
-}
-
 static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
                                 int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
 {
@@ -2858,19 +2572,22 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
     ac->frame = data;
 
     if (show_bits(gb, 12) == 0xfff) {
-        if ((err = parse_adts_frame_header(ac, gb)) < 0) {
+        if (parse_adts_frame_header(ac, gb) < 0) {
             av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
+            err = -1;
             goto fail;
         }
         if (ac->oc[1].m4ac.sampling_index > 12) {
             av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
-            err = AVERROR_INVALIDDATA;
+            err = -1;
             goto fail;
         }
     }
 
-    if ((err = frame_configure_elements(avctx)) < 0)
+    if (frame_configure_elements(avctx) < 0) {
+        err = -1;
         goto fail;
+    }
 
     ac->tags_mapped = 0;
     // parse
@@ -2881,7 +2598,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
             if (!(che=get_che(ac, elem_type, elem_id))) {
                 av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
                        elem_type, elem_id);
-                err = AVERROR_INVALIDDATA;
+                err = -1;
                 goto fail;
             }
             samples = 1024;
@@ -2939,7 +2656,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
                 elem_id += get_bits(gb, 8) - 1;
             if (get_bits_left(gb) < 8 * elem_id) {
                     av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
-                    err = AVERROR_INVALIDDATA;
+                    err = -1;
                     goto fail;
             }
             while (elem_id > 0)
@@ -2948,7 +2665,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
             break;
 
         default:
-            err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
+            err = -1; /* should not happen, but keeps compiler happy */
             break;
         }
 
@@ -2960,7 +2677,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
 
         if (get_bits_left(gb) < 3) {
             av_log(avctx, AV_LOG_ERROR, overread_err);
-            err = AVERROR_INVALIDDATA;
+            err = -1;
             goto fail;
         }
     }
@@ -3049,20 +2766,9 @@ static int aac_decode_frame(AVCodecContext *avctx, void *data,
     if (INT_MAX / 8 <= buf_size)
         return AVERROR_INVALIDDATA;
 
-    if ((err = init_get_bits(&gb, buf, buf_size * 8)) < 0)
-        return err;
+    init_get_bits(&gb, buf, buf_size * 8);
 
-    switch (ac->oc[1].m4ac.object_type) {
-    case AOT_ER_AAC_LC:
-    case AOT_ER_AAC_LTP:
-    case AOT_ER_AAC_LD:
-    case AOT_ER_AAC_ELD:
-        err = aac_decode_er_frame(avctx, data, got_frame_ptr, &gb);
-        break;
-    default:
-        err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt);
-    }
-    if (err < 0)
+    if ((err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt)) < 0)
         return err;
 
     buf_consumed = (get_bits_count(&gb) + 7) >> 3;
@@ -3088,7 +2794,6 @@ static av_cold int aac_decode_close(AVCodecContext *avctx)
 
     ff_mdct_end(&ac->mdct);
     ff_mdct_end(&ac->mdct_small);
-    ff_mdct_end(&ac->mdct_ld);
     ff_mdct_end(&ac->mdct_ltp);
     return 0;
 }
@@ -3401,13 +3106,13 @@ static const AVClass aac_decoder_class = {
 
 AVCodec ff_aac_decoder = {
     .name            = "aac",
-    .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,
+    .long_name       = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
     .sample_fmts     = (const enum AVSampleFormat[]) {
         AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
     },
@@ -3424,13 +3129,13 @@ AVCodec ff_aac_decoder = {
 */
 AVCodec ff_aac_latm_decoder = {
     .name            = "aac_latm",
-    .long_name       = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
     .type            = AVMEDIA_TYPE_AUDIO,
     .id              = AV_CODEC_ID_AAC_LATM,
     .priv_data_size  = sizeof(struct LATMContext),
     .init            = latm_decode_init,
     .close           = aac_decode_close,
     .decode          = latm_decode_frame,
+    .long_name       = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
     .sample_fmts     = (const enum AVSampleFormat[]) {
         AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
     },

libavcodec/aacenc.c

@@ -791,11 +791,7 @@ 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"},
-        {"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_coder", "", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = 2}, 0, AAC_CODER_NB-1, AACENC_FLAGS},
     {NULL}
 };
 
@@ -815,7 +811,6 @@ static const int mpeg4audio_sample_rates[16] = {
 
 AVCodec ff_aac_encoder = {
     .name           = "aac",
-    .long_name      = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = AV_CODEC_ID_AAC,
     .priv_data_size = sizeof(AACEncContext),
@@ -827,5 +822,6 @@ AVCodec ff_aac_encoder = {
                       CODEC_CAP_EXPERIMENTAL,
     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
                                                      AV_SAMPLE_FMT_NONE },
+    .long_name      = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
     .priv_class     = &aacenc_class,
 };

libavcodec/aacenc.h

@@ -30,14 +30,7 @@
 #include "audio_frame_queue.h"
 #include "psymodel.h"
 
-typedef enum AACCoder {
-    AAC_CODER_FAAC = 0,
-    AAC_CODER_ANMR,
-    AAC_CODER_TWOLOOP,
-    AAC_CODER_FAST,
-
-    AAC_CODER_NB,
-}AACCoder;
+#define AAC_CODER_NB 4
 
 typedef struct AACEncOptions {
     int stereo_mode;

libavcodec/aacps_tablegen.h

@@ -192,7 +192,7 @@ static void ps_tableinit(void)
     for (k = 0; k < NR_ALLPASS_BANDS34; k++) {
         double f_center, theta;
         if (k < FF_ARRAY_ELEMS(f_center_34))
-            f_center = f_center_34[k] / 24.0;
+            f_center = f_center_34[k] / 24.;
         else
             f_center = k - 26.5f;
         for (m = 0; m < PS_AP_LINKS; m++) {

libavcodec/aacpsdsp.c

@@ -1,20 +1,20 @@
 /*
  * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
  *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 

libavcodec/aacpsdsp.h

@@ -1,20 +1,20 @@
 /*
  * Copyright (c) 2012 Mans Rullgard
  *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 

libavcodec/aactab.c

@@ -34,17 +34,12 @@
 #include <stdint.h>
 
 DECLARE_ALIGNED(32, float,  ff_aac_kbd_long_1024)[1024];
-DECLARE_ALIGNED(32, float,  ff_aac_kbd_long_512 )[512];
 DECLARE_ALIGNED(32, float,  ff_aac_kbd_short_128)[128];
 
 const uint8_t ff_aac_num_swb_1024[] = {
     41, 41, 47, 49, 49, 51, 47, 47, 43, 43, 43, 40, 40
 };
 
