update for 0.9.4

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

.gitignore

@@ -15,7 +15,6 @@ config.*
 doc/*.1
 doc/*.html
 doc/*.pod
-doc/fate.txt
 doxy
 ffmpeg
 ffplay
@@ -44,10 +43,8 @@ tests/tiny_psnr
 tests/videogen
 tests/vsynth1
 tests/vsynth2
-tools/aviocat
 tools/cws2fws
 tools/graph2dot
-tools/ismindex
 tools/lavfi-showfiltfmts
 tools/pktdumper
 tools/probetest

Changelog

@@ -1,48 +1,11 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.
 
-version next:
-
-version 0.10:
-- Fixes: CVE-2011-3929, CVE-2011-3934, CVE-2011-3935, CVE-2011-3936,
-         CVE-2011-3937, CVE-2011-3940, CVE-2011-3941, CVE-2011-3944,
-         CVE-2011-3945, CVE-2011-3946, CVE-2011-3947, CVE-2011-3949,
-         CVE-2011-3950, CVE-2011-3951, CVE-2011-3952
-- v410 Quicktime Uncompressed 4:4:4 10-bit encoder and decoder
-- SBaGen (SBG) binaural beats script demuxer
-- OpenMG Audio muxer
-- Timecode extraction in DV and MOV
-- thumbnail video filter
-- XML output in ffprobe
-- asplit audio filter
-- tinterlace video filter
-- astreamsync audio filter
-- amerge audio filter
-- ISMV (Smooth Streaming) muxer
-- GSM audio parser
-- SMJPEG muxer
-- XWD encoder and decoder
-- Automatic thread count based on detection number of (available) CPU cores
-- y41p Brooktree Uncompressed 4:1:1 12-bit encoder and decoder
-- ffprobe -show_error option
-- Avid 1:1 10-bit RGB Packer codec
-- v308 Quicktime Uncompressed 4:4:4 encoder and decoder
-- yuv4 libquicktime packed 4:2:0 encoder and decoder
-- ffprobe -show_frames option
-- silencedetect audio filter
-- ffprobe -show_program_version, -show_library_versions, -show_versions options
-- rv34: frame-level multi-threading
-- optimized iMDCT transform on x86 using SSE for for mpegaudiodec
-- Improved PGS subtitle decoder
-- dumpgraph option to lavfi device
-- r210 and r10k encoders
-- ffwavesynth decoder
-- aviocat tool
-- ffeval tool
-
-
 version 0.9:
 
+- Indeo 4 decoder
+- SMJPEG demuxer
+- y41p Brooktree Uncompressed 4:1:1 12-bit encoder and decoder
 - openal input device added
 - boxblur filter added
 - BWF muxer
@@ -165,7 +128,7 @@ easier to use. The changes are:
 - pan audio filter
 - IFF Amiga Continuous Bitmap (ACBM) decoder
 - ass filter
-- CRI ADX audio format muxer and demuxer
+- CRI ADX audio format demuxer
 - Playstation Portable PMP format demuxer
 - Microsoft Windows ICO demuxer
 - life source
@@ -174,13 +137,11 @@ easier to use. The changes are:
 - new option: -report
 - Dxtory capture format decoder
 - cellauto source
-- Simple segmenting muxer
-- Indeo 4 decoder
-- SMJPEG demuxer
-
+- v410 Quicktime Uncompressed 4:4:4 10-bit encoder and decoder
 
 version 0.8:
 
+
 - many many things we forgot because we rather write code than changelogs
 - WebM support in Matroska de/muxer
 - low overhead Ogg muxing

Doxyfile

@@ -31,13 +31,7 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.
 
-PROJECT_NUMBER         = 0.10
-
-# 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.
-PROJECT_LOGO           =
+PROJECT_NUMBER         = 0.9.4
 
 # The OUTPUT_DIRECTORY tag is used to specify the (relative or absolute)
 # base path where the generated documentation will be put.
@@ -631,7 +625,8 @@ EXCLUDE_SYMLINKS       = NO
 # for example use the pattern */test/*
 
 EXCLUDE_PATTERNS       = *.git \
-                         *.d
+                         *.d \
+                         avconv.c
 
 # The EXCLUDE_SYMBOLS tag can be used to specify one or more symbol names
 # (namespaces, classes, functions, etc.) that should be excluded from the
@@ -766,7 +761,7 @@ ALPHABETICAL_INDEX     = YES
 # the COLS_IN_ALPHA_INDEX tag can be used to specify the number of columns
 # in which this list will be split (can be a number in the range [1..20])
 
-COLS_IN_ALPHA_INDEX    = 2
+COLS_IN_ALPHA_INDEX    = 5
 
 # In case all classes in a project start with a common prefix, all
 # classes will be put under the same header in the alphabetical index.
@@ -800,13 +795,13 @@ HTML_FILE_EXTENSION    = .html
 # each generated HTML page. If it is left blank doxygen will generate a
 # standard header.
 
-HTML_HEADER            = doc/doxy/header.html
+HTML_HEADER            =
 
 # The HTML_FOOTER tag can be used to specify a personal HTML footer for
 # each generated HTML page. If it is left blank doxygen will generate a
 # standard footer.
 
-HTML_FOOTER            = doc/doxy/footer.html
+HTML_FOOTER            =
 
 # The HTML_STYLESHEET tag can be used to specify a user-defined cascading
 # style sheet that is used by each HTML page. It can be used to
@@ -815,7 +810,7 @@ HTML_FOOTER            = doc/doxy/footer.html
 # the style sheet file to the HTML output directory, so don't put your own
 # stylesheet in the HTML output directory as well, or it will be erased!
 
-HTML_STYLESHEET        = doc/doxy/doxy_stylesheet.css
+HTML_STYLESHEET        =
 
 # The HTML_COLORSTYLE_HUE tag controls the color of the HTML output.
 # Doxygen will adjust the colors in the stylesheet and background images

MAINTAINERS

@@ -87,8 +87,6 @@ Generic Parts:
     bitstream.c, bitstream.h            Michael Niedermayer
   CABAC:
     cabac.h, cabac.c                    Michael Niedermayer
-  codec names:
-    codec_names.sh                      Nicolas George
   DSP utilities:
     dsputils.c, dsputils.h              Michael Niedermayer
   entropy coding:
@@ -142,7 +140,6 @@ Codecs:
   dv.c                                  Roman Shaposhnik
   eacmv*, eaidct*, eat*                 Peter Ross
   ffv1.c                                Michael Niedermayer
-  ffwavesynth.c                         Nicolas George
   flac*                                 Justin Ruggles
   flashsv*                              Benjamin Larsson
   flicvideo.c                           Mike Melanson
@@ -163,7 +160,6 @@ Codecs:
   jvdec.c                               Peter Ross
   kmvc.c                                Kostya Shishkov
   lcl*.c                                Roberto Togni, Reimar Doeffinger
-  libcelt_dec.c                         Nicolas George
   libgsm.c                              Michel Bardiaux
   libdirac*                             David Conrad
   libopenjpeg.c                         Jaikrishnan Menon
@@ -269,10 +265,6 @@ libavfilter
 ===========
 