-const uint8_t ff_aac_num_swb_512[] = {
-     0,  0,  0, 36, 36, 37, 31, 31,  0,  0,  0,  0,  0
-};
-
 const uint8_t ff_aac_num_swb_128[] = {
     12, 12, 12, 14, 14, 14, 15, 15, 15, 15, 15, 15, 15
 };
@@ -1119,14 +1114,6 @@ static const uint16_t swb_offset_1024_48[] = {
     928, 1024
 };
 
-static const uint16_t swb_offset_512_48[] = {
-      0,   4,   8,  12,  16,  20,  24,  28,
-     32,  36,  40,  44,  48,  52,  56,  60,
-     68,  76,  84,  92, 100, 112, 124, 136,
-    148, 164, 184, 208, 236, 268, 300, 332,
-    364, 396, 428, 460, 512
-};
-
 static const uint16_t swb_offset_128_48[] = {
      0,   4,   8,  12,  16,  20,  28,  36,
     44,  56,  68,  80,  96, 112, 128
@@ -1142,14 +1129,6 @@ static const uint16_t swb_offset_1024_32[] = {
     928, 960, 992, 1024
 };
 
-static const uint16_t swb_offset_512_32[] = {
-      0,   4,   8,  12,  16,  20,  24,  28,
-     32,  36,  40,  44,  48,  52,  56,  64,
-     72,  80,  88,  96, 108, 120, 132, 144,
-    160, 176, 192, 212, 236, 260, 288, 320,
-    352, 384, 416, 448, 480, 512
-};
-
 static const uint16_t swb_offset_1024_24[] = {
       0,   4,   8,  12,  16,  20,  24,  28,
      32,  36,  40,  44,  52,  60,  68,  76,
@@ -1159,13 +1138,6 @@ static const uint16_t swb_offset_1024_24[] = {
     600, 652, 704, 768, 832, 896, 960, 1024
 };
 
-static const uint16_t swb_offset_512_24[] = {
-      0,   4,   8,  12,  16,  20,  24,  28,
-     32,  36,  40,  44,  52,  60,  68,  80,
-     92, 104, 120, 140, 164, 192, 224, 256,
-    288, 320, 352, 384, 416, 448, 480, 512,
-};
-
 static const uint16_t swb_offset_128_24[] = {
      0,   4,   8,  12,  16,  20,  24,  28,
     36,  44,  52,  64,  76,  92, 108, 128
@@ -1207,14 +1179,6 @@ const uint16_t * const ff_swb_offset_1024[] = {
     swb_offset_1024_8
 };
 
-const uint16_t * const ff_swb_offset_512[] = {
-    NULL,               NULL,               NULL,
-    swb_offset_512_48,  swb_offset_512_48,  swb_offset_512_32,
-    swb_offset_512_24,  swb_offset_512_24,  NULL,
-    NULL,               NULL,               NULL,
-    NULL
-};
-
 const uint16_t * const ff_swb_offset_128[] = {
     /* The last entry on the following row is swb_offset_128_64 but is a
        duplicate of swb_offset_128_96. */
@@ -1241,486 +1205,3 @@ const uint8_t ff_tns_max_bands_128[] = {
     9, 9, 10, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14
 };
 // @}
-
-const DECLARE_ALIGNED(32, float, ff_aac_eld_window)[1920] = {
-     0.00338834,  0.00567745,  0.00847677,  0.01172641,
-     0.01532555,  0.01917664,  0.02318809,  0.02729259,
-     0.03144503,  0.03560261,  0.03972499,  0.04379783,
-     0.04783094,  0.05183357,  0.05581342,  0.05977723,
-     0.06373173,  0.06768364,  0.07163937,  0.07559976,
-     0.07956096,  0.08352024,  0.08747623,  0.09143035,
-     0.09538618,  0.09934771,  0.10331917,  0.10730456,
-     0.11130697,  0.11532867,  0.11937133,  0.12343922,
-     0.12753911,  0.13167705,  0.13585812,  0.14008529,
-     0.14435986,  0.14868291,  0.15305531,  0.15747594,
-     0.16194193,  0.16645070,  0.17099991,  0.17558633,
-     0.18020600,  0.18485548,  0.18953191,  0.19423322,
-     0.19895800,  0.20370512,  0.20847374,  0.21326312,
-     0.21807244,  0.22290083,  0.22774742,  0.23261210,
-     0.23749542,  0.24239767,  0.24731889,  0.25225887,
-     0.25721719,  0.26219330,  0.26718648,  0.27219630,
-     0.27722262,  0.28226514,  0.28732336,  0.29239628,
-     0.29748247,  0.30258055,  0.30768914,  0.31280508,
-     0.31792385,  0.32304172,  0.32815579,  0.33326397,
-     0.33836470,  0.34345661,  0.34853868,  0.35361188,
-     0.35867865,  0.36374072,  0.36879900,  0.37385347,
-     0.37890349,  0.38394836,  0.38898730,  0.39401912,
-     0.39904236,  0.40405575,  0.40905820,  0.41404819,
-     0.41902398,  0.42398423,  0.42892805,  0.43385441,
-     0.43876210,  0.44365014,  0.44851786,  0.45336632,
-     0.45819759,  0.46301302,  0.46781309,  0.47259722,
-     0.47736435,  0.48211365,  0.48684450,  0.49155594,
-     0.49624679,  0.50091636,  0.50556440,  0.51019132,
-     0.51479771,  0.51938391,  0.52394998,  0.52849587,
-     0.53302151,  0.53752680,  0.54201160,  0.54647575,
-     0.55091916,  0.55534181,  0.55974376,  0.56412513,
-     0.56848615,  0.57282710,  0.57714834,  0.58145030,
-     0.58492489,  0.58918511,  0.59342326,  0.59763936,
-     0.60183347,  0.60600561,  0.61015581,  0.61428412,
-     0.61839056,  0.62247517,  0.62653799,  0.63057912,
-     0.63459872,  0.63859697,  0.64257403,  0.64653001,
-     0.65046495,  0.65437887,  0.65827181,  0.66214383,
-     0.66599499,  0.66982535,  0.67363499,  0.67742394,
-     0.68119219,  0.68493972,  0.68866653,  0.69237258,
-     0.69605778,  0.69972207,  0.70336537,  0.70698758,
-     0.71058862,  0.71416837,  0.71772674,  0.72126361,
-     0.72477889,  0.72827246,  0.73174419,  0.73519392,
-     0.73862141,  0.74202643,  0.74540874,  0.74876817,
-     0.75210458,  0.75541785,  0.75870785,  0.76197437,
-     0.76521709,  0.76843570,  0.77162988,  0.77479939,
-     0.77794403,  0.78106359,  0.78415789,  0.78722670,
-     0.79026979,  0.79328694,  0.79627791,  0.79924244,
-     0.80218027,  0.80509112,  0.80797472,  0.81083081,
-     0.81365915,  0.81645949,  0.81923160,  0.82197528,
-     0.82469037,  0.82737673,  0.83003419,  0.83266262,
-     0.83526186,  0.83783176,  0.84037217,  0.84288297,
-     0.84536401,  0.84781517,  0.85023632,  0.85262739,
-     0.85498836,  0.85731921,  0.85961993,  0.86189052,
-     0.86413101,  0.86634140,  0.86852173,  0.87067211,
-     0.87279275,  0.87488384,  0.87694559,  0.87897824,
-     0.88098206,  0.88295729,  0.88490423,  0.88682332,
-     0.88871519,  0.89058048,  0.89241983,  0.89423391,
-     0.89602338,  0.89778893,  0.89953126,  0.90125142,
-     0.90295086,  0.90463104,  0.90629341,  0.90793946,
-     0.90957067,  0.91118856,  0.91279464,  0.91439073,
-     0.91597898,  0.91756153,  0.91914049,  0.92071690,
-     0.92229070,  0.92386182,  0.92542993,  0.92698946,
-     0.92852960,  0.93003929,  0.93150727,  0.93291739,
-     0.93424863,  0.93547974,  0.93658982,  0.93756587,
-     0.93894072,  0.93922780,  0.93955477,  0.93991290,
-     0.94029104,  0.94067794,  0.94106258,  0.94144084,
-     0.94181549,  0.94218963,  0.94256628,  0.94294662,
-     0.94332998,  0.94371562,  0.94410280,  0.94449122,
-     0.94488106,  0.94527249,  0.94566568,  0.94606074,
-     0.94645772,  0.94685665,  0.94725759,  0.94766054,
-     0.94806547,  0.94847234,  0.94888115,  0.94929190,
-     0.94970469,  0.95011960,  0.95053672,  0.95095604,
-     0.95137751,  0.95180105,  0.95222658,  0.95265413,
-     0.95308380,  0.95351571,  0.95394994,  0.95438653,
-     0.95482538,  0.95526643,  0.95570958,  0.95615486,
-     0.95660234,  0.95705214,  0.95750433,  0.95795892,
-     0.95841582,  0.95887493,  0.95933616,  0.95979949,
-     0.96026500,  0.96073277,  0.96120286,  0.96167526,
-     0.96214986,  0.96262655,  0.96310522,  0.96358586,
-     0.96406853,  0.96455330,  0.96504026,  0.96552936,
-     0.96602051,  0.96651360,  0.96700850,  0.96750520,
-     0.96800376,  0.96850424,  0.96900670,  0.96951112,
-     0.97001738,  0.97052533,  0.97103488,  0.97154597,
-     0.97205867,  0.97257304,  0.97308915,  0.97360694,
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-    -0.00835183, -0.00849485, -0.00863926, -0.00878522,
-    -0.00893293, -0.00908260, -0.00923444, -0.00938864,
-    -0.00954537, -0.00970482, -0.00986715, -0.01003173,
-    -0.01019711, -0.01036164, -0.01052357, -0.01068184,
-    -0.01083622, -0.01098652, -0.01113252, -0.01127409,
-    -0.01141114, -0.01154358, -0.01167135, -0.01179439,
-    -0.01191268, -0.01202619, -0.01213493, -0.01223891,
-    -0.01233817, -0.01243275, -0.01252272, -0.01260815,
-    -0.01268915, -0.01276583, -0.01283832, -0.01290685,
-    -0.01297171, -0.01303320, -0.01309168, -0.01314722,
-    -0.01319969, -0.01324889, -0.01329466, -0.01333693,
-    -0.01337577, -0.01341125, -0.01344345, -0.01347243,
-    -0.01349823, -0.01352089, -0.01354045, -0.01355700,
-    -0.01357068, -0.01358164, -0.01359003, -0.01359587,
-    -0.01359901, -0.01359931, -0.01359661, -0.01359087,
-    -0.01358219, -0.01357065, -0.01355637, -0.01353935,
-    -0.01351949, -0.01349670, -0.01347088, -0.01344214,
-    -0.01341078, -0.01337715, -0.01334158, -0.01330442,
-    -0.01326601, -0.01322671, -0.01318689, -0.01314692,
-    -0.01310123, -0.01306470, -0.01302556, -0.01298381,
-    -0.01293948, -0.01289255, -0.01284305, -0.01279095,
-    -0.01273625, -0.01267893, -0.01261897, -0.01255632,
-    -0.01249096, -0.01242283, -0.01235190, -0.01227827,
-    -0.01220213, -0.01212366, -0.01204304, -0.01196032,
-    -0.01187543, -0.01178829, -0.01169884, -0.01160718,