 Video filters:
-  graphdump.c                           Nicolas George
-  af_amerge.c                           Nicolas George
-  af_astreamsync.c                      Nicolas George
-  af_pan.c                              Nicolas George
   vsrc_mandelbrot.c                     Michael Niedermayer
   vf_yadif.c                            Michael Niedermayer
 
@@ -333,7 +325,6 @@ Muxers/Demuxers:
   msnwc_tcp.c                           Ramiro Polla
   mtv.c                                 Reynaldo H. Verdejo Pinochet
   mxf*                                  Baptiste Coudurier
-  mxfdec.c                              Tomas Härdin
   nsvdec.c                              Francois Revol
   nut.c                                 Michael Niedermayer
   nuv.c                                 Reimar Doeffinger
@@ -353,7 +344,6 @@ Muxers/Demuxers:
   rtpdec_asf.*                          Ronald S. Bultje
   rtpenc_mpv.*, rtpenc_aac.*            Martin Storsjo
   rtsp.c                                Luca Barbato
-  sbgdec.c                              Nicolas George
   sdp.c                                 Martin Storsjo
   segafilm.c                            Mike Melanson
   siff.c                                Kostya Shishkov
@@ -395,6 +385,10 @@ x86                                     Michael Niedermayer
 Releases
 ========
 
+0.5                                     *Deprecated/Unmaintained*
+0.6                                     *Deprecated/Unmaintained*
+0.7                                     Michael Niedermayer
+0.8                                     Michael Niedermayer
 0.9                                     Michael Niedermayer
 
 
@@ -417,7 +411,6 @@ Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
 Lou Logan                     7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
 Luca Barbato                  6677 4209 213C 8843 5B67 29E7 E84C 78C2 84E9 0E34
 Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
-Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
 Panagiotis Issaris            6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
 Peter Ross                    A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
 Reimar Döffinger              C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7

Makefile

@@ -8,9 +8,9 @@ vpath %.S    $(SRC_PATH)
 vpath %.asm  $(SRC_PATH)
 vpath %.v    $(SRC_PATH)
 vpath %.texi $(SRC_PATH)
-vpath %/fate_config.sh.template $(SRC_PATH)
 
 PROGS-$(CONFIG_FFMPEG)   += ffmpeg
+PROGS-$(CONFIG_AVCONV)   += avconv
 PROGS-$(CONFIG_FFPLAY)   += ffplay
 PROGS-$(CONFIG_FFPROBE)  += ffprobe
 PROGS-$(CONFIG_FFSERVER) += ffserver
@@ -23,7 +23,7 @@ HOSTPROGS  := $(TESTTOOLS:%=tests/%)
 TOOLS       = qt-faststart trasher
 TOOLS-$(CONFIG_ZLIB) += cws2fws
 
-BASENAMES   = ffmpeg ffplay ffprobe ffserver
+BASENAMES   = ffmpeg avconv ffplay ffprobe ffserver
 ALLPROGS    = $(BASENAMES:%=%$(PROGSSUF)$(EXESUF))
 ALLPROGS_G  = $(BASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
 ALLMANPAGES = $(BASENAMES:%=%.1)
@@ -38,7 +38,7 @@ FFLIBS-$(CONFIG_SWSCALE)  += swscale
 
 FFLIBS := avutil
 
-DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
+DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset)
 
 SKIPHEADERS = cmdutils_common_opts.h
 
@@ -77,8 +77,6 @@ define DOSUBDIR
 $(foreach V,$(SUBDIR_VARS),$(eval $(call RESET,$(V))))
 SUBDIR := $(1)/
 include $(SRC_PATH)/$(1)/Makefile
--include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
-include $(SRC_PATH)/library.mak
 endef
 
 $(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))

RELEASE

@@ -1 +1 @@
-0.10
+0.9.4

VERSION

@@ -1 +1 @@
-0.10
+0.9.4

cmdutils.c

@@ -33,10 +33,7 @@
 #include "libavfilter/avfilter.h"
 #include "libavdevice/avdevice.h"
 #include "libswscale/swscale.h"
-#include "libswresample/swresample.h"
-#if CONFIG_POSTPROC
 #include "libpostproc/postprocess.h"
-#endif
 #include "libavutil/avstring.h"
 #include "libavutil/mathematics.h"
 #include "libavutil/parseutils.h"
@@ -56,15 +53,14 @@
 struct SwsContext *sws_opts;
 AVDictionary *format_opts, *codec_opts;
 
-const int this_year = 2012;
+static const int this_year = 2014;
 
 static FILE *report_file;
 
 void init_opts(void)
 {
 #if CONFIG_SWSCALE
-    sws_opts = sws_getContext(16, 16, 0, 16, 16, 0, SWS_BICUBIC,
-                              NULL, NULL, NULL);
+    sws_opts = sws_getContext(16, 16, 0, 16, 16, 0, SWS_BICUBIC, NULL, NULL, NULL);
 #endif
 }
 
@@ -78,7 +74,7 @@ void uninit_opts(void)
     av_dict_free(&codec_opts);
 }
 
-void log_callback_help(void *ptr, int level, const char *fmt, va_list vl)
+void log_callback_help(void* ptr, int level, const char* fmt, va_list vl)
 {
     vfprintf(stdout, fmt, vl);
 }
@@ -97,20 +93,19 @@ static void log_callback_report(void *ptr, int level, const char *fmt, va_list v
     fflush(report_file);
 }
 
-double parse_number_or_die(const char *context, const char *numstr, int type,
-                           double min, double max)
+double parse_number_or_die(const char *context, const char *numstr, int type, double min, double max)
 {
     char *tail;
     const char *error;
     double d = av_strtod(numstr, &tail);
     if (*tail)
-        error = "Expected number for %s but found: %s\n";
+        error= "Expected number for %s but found: %s\n";
     else if (d < min || d > max)
-        error = "The value for %s was %s which is not within %f - %f\n";
-    else if (type == OPT_INT64 && (int64_t)d != d)
-        error = "Expected int64 for %s but found %s\n";
+        error= "The value for %s was %s which is not within %f - %f\n";
+    else if(type == OPT_INT64 && (int64_t)d != d)
+        error= "Expected int64 for %s but found %s\n";
     else if (type == OPT_INT && (int)d != d)
-        error = "Expected int for %s but found %s\n";
+        error= "Expected int for %s but found %s\n";
     else
         return d;
     av_log(NULL, AV_LOG_FATAL, error, context, numstr, min, max);
@@ -118,8 +113,7 @@ double parse_number_or_die(const char *context, const char *numstr, int type,
     return 0;
 }
 
-int64_t parse_time_or_die(const char *context, const char *timestr,
-                          int is_duration)
+int64_t parse_time_or_die(const char *context, const char *timestr, int is_duration)
 {
     int64_t us;
     if (av_parse_time(&us, timestr, is_duration) < 0) {
@@ -130,14 +124,13 @@ int64_t parse_time_or_die(const char *context, const char *timestr,
     return us;
 }
 