-    -0.01151352, -0.01141809, -0.01132111, -0.01122272,
-    -0.01112304, -0.01102217, -0.01092022, -0.01081730,
-    -0.01071355, -0.01060912, -0.01050411, -0.01039854,
-    -0.01029227, -0.01018521, -0.01007727, -0.00996859,
-    -0.00985959, -0.00975063, -0.00964208, -0.00953420,
-    -0.00942723, -0.00932135, -0.00921677, -0.00911364,
-    -0.00901208, -0.00891220, -0.00881412, -0.00871792,
-    -0.00862369, -0.00853153, -0.00844149, -0.00835360,
-    -0.00826785, -0.00818422, -0.00810267, -0.00802312,
-    -0.00794547, -0.00786959, -0.00779533, -0.00772165,
-    -0.00764673, -0.00756886, -0.00748649, -0.00739905,
-    -0.00730681, -0.00721006, -0.00710910, -0.00700419,
-    -0.00689559, -0.00678354, -0.00666829, -0.00655007,
-    -0.00642916, -0.00630579, -0.00618022, -0.00605267,
-    -0.00592333, -0.00579240, -0.00566006, -0.00552651,
-    -0.00539194, -0.00525653, -0.00512047, -0.00498390,
-    -0.00484693, -0.00470969, -0.00457228, -0.00443482,
-    -0.00429746, -0.00416034, -0.00402359, -0.00388738,
-    -0.00375185, -0.00361718, -0.00348350, -0.00335100,
-    -0.00321991, -0.00309043, -0.00296276, -0.00283698,
-    -0.00271307, -0.00259098, -0.00247066, -0.00235210,
-    -0.00223531, -0.00212030, -0.00200709, -0.00189576,
-    -0.00178647, -0.00167936, -0.00157457, -0.00147216,
-    -0.00137205, -0.00127418, -0.00117849, -0.00108498,
-    -0.00099375, -0.00090486, -0.00081840, -0.00073444,
-    -0.00065309, -0.00057445, -0.00049860, -0.00042551,
-    -0.00035503, -0.00028700, -0.00022125, -0.00015761,
-    -0.00009588, -0.00003583,  0.00002272,  0.00007975,
-     0.00013501,  0.00018828,  0.00023933,  0.00028784,
-     0.00033342,  0.00037572,  0.00041438,  0.00044939,
-     0.00048103,  0.00050958,  0.00053533,  0.00055869,
-     0.00058015,  0.00060022,  0.00061935,  0.00063781,
-     0.00065568,  0.00067303,  0.00068991,  0.00070619,
-     0.00072155,  0.00073567,  0.00074826,  0.00075912,
-     0.00076811,  0.00077509,  0.00077997,  0.00078275,
-     0.00078351,  0.00078237,  0.00077943,  0.00077484,
-     0.00076884,  0.00076160,  0.00075335,  0.00074423,
-     0.00073442,  0.00072404,  0.00071323,  0.00070209,
-     0.00069068,  0.00067906,  0.00066728,  0.00065534,
-     0.00064321,  0.00063086,  0.00061824,  0.00060534,
-     0.00059211,  0.00057855,  0.00056462,  0.00055033,
-     0.00053566,  0.00052063,  0.00050522,  0.00048949,
-     0.00047349,  0.00045728,  0.00044092,  0.00042447,
-     0.00040803,  0.00039166,  0.00037544,  0.00035943,
-     0.00034371,  0.00032833,  0.00031333,  0.00029874,
-     0.00028452,  0.00027067,  0.00025715,  0.00024395,
-     0.00023104,  0.00021842,  0.00020606,  0.00019398,
-     0.00018218,  0.00017069,  0.00015953,  0.00014871,
-     0.00013827,  0.00012823,  0.00011861,  0.00010942,
-     0.00010067,  0.00009236,  0.00008448,  0.00007703,
-     0.00006999,  0.00006337,  0.00005714,  0.00005129,
-     0.00004583,  0.00004072,  0.00003597,  0.00003157,
-     0.00002752,  0.00002380,  0.00002042,  0.00001736,
-     0.00001461,  0.00001215,  0.00000998,  0.00000807,
-     0.00000641,  0.00000499,  0.00000378,  0.00000278,
-     0.00000196,  0.00000132,  0.00000082,  0.00000046,
-     0.00000020,  0.00000005, -0.00000003, -0.00000006,
-    -0.00000004, -0.00000001,  0.00000001,  0.00000001,
-     0.00000001,  0.00000001, -0.00000001, -0.00000004,
-    -0.00000005, -0.00000003,  0.00000005,  0.00000020,
-     0.00000043,  0.00000077,  0.00000123,  0.00000183,
-     0.00000257,  0.00000348,  0.00000455,  0.00000581,
-     0.00000727,  0.00000893,  0.00001080,  0.00001290,
-     0.00001522,  0.00001778,  0.00002057,  0.00002362,
-     0.00002691,  0.00003044,  0.00003422,  0.00003824,
-     0.00004250,  0.00004701,  0.00005176,  0.00005676,
-     0.00006200,  0.00006749,  0.00007322,  0.00007920,
-     0.00008541,  0.00009186,  0.00009854,  0.00010543,
-     0.00011251,  0.00011975,  0.00012714,  0.00013465,
-     0.00014227,  0.00014997,  0.00015775,  0.00016558,
-     0.00017348,  0.00018144,  0.00018947,  0.00019756,
-     0.00020573,  0.00021399,  0.00022233,  0.00023076,
-     0.00023924,  0.00024773,  0.00025621,  0.00026462,
-     0.00027293,  0.00028108,  0.00028904,  0.00029675,
-     0.00030419,  0.00031132,  0.00031810,  0.00032453,
-     0.00033061,  0.00033632,  0.00034169,  0.00034672,
-     0.00035142,  0.00035580,  0.00035988,  0.00036369,
-     0.00036723,  0.00037053,  0.00037361,  0.00037647,
-     0.00037909,  0.00038145,  0.00038352,  0.00038527,
-     0.00038663,  0.00038757,  0.00038801,  0.00038790,
-     0.00038717,  0.00038572,  0.00038350,  0.00038044,
-     0.00037651,  0.00037170,  0.00036597,  0.00035936,
-     0.00035191,  0.00034370,  0.00033480,  0.00032531,
-     0.00031537,  0.00030512,  0.00029470,  0.00028417,
-     0.00027354,  0.00026279,  0.00025191,  0.00024081,
-     0.00022933,  0.00021731,  0.00020458,  0.00019101,
-     0.00017654,  0.00016106,  0.00014452,  0.00012694,
-     0.00010848,  0.00008929,  0.00006953,  0.00004935,
-     0.00002884,  0.00000813, -0.00001268, -0.00003357,
-    -0.00005457, -0.00007574, -0.00009714, -0.00011882,
-    -0.00014082, -0.00016318, -0.00018595, -0.00020912,
-    -0.00023265, -0.00025650, -0.00028060, -0.00030492,
-    -0.00032941, -0.00035400, -0.00037865, -0.00040333,
-    -0.00042804, -0.00045279, -0.00047759, -0.00050243,
-    -0.00052728, -0.00055209, -0.00057685, -0.00060153,
-    -0.00062611, -0.00065056, -0.00067485, -0.00069895,
-    -0.00072287, -0.00074660, -0.00077013, -0.00079345,
-    -0.00081653, -0.00083936, -0.00086192, -0.00088421,
-    -0.00090619, -0.00092786, -0.00094919, -0.00097017,
-    -0.00099077, -0.00101098, -0.00103077, -0.00105012,
-    -0.00106904, -0.00108750, -0.00110549, -0.00112301,
-    -0.00114005, -0.00115660, -0.00117265, -0.00118821,
-    -0.00120325, -0.00121779, -0.00123180, -0.00124528,
-    -0.00125822, -0.00127061, -0.00128243, -0.00129368,
-    -0.00130435, -0.00131445, -0.00132395, -0.00133285,
-    -0.00134113, -0.00134878, -0.00135577, -0.00136215,
-    -0.00136797, -0.00137333, -0.00137834, -0.00138305,
-    -0.00138748, -0.00139163, -0.00139551, -0.00139913,
-    -0.00140249, -0.00140559, -0.00140844, -0.00141102,
-    -0.00141334, -0.00141538, -0.00141714, -0.00141861,
-    -0.00141978, -0.00142064, -0.00142117, -0.00142138,
-    -0.00142125, -0.00142077, -0.00141992, -0.00141870,
-    -0.00141710, -0.00141510, -0.00141268, -0.00140986,
-    -0.00140663, -0.00140301, -0.00139900, -0.00139460,
-    -0.00138981, -0.00138464, -0.00137908, -0.00137313,
-    -0.00136680, -0.00136010, -0.00135301, -0.00134555,
-    -0.00133772, -0.00132952, -0.00132095, -0.00131201,
-    -0.00130272, -0.00129307, -0.00128309, -0.00127277,
-    -0.00126211, -0.00125113, -0.00123981, -0.00122817,
-    -0.00121622, -0.00120397, -0.00119141, -0.00117859,
-    -0.00116552, -0.00115223, -0.00113877, -0.00112517,
-    -0.00111144, -0.00109764, -0.00108377, -0.00106989,
-};

libavcodec/aactab.h

@@ -45,16 +45,13 @@
  * @{
  */
 DECLARE_ALIGNED(32, extern float,  ff_aac_kbd_long_1024)[1024];
-DECLARE_ALIGNED(32, extern float,  ff_aac_kbd_long_512 )[512];
 DECLARE_ALIGNED(32, extern float,  ff_aac_kbd_short_128)[128];
-const DECLARE_ALIGNED(32, extern float, ff_aac_eld_window)[1920];
 // @}
 
 /* @name number of scalefactor window bands for long and short transform windows respectively
  * @{
  */
 extern const uint8_t ff_aac_num_swb_1024[];
-extern const uint8_t ff_aac_num_swb_512 [];
 extern const uint8_t ff_aac_num_swb_128 [];
 // @}
 
@@ -72,7 +69,6 @@ extern const float *ff_aac_codebook_vector_vals[];
 extern const uint16_t *ff_aac_codebook_vector_idx[];
 
 extern const uint16_t * const ff_swb_offset_1024[13];
-extern const uint16_t * const ff_swb_offset_512 [13];
 extern const uint16_t * const ff_swb_offset_128 [13];
 
 extern const uint8_t ff_tns_max_bands_1024[13];

libavcodec/aasc.c

@@ -107,9 +107,11 @@ static int aasc_decode_frame(AVCodecContext *avctx,
     switch (compr) {
     case 0:
         stride = (avctx->width * psize + psize) & ~psize;
-        if (buf_size < stride * avctx->height)
-            return AVERROR_INVALIDDATA;
         for (i = avctx->height - 1; i >= 0; i--) {
+            if (avctx->width * psize > buf_size) {
+                av_log(avctx, AV_LOG_ERROR, "Next line is beyond buffer bounds\n");
+                break;
+            }
             memcpy(s->frame->data[0] + i * s->frame->linesize[0], buf, avctx->width * psize);
             buf += stride;
             buf_size -= stride;
@@ -151,7 +153,6 @@ static av_cold int aasc_decode_end(AVCodecContext *avctx)
 
 AVCodec ff_aasc_decoder = {
     .name           = "aasc",
-    .long_name      = NULL_IF_CONFIG_SMALL("Autodesk RLE"),
     .type           = AVMEDIA_TYPE_VIDEO,
     .id             = AV_CODEC_ID_AASC,
     .priv_data_size = sizeof(AascContext),
@@ -159,4 +160,5 @@ AVCodec ff_aasc_decoder = {
     .close          = aasc_decode_end,
     .decode         = aasc_decode_frame,
     .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("Autodesk RLE"),
 };

libavcodec/ac3dec.c

@@ -290,7 +290,7 @@ static int parse_frame_header(AC3DecodeContext *s)
         return ff_eac3_parse_header(s);
     } else {
         av_log(s->avctx, AV_LOG_ERROR, "E-AC-3 support not compiled in\n");
-        return AVERROR(ENOSYS);
+        return -1;
     }
 }
 
@@ -429,7 +429,7 @@ static void ac3_decode_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, ma
     int end_freq   = s->end_freq[ch_index];
     uint8_t *baps  = s->bap[ch_index];
     int8_t *exps   = s->dexps[ch_index];
-    int32_t *coeffs = s->fixed_coeffs[ch_index];
+    int32_t *coeffs    = s->fixed_coeffs[ch_index];
     int dither     = (ch_index == CPL_CH) || s->dither_flag[ch_index];
     GetBitContext *gbc = &s->gbc;
     int freq;
@@ -488,10 +488,6 @@ static void ac3_decode_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, ma
             break;
         default: /* 6 to 15 */
             /* Shift mantissa and sign-extend it. */
-            if (bap > 15) {
-                av_log(s->avctx, AV_LOG_ERROR, "bap %d is invalid in plain AC-3\n", bap);
-                bap = 15;
-            }
             mantissa = get_sbits(gbc, quantization_tab[bap]);
             mantissa <<= 24 - quantization_tab[bap];
             break;
@@ -790,12 +786,12 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
             if (start_subband >= end_subband) {
                 av_log(s->avctx, AV_LOG_ERROR, "invalid spectral extension "
                        "range (%d >= %d)\n", start_subband, end_subband);
-                return AVERROR_INVALIDDATA;
+                return -1;
             }
             if (dst_start_freq >= src_start_freq) {
                 av_log(s->avctx, AV_LOG_ERROR, "invalid spectral extension "
                        "copy start bin (%d >= %d)\n", dst_start_freq, src_start_freq);
-                return AVERROR_INVALIDDATA;
+                return -1;
             }
 
             s->spx_dst_start_freq = dst_start_freq;
@@ -872,7 +868,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
 
             if (channel_mode < AC3_CHMODE_STEREO) {
                 av_log(s->avctx, AV_LOG_ERROR, "coupling not allowed in mono or dual-mono\n");
-                return AVERROR_INVALIDDATA;
+                return -1;
             }
 