-void show_help_options(const OptionDef *options, const char *msg, int mask,
-                       int value)
+void show_help_options(const OptionDef *options, const char *msg, int mask, int value)
 {
     const OptionDef *po;
     int first;
 
     first = 1;
-    for (po = options; po->name != NULL; po++) {
+    for(po = options; po->name != NULL; po++) {
         char buf[64];
         if ((po->flags & mask) == value) {
             if (first) {
@@ -164,8 +157,7 @@ void show_help_children(const AVClass *class, int flags)
         show_help_children(child, flags);
 }
 
-static const OptionDef *find_option(const OptionDef *po, const char *name)
-{
+static const OptionDef* find_option(const OptionDef *po, const char *name){
     const char *p = strchr(name, ':');
     int len = p ? p - name : strlen(name);
 
@@ -212,8 +204,8 @@ static void prepare_app_arguments(int *argc_ptr, char ***argv_ptr)
         buffsize += WideCharToMultiByte(CP_UTF8, 0, argv_w[i], -1,
                                         NULL, 0, NULL, NULL);
 
-    win32_argv_utf8 = av_mallocz(sizeof(char *) * (win32_argc + 1) + buffsize);
-    argstr_flat     = (char *)win32_argv_utf8 + sizeof(char *) * (win32_argc + 1);
+    win32_argv_utf8 = av_mallocz(sizeof(char*) * (win32_argc + 1) + buffsize);
+    argstr_flat     = (char*)win32_argv_utf8 + sizeof(char*) * (win32_argc + 1);
     if (win32_argv_utf8 == NULL) {
         LocalFree(argv_w);
         return;
@@ -238,8 +230,8 @@ static inline void prepare_app_arguments(int *argc_ptr, char ***argv_ptr)
 }
 #endif /* WIN32 && !__MINGW32CE__ */
 
-int parse_option(void *optctx, const char *opt, const char *arg,
-                 const OptionDef *options)
+
+int parse_option(void *optctx, const char *opt, const char *arg, const OptionDef *options)
 {
     const OptionDef *po;
     int bool_val = 1;
@@ -268,14 +260,13 @@ unknown_opt:
 
     /* new-style options contain an offset into optctx, old-style address of
      * a global var*/
-    dst = po->flags & (OPT_OFFSET | OPT_SPEC) ? (uint8_t *)optctx + po->u.off
-                                              : po->u.dst_ptr;
+    dst = po->flags & (OPT_OFFSET|OPT_SPEC) ? (uint8_t*)optctx + po->u.off : po->u.dst_ptr;
 
     if (po->flags & OPT_SPEC) {
         SpecifierOpt **so = dst;
         char *p = strchr(opt, ':');
 
-        dstcount = (int *)(so + 1);
+        dstcount = (int*)(so + 1);
         *so = grow_array(*so, sizeof(**so), dstcount, *dstcount + 1);
         (*so)[*dstcount - 1].specifier = av_strdup(p ? p + 1 : "");
         dst = &(*so)[*dstcount - 1].u;
@@ -284,25 +275,24 @@ unknown_opt:
     if (po->flags & OPT_STRING) {
         char *str;
         str = av_strdup(arg);
-        *(char **)dst = str;
+        *(char**)dst = str;
     } else if (po->flags & OPT_BOOL) {
-        *(int *)dst = bool_val;
+        *(int*)dst = bool_val;
     } else if (po->flags & OPT_INT) {
-        *(int *)dst = parse_number_or_die(opt, arg, OPT_INT64, INT_MIN, INT_MAX);
+        *(int*)dst = parse_number_or_die(opt, arg, OPT_INT64, INT_MIN, INT_MAX);
     } else if (po->flags & OPT_INT64) {
-        *(int64_t *)dst = parse_number_or_die(opt, arg, OPT_INT64, INT64_MIN, INT64_MAX);
+        *(int64_t*)dst = parse_number_or_die(opt, arg, OPT_INT64, INT64_MIN, INT64_MAX);
     } else if (po->flags & OPT_TIME) {
-        *(int64_t *)dst = parse_time_or_die(opt, arg, 1);
+        *(int64_t*)dst = parse_time_or_die(opt, arg, 1);
     } else if (po->flags & OPT_FLOAT) {
-        *(float *)dst = parse_number_or_die(opt, arg, OPT_FLOAT, -INFINITY, INFINITY);
+        *(float*)dst = parse_number_or_die(opt, arg, OPT_FLOAT, -INFINITY, INFINITY);
     } else if (po->flags & OPT_DOUBLE) {
-        *(double *)dst = parse_number_or_die(opt, arg, OPT_DOUBLE, -INFINITY, INFINITY);
+        *(double*)dst = parse_number_or_die(opt, arg, OPT_DOUBLE, -INFINITY, INFINITY);
     } else if (po->u.func_arg) {
-        int ret = po->flags & OPT_FUNC2 ? po->u.func2_arg(optctx, opt, arg)
-                                        : po->u.func_arg(opt, arg);
+        int ret = po->flags & OPT_FUNC2 ? po->u.func2_arg(optctx, opt, arg) :
+                                          po->u.func_arg(opt, arg);
         if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR,
-                   "Failed to set value '%s' for option '%s'\n", arg, opt);
+            av_log(NULL, AV_LOG_ERROR, "Failed to set value '%s' for option '%s'\n", arg, opt);
             return ret;
         }
     }
@@ -312,7 +302,7 @@ unknown_opt:
 }
 
 void parse_options(void *optctx, int argc, char **argv, const OptionDef *options,
-                   void (*parse_arg_function)(void *, const char*))
+                   void (* parse_arg_function)(void *, const char*))
 {
     const char *opt;
     int optindex, handleoptions = 1, ret;
@@ -345,8 +335,7 @@ void parse_options(void *optctx, int argc, char **argv, const OptionDef *options
 /*
  * Return index of option opt in argv or 0 if not found.
  */
-static int locate_option(int argc, char **argv, const OptionDef *options,
-                         const char *optname)
+static int locate_option(int argc, char **argv, const OptionDef *options, const char *optname)
 {
     const OptionDef *po;
     int i;
@@ -429,18 +418,15 @@ int opt_default(const char *opt, const char *arg)
         p = opt + strlen(opt);
     av_strlcpy(opt_stripped, opt, FFMIN(sizeof(opt_stripped), p - opt + 1));
 