             /* check for enhanced coupling */
@@ -902,7 +898,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
             if (cpl_start_subband >= cpl_end_subband) {
                 av_log(s->avctx, AV_LOG_ERROR, "invalid coupling range (%d >= %d)\n",
                        cpl_start_subband, cpl_end_subband);
-                return AVERROR_INVALIDDATA;
+                return -1;
             }
             s->start_freq[CPL_CH] = cpl_start_subband * 12 + 37;
             s->end_freq[CPL_CH]   = cpl_end_subband   * 12 + 37;
@@ -924,7 +920,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
         if (!blk) {
             av_log(s->avctx, AV_LOG_ERROR, "new coupling strategy must "
                    "be present in block 0\n");
-            return AVERROR_INVALIDDATA;
+            return -1;
         } else {
             s->cpl_in_use[blk] = s->cpl_in_use[blk-1];
         }
@@ -954,7 +950,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
                 } else if (!blk) {
                     av_log(s->avctx, AV_LOG_ERROR, "new coupling coordinates must "
                            "be present in block 0\n");
-                    return AVERROR_INVALIDDATA;
+                    return -1;
                 }
             } else {
                 /* channel not in coupling */
@@ -1009,7 +1005,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
                 int bandwidth_code = get_bits(gbc, 6);
                 if (bandwidth_code > 60) {
                     av_log(s->avctx, AV_LOG_ERROR, "bandwidth code = %d > 60\n", bandwidth_code);
-                    return AVERROR_INVALIDDATA;
+                    return -1;
                 }
                 s->end_freq[ch] = bandwidth_code * 3 + 73;
             }
@@ -1032,7 +1028,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
                                  s->num_exp_groups[ch], s->dexps[ch][0],
                                  &s->dexps[ch][s->start_freq[ch]+!!ch])) {
                 av_log(s->avctx, AV_LOG_ERROR, "exponent out-of-range\n");
-                return AVERROR_INVALIDDATA;
+                return -1;
             }
             if (ch != CPL_CH && ch != s->lfe_ch)
                 skip_bits(gbc, 2); /* skip gainrng */
@@ -1052,7 +1048,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
         } else if (!blk) {
             av_log(s->avctx, AV_LOG_ERROR, "new bit allocation info must "
                    "be present in block 0\n");
-            return AVERROR_INVALIDDATA;
+            return -1;
         }
     }
 
@@ -1083,7 +1079,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
             }
         } else if (!s->eac3 && !blk) {
             av_log(s->avctx, AV_LOG_ERROR, "new snr offsets must be present in block 0\n");
-            return AVERROR_INVALIDDATA;
+            return -1;
         }
     }
 
@@ -1122,7 +1118,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
         } else if (!s->eac3 && !blk) {
             av_log(s->avctx, AV_LOG_ERROR, "new coupling leak info must "
                    "be present in block 0\n");
-            return AVERROR_INVALIDDATA;
+            return -1;
         }
         s->first_cpl_leak = 0;
     }
@@ -1134,7 +1130,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
             s->dba_mode[ch] = get_bits(gbc, 2);
             if (s->dba_mode[ch] == DBA_RESERVED) {
                 av_log(s->avctx, AV_LOG_ERROR, "delta bit allocation strategy reserved\n");
-                return AVERROR_INVALIDDATA;
+                return -1;
             }
             bit_alloc_stages[ch] = FFMAX(bit_alloc_stages[ch], 2);
         }
@@ -1175,7 +1171,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
                                            s->dba_offsets[ch], s->dba_lengths[ch],
                                            s->dba_values[ch],  s->mask[ch])) {
                 av_log(s->avctx, AV_LOG_ERROR, "error in bit allocation\n");
-                return AVERROR_INVALIDDATA;
+                return -1;
             }
         }
         if (bit_alloc_stages[ch] > 0) {
@@ -1295,7 +1291,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
         switch (err) {
         case AAC_AC3_PARSE_ERROR_SYNC:
             av_log(avctx, AV_LOG_ERROR, "frame sync error\n");
-            return AVERROR_INVALIDDATA;
+            return -1;
         case AAC_AC3_PARSE_ERROR_BSID:
             av_log(avctx, AV_LOG_ERROR, "invalid bitstream id\n");
             break;
@@ -1309,20 +1305,17 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
             /* skip frame if CRC is ok. otherwise use error concealment. */
             /* TODO: add support for substreams and dependent frames */
             if (s->frame_type == EAC3_FRAME_TYPE_DEPENDENT || s->substreamid) {
-                av_log(avctx, AV_LOG_WARNING, "unsupported frame type : "
+                av_log(avctx, AV_LOG_ERROR, "unsupported frame type : "
                        "skipping frame\n");
                 *got_frame_ptr = 0;
-                return buf_size;
+                return s->frame_size;
             } else {
                 av_log(avctx, AV_LOG_ERROR, "invalid frame type\n");
             }
             break;
-        case AAC_AC3_PARSE_ERROR_CRC:
-        case AAC_AC3_PARSE_ERROR_CHANNEL_CFG:
+        default:
+            av_log(avctx, AV_LOG_ERROR, "invalid header\n");
             break;
-        default: // Normal AVERROR do not try to recover.
-            *got_frame_ptr = 0;
-            return err;
         }
     } else {
         /* check that reported frame size fits in input buffer */
@@ -1381,7 +1374,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
         avctx->audio_service_type = AV_AUDIO_SERVICE_TYPE_KARAOKE;
 
     /* get output buffer */
-    frame->nb_samples = s->num_blocks * AC3_BLOCK_SIZE;
+    frame->nb_samples = s->num_blocks * 256;
     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
         return ret;
 
@@ -1402,7 +1395,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
         }
         if (err)
             for (ch = 0; ch < s->out_channels; ch++)
-                memcpy(((float*)frame->data[ch]) + AC3_BLOCK_SIZE*blk, output[ch], sizeof(**output) * AC3_BLOCK_SIZE);
+                memcpy(((float*)frame->data[ch]) + AC3_BLOCK_SIZE*blk, output[ch], 1024);
         for (ch = 0; ch < s->out_channels; ch++)
             output[ch] = s->outptr[channel_map[ch]];
         for (ch = 0; ch < s->out_channels; ch++) {
@@ -1415,7 +1408,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
 
     /* keep last block for error concealment in next frame */
     for (ch = 0; ch < s->out_channels; ch++)
-        memcpy(s->output[ch], output[ch], sizeof(**output) * AC3_BLOCK_SIZE);
+        memcpy(s->output[ch], output[ch], 1024);
 
     *got_frame_ptr = 1;
 
@@ -1457,7 +1450,6 @@ static const AVClass ac3_decoder_class = {
 
 AVCodec ff_ac3_decoder = {
     .name           = "ac3",
-    .long_name      = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = AV_CODEC_ID_AC3,
     .priv_data_size = sizeof (AC3DecodeContext),
@@ -1465,6 +1457,7 @@ AVCodec ff_ac3_decoder = {
     .close          = ac3_decode_end,
     .decode         = ac3_decode_frame,
     .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
                                                       AV_SAMPLE_FMT_NONE },
     .priv_class     = &ac3_decoder_class,
@@ -1480,7 +1473,6 @@ static const AVClass eac3_decoder_class = {
 