-    if ((oc = 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') &&
-         (oc = av_opt_find(&cc, opt + 1, NULL, 0, AV_OPT_SEARCH_FAKE_OBJ))))
+    if ((oc = 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') &&
+          (oc = av_opt_find(&cc, opt+1, NULL, 0, AV_OPT_SEARCH_FAKE_OBJ))))
         av_dict_set(&codec_opts, opt, arg, FLAGS(oc));
-    if ((of = av_opt_find(&fc, opt, NULL, 0,
-                          AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ)))
+    if ((of = av_opt_find(&fc, opt, NULL, 0, AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ)))
         av_dict_set(&format_opts, opt, arg, FLAGS(of));
 #if CONFIG_SWSCALE
     sc = sws_get_class();
-    if ((os = av_opt_find(&sc, opt, NULL, 0,
-                          AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
+    if ((os = 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);
         if (ret < 0) {
@@ -523,20 +509,6 @@ int opt_report(const char *opt)
     return 0;
 }
 
-int opt_max_alloc(const char *opt, const char *arg)
-{
-    char *tail;
-    size_t max;
-
-    max = strtol(arg, &tail, 10);
-    if (*tail) {
-        av_log(NULL, AV_LOG_FATAL, "Invalid max_alloc \"%s\".\n", arg);
-        exit_program(1);
-    }
-    av_max_alloc(max);
-    return 0;
-}
-
 int opt_codec_debug(const char *opt, const char *arg)
 {
     av_log_set_level(AV_LOG_DEBUG);
@@ -571,14 +543,13 @@ static int warned_cfg = 0;
 #define INDENT        1
 #define SHOW_VERSION  2
 #define SHOW_CONFIG   4
-#define SHOW_COPYRIGHT 8
 
 #define PRINT_LIB_INFO(libname, LIBNAME, flags, level)                  \
     if (CONFIG_##LIBNAME) {                                             \
         const char *indent = flags & INDENT? "  " : "";                 \
         if (flags & SHOW_VERSION) {                                     \
             unsigned int version = libname##_version();                 \
-            av_log(NULL, level, "%slib%-11s %2d.%3d.%3d / %2d.%3d.%3d\n",\
+            av_log(NULL, level, "%slib%-9s %2d.%3d.%2d / %2d.%3d.%2d\n",\
                    indent, #libname,                                    \
                    LIB##LIBNAME##_VERSION_MAJOR,                        \
                    LIB##LIBNAME##_VERSION_MINOR,                        \
@@ -608,24 +579,7 @@ static void print_all_libs_info(int flags, int level)
     PRINT_LIB_INFO(avdevice, AVDEVICE, flags, level);
     PRINT_LIB_INFO(avfilter, AVFILTER, flags, level);
     PRINT_LIB_INFO(swscale,  SWSCALE,  flags, level);
-    PRINT_LIB_INFO(swresample,SWRESAMPLE,  flags, level);
-#if CONFIG_POSTPROC
     PRINT_LIB_INFO(postproc, POSTPROC, flags, level);
-#endif
-}
-
-static void print_program_info(int flags, int level)
-{
-    const char *indent = flags & INDENT? "  " : "";
-
-    av_log(NULL, level, "%s version " FFMPEG_VERSION, program_name);
-    if (flags & SHOW_COPYRIGHT)
-        av_log(NULL, level, " Copyright (c) %d-%d the FFmpeg developers",
-               program_birth_year, this_year);
-    av_log(NULL, level, "\n");
-    av_log(NULL, level, "%sbuilt on %s %s with %s %s\n",
-           indent, __DATE__, __TIME__, CC_TYPE, CC_VERSION);
-    av_log(NULL, level, "%sconfiguration: " FFMPEG_CONFIGURATION "\n", indent);
 }
 
 void show_banner(int argc, char **argv, const OptionDef *options)
@@ -634,14 +588,18 @@ void show_banner(int argc, char **argv, const OptionDef *options)
     if (idx)
         return;
 
-    print_program_info (INDENT|SHOW_COPYRIGHT, AV_LOG_INFO);
+    av_log(NULL, AV_LOG_INFO, "%s version " FFMPEG_VERSION ", Copyright (c) %d-%d the FFmpeg developers\n",
+           program_name, program_birth_year, this_year);
+    av_log(NULL, AV_LOG_INFO, "  built on %s %s with %s %s\n",
+           __DATE__, __TIME__, CC_TYPE, CC_VERSION);
+    av_log(NULL, AV_LOG_INFO, "  configuration: " FFMPEG_CONFIGURATION "\n");
     print_all_libs_info(INDENT|SHOW_CONFIG,  AV_LOG_INFO);
     print_all_libs_info(INDENT|SHOW_VERSION, AV_LOG_INFO);
 }
 
 int opt_version(const char *opt, const char *arg) {
     av_log_set_callback(log_callback_help);
-    print_program_info (0           , AV_LOG_INFO);
+    printf("%s " FFMPEG_VERSION "\n", program_name);
     print_all_libs_info(SHOW_VERSION, AV_LOG_INFO);
     return 0;
 }
@@ -718,133 +676,137 @@ int opt_license(const char *opt, const char *arg)
 
 int opt_formats(const char *opt, const char *arg)
 {
-    AVInputFormat *ifmt  = NULL;
-    AVOutputFormat *ofmt = NULL;
+    AVInputFormat *ifmt=NULL;
+    AVOutputFormat *ofmt=NULL;
     const char *last_name;
 
-    printf("File formats:\n"
-           " D. = Demuxing supported\n"
-           " .E = Muxing supported\n"
-           " --\n");
-    last_name = "000";
-    for (;;) {
-        int decode = 0;
-        int encode = 0;
-        const char *name      = NULL;
-        const char *long_name = NULL;
+    printf(
+        "File formats:\n"
+        " D. = Demuxing supported\n"
+        " .E = Muxing supported\n"
+        " --\n");
+    last_name= "000";
+    for(;;){
+        int decode=0;
+        int encode=0;
+        const char *name=NULL;
+        const char *long_name=NULL;
 
-        while ((ofmt = av_oformat_next(ofmt))) {
-            if ((name == NULL || strcmp(ofmt->name, name) < 0) &&
-                strcmp(ofmt->name, last_name) > 0) {
-                name      = ofmt->name;
-                long_name = ofmt->long_name;
-                encode    = 1;
+        while((ofmt= av_oformat_next(ofmt))) {
+            if((name == NULL || strcmp(ofmt->name, name)<0) &&
+                strcmp(ofmt->name, last_name)>0){
+                name= ofmt->name;
+                long_name= ofmt->long_name;
+                encode=1;
             }
         }
-        while ((ifmt = av_iformat_next(ifmt))) {
-            if ((name == NULL || strcmp(ifmt->name, name) < 0) &&
-                strcmp(ifmt->name, last_name) > 0) {
-                name      = ifmt->name;
-                long_name = ifmt->long_name;
-                encode    = 0;
+        while((ifmt= av_iformat_next(ifmt))) {
+            if((name == NULL || strcmp(ifmt->name, name)<0) &&
+                strcmp(ifmt->name, last_name)>0){
+                name= ifmt->name;
+                long_name= ifmt->long_name;
+                encode=0;
             }
-            if (name && strcmp(ifmt->name, name) == 0)
-                decode = 1;
+            if(name && strcmp(ifmt->name, name)==0)
+                decode=1;
         }
-        if (name == NULL)
+        if(name==NULL)
             break;
-        last_name = name;
+        last_name= name;
 