 AVCodec ff_eac3_decoder = {
     .name           = "eac3",
-    .long_name      = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"),
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = AV_CODEC_ID_EAC3,
     .priv_data_size = sizeof (AC3DecodeContext),
@@ -1488,6 +1480,7 @@ AVCodec ff_eac3_decoder = {
     .close          = ac3_decode_end,
     .decode         = ac3_decode_frame,
     .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"),
     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
                                                       AV_SAMPLE_FMT_NONE },
     .priv_class     = &eac3_decoder_class,

libavcodec/ac3enc.h

@@ -3,20 +3,20 @@
  * Copyright (c) 2000 Fabrice Bellard
  * Copyright (c) 2006-2010 Justin Ruggles <justin.ruggles@gmail.com>
  *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 

libavcodec/ac3enc_fixed.c

@@ -154,7 +154,6 @@ static av_cold int ac3_fixed_encode_init(AVCodecContext *avctx)
 
 AVCodec ff_ac3_fixed_encoder = {
     .name            = "ac3_fixed",
-    .long_name       = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
     .type            = AVMEDIA_TYPE_AUDIO,
     .id              = AV_CODEC_ID_AC3,
     .priv_data_size  = sizeof(AC3EncodeContext),
@@ -163,6 +162,7 @@ AVCodec ff_ac3_fixed_encoder = {
     .close           = ff_ac3_encode_close,
     .sample_fmts     = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16P,
                                                       AV_SAMPLE_FMT_NONE },
+    .long_name       = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
     .priv_class      = &ac3enc_class,
     .channel_layouts = ff_ac3_channel_layouts,
     .defaults        = ac3_defaults,

libavcodec/ac3enc_float.c

@@ -152,7 +152,6 @@ static CoefType calc_cpl_coord(CoefSumType energy_ch, CoefSumType energy_cpl)
 #if CONFIG_AC3_ENCODER
 AVCodec ff_ac3_encoder = {
     .name            = "ac3",
-    .long_name       = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
     .type            = AVMEDIA_TYPE_AUDIO,
     .id              = AV_CODEC_ID_AC3,
     .priv_data_size  = sizeof(AC3EncodeContext),
@@ -161,6 +160,7 @@ AVCodec ff_ac3_encoder = {
     .close           = ff_ac3_encode_close,
     .sample_fmts     = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
                                                       AV_SAMPLE_FMT_NONE },
+    .long_name       = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
     .priv_class      = &ac3enc_class,
     .channel_layouts = ff_ac3_channel_layouts,
     .defaults        = ac3_defaults,

libavcodec/ac3enc_opts_template.c

@@ -2,20 +2,20 @@
  * AC-3 encoder options
  * Copyright (c) 2011 Justin Ruggles <justin.ruggles@gmail.com>
  *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 

libavcodec/ac3enc_template.c

@@ -361,7 +361,7 @@ static void compute_rematrixing_strategy(AC3EncodeContext *s)
         }
 
         for (bnd = 0; bnd < block->num_rematrixing_bands; bnd++) {
-            /* calculate sum of squared coeffs for one band in one block */
+            /* calculate calculate sum of squared coeffs for one band in one block */
             int start = ff_ac3_rematrix_band_tab[bnd];
             int end   = FFMIN(nb_coefs, ff_ac3_rematrix_band_tab[bnd+1]);
             CoefSumType sum[4];

libavcodec/ac3tab.c

@@ -114,7 +114,7 @@ const uint8_t ff_ac3_enc_channel_map[8][2][6] = {
 };
 
 /**
- * Table to remap channels from AC-3 order to SMPTE order.
+ * Table to remap channels from from AC-3 order to SMPTE order.
  * [channel_mode][lfe][ch]
  */
 const uint8_t ff_ac3_dec_channel_map[8][2][6] = {

libavcodec/acelp_vectors.c

@@ -114,10 +114,10 @@ const float ff_b60_sinc[61] = {
  0.898529  ,  0.865051  ,  0.769257  ,  0.624054  ,  0.448639  ,  0.265289   ,
  0.0959167 , -0.0412598 , -0.134338  , -0.178986  , -0.178528  , -0.142609   ,
 -0.0849304 , -0.0205078 ,  0.0369568 ,  0.0773926 ,  0.0955200 ,  0.0912781  ,
- 0.0689392 ,  0.0357056 ,  0.0       , -0.0305481 , -0.0504150 , -0.0570068  ,
+ 0.0689392 ,  0.0357056 ,  0.        , -0.0305481 , -0.0504150 , -0.0570068  ,
 -0.0508423 , -0.0350037 , -0.0141602 ,  0.00665283,  0.0230713 ,  0.0323486  ,
  0.0335388 ,  0.0275879 ,  0.0167847 ,  0.00411987, -0.00747681, -0.0156860  ,
--0.0193481 , -0.0183716 , -0.0137634 , -0.00704956,  0.0       ,  0.00582886 ,
+-0.0193481 , -0.0183716 , -0.0137634 , -0.00704956,  0.        ,  0.00582886 ,
  0.00939941,  0.0103760 ,  0.00903320,  0.00604248,  0.00238037, -0.00109863 ,
 -0.00366211, -0.00497437, -0.00503540, -0.00402832, -0.00241089, -0.000579834,
  0.00103760,  0.00222778,  0.00277710,  0.00271606,  0.00213623,  0.00115967 ,

libavcodec/adpcm.c

@@ -1532,13 +1532,13 @@ static const enum AVSampleFormat sample_fmts_both[] = { AV_SAMPLE_FMT_S16,
 #define ADPCM_DECODER(id_, sample_fmts_, name_, long_name_) \
 AVCodec ff_ ## name_ ## _decoder = {                        \
     .name           = #name_,                               \
-    .long_name      = NULL_IF_CONFIG_SMALL(long_name_),     \
     .type           = AVMEDIA_TYPE_AUDIO,                   \
     .id             = id_,                                  \
     .priv_data_size = sizeof(ADPCMDecodeContext),           \
     .init           = adpcm_decode_init,                    \
     .decode         = adpcm_decode_frame,                   \
     .capabilities   = CODEC_CAP_DR1,                        \
+    .long_name      = NULL_IF_CONFIG_SMALL(long_name_),     \
     .sample_fmts    = sample_fmts_,                         \
 }
 

libavcodec/adpcmenc.c

@@ -708,7 +708,6 @@ static const enum AVSampleFormat sample_fmts_p[] = {
 #define ADPCM_ENCODER(id_, name_, sample_fmts_, long_name_) \
 AVCodec ff_ ## name_ ## _encoder = {                        \
     .name           = #name_,                               \
-    .long_name      = NULL_IF_CONFIG_SMALL(long_name_),     \
     .type           = AVMEDIA_TYPE_AUDIO,                   \
     .id             = id_,                                  \
     .priv_data_size = sizeof(ADPCMEncodeContext),           \
@@ -716,6 +715,7 @@ AVCodec ff_ ## name_ ## _encoder = {                        \
     .encode2        = adpcm_encode_frame,                   \
     .close          = adpcm_encode_close,                   \
     .sample_fmts    = sample_fmts_,                         \
+    .long_name      = NULL_IF_CONFIG_SMALL(long_name_),     \
 }
 