-        printf(" %s%s %-15s %s\n",
-               decode ? "D" : " ",
-               encode ? "E" : " ",
-               name,
+        printf(
+            " %s%s %-15s %s\n",
+            decode ? "D":" ",
+            encode ? "E":" ",
+            name,
             long_name ? long_name:" ");
     }
     return 0;
 }
 
-static char get_media_type_char(enum AVMediaType type)
-{
-    static const char map[AVMEDIA_TYPE_NB] = {
-        [AVMEDIA_TYPE_VIDEO]      = 'V',
-        [AVMEDIA_TYPE_AUDIO]      = 'A',
-        [AVMEDIA_TYPE_DATA]       = 'D',
-        [AVMEDIA_TYPE_SUBTITLE]   = 'S',
-        [AVMEDIA_TYPE_ATTACHMENT] = 'T',
-    };
-    return type >= 0 && type < AVMEDIA_TYPE_NB && map[type] ? map[type] : '?';
-}
-
 int opt_codecs(const char *opt, const char *arg)
 {
-    AVCodec *p = NULL, *p2;
+    AVCodec *p=NULL, *p2;
     const char *last_name;
-    printf("Codecs:\n"
-           " D..... = Decoding supported\n"
-           " .E.... = Encoding supported\n"
-           " ..V... = Video codec\n"
-           " ..A... = Audio codec\n"
-           " ..S... = Subtitle codec\n"
-           " ...S.. = Supports draw_horiz_band\n"
-           " ....D. = Supports direct rendering method 1\n"
-           " .....T = Supports weird frame truncation\n"
-           " ------\n");
+    printf(
+        "Codecs:\n"
+        " D..... = Decoding supported\n"
+        " .E.... = Encoding supported\n"
+        " ..V... = Video codec\n"
+        " ..A... = Audio codec\n"
+        " ..S... = Subtitle codec\n"
+        " ...S.. = Supports draw_horiz_band\n"
+        " ....D. = Supports direct rendering method 1\n"
+        " .....T = Supports weird frame truncation\n"
+        " ------\n");
     last_name= "000";
-    for (;;) {
-        int decode = 0;
-        int encode = 0;
-        int cap    = 0;
+    for(;;){
+        int decode=0;
+        int encode=0;
+        int cap=0;
+        const char *type_str;
 
-        p2 = NULL;
-        while ((p = av_codec_next(p))) {
-            if ((p2 == NULL || strcmp(p->name, p2->name) < 0) &&
-                strcmp(p->name, last_name) > 0) {
-                p2 = p;
-                decode = encode = cap = 0;
+        p2=NULL;
+        while((p= av_codec_next(p))) {
+            if((p2==NULL || strcmp(p->name, p2->name)<0) &&
+                strcmp(p->name, last_name)>0){
+                p2= p;
+                decode= encode= cap=0;
             }
-            if (p2 && strcmp(p->name, p2->name) == 0) {
-                if (p->decode)
-                    decode = 1;
-                if (p->encode)
-                    encode = 1;
+            if(p2 && strcmp(p->name, p2->name)==0){
+                if(p->decode) decode=1;
+                if(p->encode) encode=1;
                 cap |= p->capabilities;
             }
         }
-        if (p2 == NULL)
+        if(p2==NULL)
             break;
-        last_name = p2->name;
+        last_name= p2->name;
 
-        printf(" %s%s%c%s%s%s %-15s %s",
-               decode ? "D" : (/* p2->decoder ? "d" : */ " "),
-               encode ? "E" : " ",
-               get_media_type_char(p2->type),
-               cap & CODEC_CAP_DRAW_HORIZ_BAND ? "S" : " ",
-               cap & CODEC_CAP_DR1 ? "D" : " ",
-               cap & CODEC_CAP_TRUNCATED ? "T" : " ",
-               p2->name,
-               p2->long_name ? p2->long_name : "");
-#if 0
-            if (p2->decoder && decode == 0)
-                printf(" use %s for decoding", p2->decoder->name);
-#endif
+        switch(p2->type) {
+        case AVMEDIA_TYPE_VIDEO:
+            type_str = "V";
+            break;
+        case AVMEDIA_TYPE_AUDIO:
+            type_str = "A";
+            break;
+        case AVMEDIA_TYPE_SUBTITLE:
+            type_str = "S";
+            break;
+        default:
+            type_str = "?";
+            break;
+        }
+        printf(
+            " %s%s%s%s%s%s %-15s %s",
+            decode ? "D": (/*p2->decoder ? "d":*/" "),
+            encode ? "E":" ",
+            type_str,
+            cap & CODEC_CAP_DRAW_HORIZ_BAND ? "S":" ",
+            cap & CODEC_CAP_DR1 ? "D":" ",
+            cap & CODEC_CAP_TRUNCATED ? "T":" ",
+            p2->name,
+            p2->long_name ? p2->long_name : "");
+       /* if(p2->decoder && decode==0)
+            printf(" use %s for decoding", p2->decoder->name);*/
         printf("\n");
     }
     printf("\n");
-    printf("Note, the names of encoders and decoders do not always match, so there are\n"
-           "several cases where the above table shows encoder only or decoder only entries\n"
-           "even though both encoding and decoding are supported. For example, the h263\n"
-           "decoder corresponds to the h263 and h263p encoders, for file formats it is even\n"
-           "worse.\n");
+    printf(
+"Note, the names of encoders and decoders do not always match, so there are\n"
+"several cases where the above table shows encoder only or decoder only entries\n"
+"even though both encoding and decoding are supported. For example, the h263\n"
+"decoder corresponds to the h263 and h263p encoders, for file formats it is even\n"
+"worse.\n");
     return 0;
 }
 
 int opt_bsfs(const char *opt, const char *arg)
 {
-    AVBitStreamFilter *bsf = NULL;
+    AVBitStreamFilter *bsf=NULL;
 
     printf("Bitstream filters:\n");
-    while ((bsf = av_bitstream_filter_next(bsf)))
+    while((bsf = av_bitstream_filter_next(bsf)))
         printf("%s\n", bsf->name);
     printf("\n");
     return 0;
@@ -872,31 +834,11 @@ int opt_protocols(const char *opt, const char *arg)
 int opt_filters(const char *opt, const char *arg)
 {
     AVFilter av_unused(**filter) = NULL;
-    char descr[64], *descr_cur;
-    int i, j;
-    const AVFilterPad *pad;
 
     printf("Filters:\n");
 #if CONFIG_AVFILTER
-    while ((filter = av_filter_next(filter)) && *filter) {
-        descr_cur = descr;
-        for (i = 0; i < 2; i++) {
-            if (i) {
-                *(descr_cur++) = '-';
-                *(descr_cur++) = '>';
-            }
-            pad = i ? (*filter)->outputs : (*filter)->inputs;
-            for (j = 0; pad[j].name; j++) {
-                if (descr_cur >= descr + sizeof(descr) - 4)
-                    break;
-                *(descr_cur++) = get_media_type_char(pad[j].type);
-            }
-            if (!j)
-                *(descr_cur++) = '|';
-        }
-        *descr_cur = 0;
-        printf("%-16s %-10s %s\n", (*filter)->name, descr, (*filter)->description);
-    }
+    while ((filter = av_filter_next(filter)) && *filter)
+        printf("%-16s %s\n", (*filter)->name, (*filter)->description);
 #endif
     return 0;
 }
@@ -905,14 +847,15 @@ int opt_pix_fmts(const char *opt, const char *arg)
 {
     enum PixelFormat pix_fmt;
 
-    printf("Pixel formats:\n"
-           "I.... = Supported Input  format for conversion\n"
-           ".O... = Supported Output format for conversion\n"
-           "..H.. = Hardware accelerated format\n"
-           "...P. = Paletted format\n"
-           "....B = Bitstream format\n"
-           "FLAGS NAME            NB_COMPONENTS BITS_PER_PIXEL\n"
-           "-----\n");
+    printf(
+        "Pixel formats:\n"
+        "I.... = Supported Input  format for conversion\n"
+        ".O... = Supported Output format for conversion\n"
+        "..H.. = Hardware accelerated format\n"
+        "...P. = Paletted format\n"
+        "....B = Bitstream format\n"
+        "FLAGS NAME            NB_COMPONENTS BITS_PER_PIXEL\n"
+        "-----\n");
 
 #if !CONFIG_SWSCALE
 #   define sws_isSupportedInput(x)  0
@@ -962,8 +905,7 @@ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size)
     FILE *f = fopen(filename, "rb");
 
     if (!f) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot read file '%s': %s\n", filename,
-               strerror(errno));
+        av_log(NULL, AV_LOG_ERROR, "Cannot read file '%s': %s\n", filename, strerror(errno));
         return AVERROR(errno);
     }
     fseek(f, 0, SEEK_END);
@@ -994,14 +936,14 @@ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size)
 }
 
 FILE *get_preset_file(char *filename, size_t filename_size,
-                      const char *preset_name, int is_path,
-                      const char *codec_name)
+                      const char *preset_name, int is_path, const char *codec_name)
 {
     FILE *f = NULL;
     int i;
-    const char *base[3] = { getenv("FFMPEG_DATADIR"),
-                            getenv("HOME"),
-                            FFMPEG_DATADIR, };
+    const char *base[3]= { getenv("FFMPEG_DATADIR"),
+                           getenv("HOME"),
+                           FFMPEG_DATADIR,
+                         };
 
     if (is_path) {
         av_strlcpy(filename, preset_name, filename_size);
@@ -1027,14 +969,11 @@ FILE *get_preset_file(char *filename, size_t filename_size,
         for (i = 0; i < 3 && !f; i++) {
             if (!base[i])
                 continue;
-            snprintf(filename, filename_size, "%s%s/%s.ffpreset", base[i],
-                     i != 1 ? "" : "/.ffmpeg", preset_name);
+            snprintf(filename, filename_size, "%s%s/%s.ffpreset", base[i], i != 1 ? "" : "/.ffmpeg", preset_name);
             f = fopen(filename, "r");
             if (!f && codec_name) {
                 snprintf(filename, filename_size,
-                         "%s%s/%s-%s.ffpreset",
-                         base[i],  i != 1 ? "" : "/.ffmpeg", codec_name,
-                         preset_name);
+                         "%s%s/%s-%s.ffpreset", base[i],  i != 1 ? "" : "/.ffmpeg", codec_name, preset_name);
                 f = fopen(filename, "r");
             }
         }
@@ -1045,23 +984,22 @@ FILE *get_preset_file(char *filename, size_t filename_size,
 
 int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec)
 {
-    if (*spec <= '9' && *spec >= '0') /* opt:index */
+    if (*spec <= '9' && *spec >= '0')                                        /* opt:index */
         return strtol(spec, NULL, 0) == st->index;
-    else if (*spec == 'v' || *spec == 'a' || *spec == 's' || *spec == 'd' ||
-             *spec == 't') { /* opt:[vasdt] */
+    else if (*spec == 'v' || *spec == 'a' || *spec == 's' || *spec == 'd' || *spec == 't') { /* opt:[vasdt] */
         enum AVMediaType type;
 
         switch (*spec++) {
-        case 'v': type = AVMEDIA_TYPE_VIDEO;      break;
-        case 'a': type = AVMEDIA_TYPE_AUDIO;      break;
-        case 's': type = AVMEDIA_TYPE_SUBTITLE;   break;
-        case 'd': type = AVMEDIA_TYPE_DATA;       break;
+        case 'v': type = AVMEDIA_TYPE_VIDEO;    break;
+        case 'a': type = AVMEDIA_TYPE_AUDIO;    break;
+        case 's': type = AVMEDIA_TYPE_SUBTITLE; break;
+        case 'd': type = AVMEDIA_TYPE_DATA;     break;
         case 't': type = AVMEDIA_TYPE_ATTACHMENT; break;
         default: abort(); // never reached, silence warning
         }
         if (type != st->codec->codec_type)
             return 0;
-        if (*spec++ == ':') { /* possibly followed by :index */
+        if (*spec++ == ':') {                                   /* possibly followed by :index */
             int i, index = strtol(spec, NULL, 0);
             for (i = 0; i < s->nb_streams; i++)
                 if (s->streams[i]->codec->codec_type == type && index-- == 0)
@@ -1080,9 +1018,8 @@ int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec)
 
             if (*endptr++ == ':') {
                 int stream_idx = strtol(endptr, NULL, 0);
-                return stream_idx >= 0 &&
-                    stream_idx < s->programs[i]->nb_stream_indexes &&
-                    st->index == s->programs[i]->stream_index[stream_idx];
+                return (stream_idx >= 0 && stream_idx < s->programs[i]->nb_stream_indexes &&
+                        st->index == s->programs[i]->stream_index[stream_idx]);
             }
 
             for (j = 0; j < s->programs[i]->nb_stream_indexes; j++)
@@ -1097,13 +1034,11 @@ int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec)
     return AVERROR(EINVAL);
 }
 
-AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec,
-                                AVFormatContext *s, AVStream *st)
+AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec, AVFormatContext *s, AVStream *st)
 {
     AVDictionary    *ret = NULL;
     AVDictionaryEntry *t = NULL;
-    int            flags = s->oformat ? AV_OPT_FLAG_ENCODING_PARAM
-                                      : AV_OPT_FLAG_DECODING_PARAM;
+    int            flags = s->oformat ? AV_OPT_FLAG_ENCODING_PARAM : AV_OPT_FLAG_DECODING_PARAM;
     char          prefix = 0;
     const AVClass    *cc = avcodec_get_class();
 