 ADPCM_ENCODER(AV_CODEC_ID_ADPCM_IMA_QT,  adpcm_ima_qt,  sample_fmts_p, "ADPCM IMA QuickTime");

libavcodec/adx.c

@@ -1,20 +1,20 @@
 /*
  * Copyright (c) 2011  Justin Ruggles
  *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 

libavcodec/adx_parser.c

@@ -1,20 +1,20 @@
 /*
  * Copyright (c) 2011  Justin Ruggles
  *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 

libavcodec/adxdec.c

@@ -174,7 +174,6 @@ static void adx_decode_flush(AVCodecContext *avctx)
 
 AVCodec ff_adpcm_adx_decoder = {
     .name           = "adpcm_adx",
-    .long_name      = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = AV_CODEC_ID_ADPCM_ADX,
     .priv_data_size = sizeof(ADXContext),
@@ -182,6 +181,7 @@ AVCodec ff_adpcm_adx_decoder = {
     .decode         = adx_decode_frame,
     .flush          = adx_decode_flush,
     .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
                                                       AV_SAMPLE_FMT_NONE },
 };

libavcodec/adxenc.c

@@ -158,7 +158,6 @@ static int adx_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
 
 AVCodec ff_adpcm_adx_encoder = {
     .name           = "adpcm_adx",
-    .long_name      = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = AV_CODEC_ID_ADPCM_ADX,
     .priv_data_size = sizeof(ADXContext),
@@ -166,4 +165,5 @@ AVCodec ff_adpcm_adx_encoder = {
     .encode2        = adx_encode_frame,
     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
                                                       AV_SAMPLE_FMT_NONE },
+    .long_name      = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
 };

libavcodec/aic.c

@@ -470,7 +470,6 @@ static av_cold int aic_decode_close(AVCodecContext *avctx)
 
 AVCodec ff_aic_decoder = {
     .name           = "aic",
-    .long_name      = NULL_IF_CONFIG_SMALL("Apple Intermediate Codec"),
     .type           = AVMEDIA_TYPE_VIDEO,
     .id             = AV_CODEC_ID_AIC,
     .priv_data_size = sizeof(AICContext),
@@ -478,4 +477,5 @@ AVCodec ff_aic_decoder = {
     .close          = aic_decode_close,
     .decode         = aic_decode_frame,
     .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("Apple Intermediate Codec")
 };

libavcodec/alac.c

@@ -320,9 +320,6 @@ static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index,
             rice_history_mult[ch] = get_bits(&alac->gb, 3);
             lpc_order[ch]         = get_bits(&alac->gb, 5);
 
-            if (lpc_order[ch] >= alac->max_samples_per_frame)
-                return AVERROR_INVALIDDATA;
-
             /* read the predictor table */
             for (i = lpc_order[ch] - 1; i >= 0; i--)
                 lpc_coefs[ch][i] = get_sbits(&alac->gb, 16);
@@ -466,8 +463,9 @@ static int alac_decode_frame(AVCodecContext *avctx, void *data,
         }
 
         channels = (element == TYPE_CPE) ? 2 : 1;
-        if (ch + channels > alac->channels ||
-            ff_alac_channel_layout_offsets[alac->channels - 1][ch] + channels > alac->channels) {
+        if (   ch + channels > alac->channels
+            || ff_alac_channel_layout_offsets[alac->channels - 1][ch] + channels > alac->channels
+        ) {
             av_log(avctx, AV_LOG_ERROR, "invalid element channel count\n");
             return AVERROR_INVALIDDATA;
         }
@@ -490,8 +488,7 @@ static int alac_decode_frame(AVCodecContext *avctx, void *data,
                avpkt->size * 8 - get_bits_count(&alac->gb));
     }
 
-    if (alac->channels == ch)
-        *got_frame_ptr = 1;
+    *got_frame_ptr = 1;
 
     return avpkt->size;
 }
@@ -514,7 +511,11 @@ static av_cold int alac_decode_close(AVCodecContext *avctx)
 static int allocate_buffers(ALACContext *alac)
 {
     int ch;
-    int buf_size = alac->max_samples_per_frame * sizeof(int32_t);
+    int buf_size;
+
+    if (alac->max_samples_per_frame > INT_MAX / sizeof(int32_t))
+        goto buf_alloc_fail;
+    buf_size = alac->max_samples_per_frame * sizeof(int32_t);
 
     for (ch = 0; ch < FFMIN(alac->channels, 2); ch++) {
         FF_ALLOC_OR_GOTO(alac->avctx, alac->predict_error_buffer[ch],
@@ -545,8 +546,7 @@ static int alac_set_info(ALACContext *alac)
     bytestream2_skipu(&gb, 12); // size:4, alac:4, version:4
 
     alac->max_samples_per_frame = bytestream2_get_be32u(&gb);
-    if (!alac->max_samples_per_frame ||
-        alac->max_samples_per_frame > INT_MAX / sizeof(int32_t)) {
+    if (!alac->max_samples_per_frame || alac->max_samples_per_frame > INT_MAX) {
         av_log(alac->avctx, AV_LOG_ERROR, "max samples per frame invalid: %u\n",
                alac->max_samples_per_frame);
         return AVERROR_INVALIDDATA;
@@ -627,7 +627,6 @@ static int init_thread_copy(AVCodecContext *avctx)
 
 AVCodec ff_alac_decoder = {
     .name           = "alac",
-    .long_name      = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = AV_CODEC_ID_ALAC,
     .priv_data_size = sizeof(ALACContext),
@@ -636,4 +635,5 @@ AVCodec ff_alac_decoder = {
     .decode         = alac_decode_frame,
     .init_thread_copy = ONLY_IF_THREADS_ENABLED(init_thread_copy),
     .capabilities   = CODEC_CAP_DR1 | CODEC_CAP_FRAME_THREADS,
+    .long_name      = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
 };