@@ -1111,18 +1046,9 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec,
         return NULL;
 
     switch (codec->type) {
-    case AVMEDIA_TYPE_VIDEO:
-        prefix  = 'v';
-        flags  |= AV_OPT_FLAG_VIDEO_PARAM;
-        break;
-    case AVMEDIA_TYPE_AUDIO:
-        prefix  = 'a';
-        flags  |= AV_OPT_FLAG_AUDIO_PARAM;
-        break;
-    case AVMEDIA_TYPE_SUBTITLE:
-        prefix  = 's';
-        flags  |= AV_OPT_FLAG_SUBTITLE_PARAM;
-        break;
+    case AVMEDIA_TYPE_VIDEO:    prefix = 'v'; flags |= AV_OPT_FLAG_VIDEO_PARAM;    break;
+    case AVMEDIA_TYPE_AUDIO:    prefix = 'a'; flags |= AV_OPT_FLAG_AUDIO_PARAM;    break;
+    case AVMEDIA_TYPE_SUBTITLE: prefix = 's'; flags |= AV_OPT_FLAG_SUBTITLE_PARAM; break;
     }
 
     while (t = av_dict_get(opts, "", t, AV_DICT_IGNORE_SUFFIX)) {
@@ -1137,14 +1063,10 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec,
             }
 
         if (av_opt_find(&cc, t->key, NULL, flags, AV_OPT_SEARCH_FAKE_OBJ) ||
-            (codec && codec->priv_class &&
-             av_opt_find(&codec->priv_class, t->key, NULL, flags,
-                         AV_OPT_SEARCH_FAKE_OBJ)))
+            (codec && codec->priv_class && av_opt_find(&codec->priv_class, t->key, NULL, flags, AV_OPT_SEARCH_FAKE_OBJ)))
             av_dict_set(&ret, t->key, t->value, 0);
-        else if (t->key[0] == prefix &&
-                 av_opt_find(&cc, t->key + 1, NULL, flags,
-                             AV_OPT_SEARCH_FAKE_OBJ))
-            av_dict_set(&ret, t->key + 1, t->value, 0);
+        else if (t->key[0] == prefix && av_opt_find(&cc, t->key+1, NULL, flags, AV_OPT_SEARCH_FAKE_OBJ))
+            av_dict_set(&ret, t->key+1, t->value, 0);
 
         if (p)
             *p = ':';
@@ -1152,8 +1074,7 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec,
     return ret;
 }
 
-AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
-                                           AVDictionary *codec_opts)
+AVDictionary **setup_find_stream_info_opts(AVFormatContext *s, AVDictionary *codec_opts)
 {
     int i;
     AVDictionary **opts;
@@ -1162,13 +1083,11 @@ AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
         return NULL;
     opts = av_mallocz(s->nb_streams * sizeof(*opts));
     if (!opts) {
-        av_log(NULL, AV_LOG_ERROR,
-               "Could not alloc memory for stream options.\n");
+        av_log(NULL, AV_LOG_ERROR, "Could not alloc memory for stream options.\n");
         return NULL;
     }
     for (i = 0; i < s->nb_streams; i++)
-        opts[i] = filter_codec_opts(codec_opts, avcodec_find_decoder(s->streams[i]->codec->codec_id),
-                                    s, s->streams[i]);
+        opts[i] = filter_codec_opts(codec_opts, avcodec_find_decoder(s->streams[i]->codec->codec_id), s, s->streams[i]);
     return opts;
 }
 

cmdutils.h

@@ -43,11 +43,6 @@ extern const char program_name[];
  */
 extern const int program_birth_year;
 
-/**
- * this year, defined by the program for show_banner()
- */
-extern const int this_year;
-
 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
 extern struct SwsContext *sws_opts;
@@ -83,8 +78,6 @@ int opt_loglevel(const char *opt, const char *arg);
 
 int opt_report(const char *opt);
 
-int opt_max_alloc(const char *opt, const char *arg);
-
 int opt_codec_debug(const char *opt, const char *arg);
 
 /**
@@ -98,15 +91,14 @@ int opt_timelimit(const char *opt, const char *arg);
  * parsed or the corresponding value is invalid.
  *
  * @param context the context of the value to be set (e.g. the
- * corresponding command line option name)
+ * corresponding commandline option name)
  * @param numstr the string to be parsed
  * @param type the type (OPT_INT64 or OPT_FLOAT) as which the
  * string should be parsed
  * @param min the minimum valid accepted value
  * @param max the maximum valid accepted value
  */
-double parse_number_or_die(const char *context, const char *numstr, int type,
-                           double min, double max);
+double parse_number_or_die(const char *context, const char *numstr, int type, double min, double max);
 
 /**
  * Parse a string specifying a time and return its corresponding
@@ -114,7 +106,7 @@ double parse_number_or_die(const char *context, const char *numstr, int type,
  * the string cannot be correctly parsed.
  *
  * @param context the context of the value to be set (e.g. the
- * corresponding command line option name)
+ * corresponding commandline option name)
  * @param timestr the string to be parsed
  * @param is_duration a flag which tells how to interpret timestr, if
  * not zero timestr is interpreted as a duration, otherwise as a
@@ -122,8 +114,7 @@ double parse_number_or_die(const char *context, const char *numstr, int type,
  *
  * @see parse_date()
  */
-int64_t parse_time_or_die(const char *context, const char *timestr,
-                          int is_duration);
+int64_t parse_time_or_die(const char *context, const char *timestr, int is_duration);
 
 typedef struct SpecifierOpt {
     char *specifier;    /**< stream/chapter/program/... specifier */
@@ -169,8 +160,7 @@ typedef struct {
     const char *argname;
 } OptionDef;
 
-void show_help_options(const OptionDef *options, const char *msg, int mask,
-                       int value);
+void show_help_options(const OptionDef *options, const char *msg, int mask, int value);
 
 /**
  * Show help for all options with given flags in class and all its
@@ -196,11 +186,10 @@ void parse_options(void *optctx, int argc, char **argv, const OptionDef *options
  *
  * @return on success 1 if arg was consumed, 0 otherwise; negative number on error
  */
-int parse_option(void *optctx, const char *opt, const char *arg,
-                 const OptionDef *options);
+int parse_option(void *optctx, const char *opt, const char *arg, const OptionDef *options);
 
 /**
- * Find the '-loglevel' option in the command line args and apply it.
+ * Find the '-loglevel' option in the commandline args and apply it.
  */
 void parse_loglevel(int argc, char **argv, const OptionDef *options);
 
@@ -225,8 +214,7 @@ int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec);
  * @param st A stream from s for which the options should be filtered.
  * @return a pointer to the created dictionary
  */
-AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec,
-                                AVFormatContext *s, AVStream *st);
+AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec, AVFormatContext *s, AVStream *st);
 
 /**
  * Setup AVCodecContext options for avformat_find_stream_info().
@@ -239,8 +227,7 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, AVCodec *codec,
  * @return pointer to the created array of dictionaries, NULL if it
  * cannot be created
  */
-AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
-                                           AVDictionary *codec_opts);
+AVDictionary **setup_find_stream_info_opts(AVFormatContext *s, AVDictionary *codec_opts);
 
 /**
  * Print an error message to stderr, indicating filename and a human

cmdutils_common_opts.h

@@ -15,4 +15,3 @@
     { "v", HAS_ARG, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },
     { "debug", HAS_ARG, {(void*)opt_codec_debug}, "set debug flags", "flags" },
     { "report", 0, {(void*)opt_report}, "generate a report" },
-    { "max_alloc", HAS_ARG, {(void*)opt_max_alloc}, "set maximum size of a single allocated block", "bytes" },

configure

@@ -81,6 +81,7 @@ Configuration options:
                            and binaries will be unredistributable [no]
   --disable-doc            do not build documentation
   --disable-ffmpeg         disable ffmpeg build
+  --enable-avconv          enable  avconv build
   --disable-ffplay         disable ffplay build
   --disable-ffprobe        disable ffprobe build
   --disable-ffserver       disable ffserver build
@@ -217,13 +218,12 @@ Advanced options (experts only):
   --target-os=OS           compiler targets OS [$target_os]
   --target-exec=CMD        command to run executables on target
   --target-path=DIR        path to view of build directory on target
-  --nm=NM                  use nm tool NM [$nm_default]
+  --nm=NM                  use nm tool
   --ar=AR                  use archive tool AR [$ar_default]
   --as=AS                  use assembler AS [$as_default]
-  --yasmexe=EXE            use yasm-compatible assembler EXE [$yasmexe_default]
   --cc=CC                  use C compiler CC [$cc_default]
   --cxx=CXX                use C compiler CXX [$cxx_default]
-  --ld=LD                  use linker LD [$ld_default]
+  --ld=LD                  use linker LD
   --host-cc=HOSTCC         use host C compiler HOSTCC
   --host-cflags=HCFLAGS    use HCFLAGS when compiling for host
   --host-ldflags=HLDFLAGS  use HLDFLAGS when linking for host
@@ -253,7 +253,7 @@ Advanced options (experts only):
   --disable-armvfp         disable ARM VFP optimizations
   --disable-iwmmxt         disable iwmmxt optimizations
   --disable-mmi            disable MMI optimizations
-  --disable-neon           disable NEON optimizations
+  --disable-neon           disable neon optimizations
   --disable-vis            disable VIS optimizations
   --disable-yasm           disable use of yasm assembler
   --enable-pic             build position-independent code
@@ -261,8 +261,6 @@ Advanced options (experts only):
   --enable-sram            allow use of on-chip SRAM
   --disable-symver         disable symbol versioning
   --optflags               override optimization-related compiler flags
-  --postproc-version=V     build libpostproc version V.
-                           Where V can be '$ALT_PP_VER_MAJOR.$ALT_PP_VER_MINOR.$ALT_PP_VER_MICRO' or 'current'. [$postproc_version_default]
 
 Developer options (useful when working on FFmpeg itself):
   --enable-coverage        build with test coverage instrumentation
@@ -271,9 +269,6 @@ Developer options (useful when working on FFmpeg itself):
   --disable-optimizations  disable compiler optimizations
   --enable-extra-warnings  enable more compiler warnings
   --disable-stripping      disable stripping of executables and shared libraries
-  --valgrind=VALGRIND      run "make fate" tests through valgrind to detect memory
-                           leaks and errors, using the specified valgrind binary.
-                           Cannot be combined with --target-exec
   --samples=PATH           location of test samples for FATE, if not set use
                            \$FATE_SAMPLES at make invocation time.
 
@@ -986,19 +981,8 @@ COMPONENT_LIST="
     protocols
 "
 
-PROGRAM_LIST="
-    ffplay
-    ffprobe
-    ffserver
-    ffmpeg
-"
-
 CONFIG_LIST="
     $COMPONENT_LIST
-    $PROGRAM_LIST
-    avplay
-    avprobe
-    avserver
     aandct
     ac3dsp
     avcodec
@@ -1013,13 +997,17 @@ CONFIG_LIST="
     dwt
     dxva2
     fastdiv
+    ffmpeg
+    avconv
+    ffplay
+    ffprobe
+    ffserver
     fft
     frei0r
     gnutls
     golomb
     gpl
     gray
-    h264chroma
     h264dsp
     h264pred
     hardcoded_tables
@@ -1156,6 +1144,7 @@ HAVE_LIST="
     asm_types_h
     attribute_may_alias
     attribute_packed
+    bswap
     cbrtf
     closesocket
     cmov
@@ -1163,11 +1152,12 @@ HAVE_LIST="
     dev_bktr_ioctl_bt848_h
     dev_bktr_ioctl_meteor_h
     dev_ic_bt8xx_h
-    dev_video_bktr_ioctl_bt848_h
     dev_video_meteor_ioctl_meteor_h
+    dev_video_bktr_ioctl_bt848_h
     dlfcn_h
     dlopen
     dos_paths
+    dxva_h
     ebp_available
     ebx_available
     exp2
@@ -1179,11 +1169,11 @@ HAVE_LIST="
     fork
     getaddrinfo
     gethrtime
-    GetProcessAffinityMask
     GetProcessMemoryInfo
     GetProcessTimes
     getrusage
     gnu_as
+    struct_rusage_ru_maxrss
     ibm_asm
     inet_aton
     inline_asm
@@ -1203,41 +1193,34 @@ HAVE_LIST="
     lzo1x_999_compress
     machine_ioctl_bt848_h
     machine_ioctl_meteor_h
-    makeinfo
     malloc_h
     MapViewOfFile
     memalign
     mkstemp
     mmap
     PeekNamedPipe
-    poll_h
     posix_memalign
     round
     roundf
-    sched_getaffinity
     sdl
     sdl_video_size
     setmode
-    setrlimit
     sndio_h
     socklen_t
     soundcard_h
+    poll_h
+    setrlimit
     strerror_r
     strptime
     struct_addrinfo
     struct_ipv6_mreq
-    struct_rusage_ru_maxrss
     struct_sockaddr_in6
     struct_sockaddr_sa_len
     struct_sockaddr_storage
-    struct_v4l2_frmivalenum_discrete
     symver
-    symver_asm_label
     symver_gnu_asm
-    sysconf
-    sysctl
+    symver_asm_label
     sys_mman_h
-    sys_param_h
     sys_resource_h
     sys_select_h
     sys_soundcard_h
@@ -1319,9 +1302,6 @@ CMDLINE_SET="
     target_exec
     target_os
     target_path
-    postproc_version
-    valgrind
-    yasmexe
 "
 
 CMDLINE_APPEND="
@@ -1362,8 +1342,8 @@ fast_64bit_if_any="alpha ia64 mips64 parisc64 ppc64 sparc64 x86_64"
 fast_clz_if_any="alpha armv5te avr32 mips ppc x86"
 fast_unaligned_if_any="armv6 ppc x86"
 
-inline_asm_deps="!tms470"
 need_memalign="altivec neon sse"
+inline_asm_deps="!tms470"
 
 symver_if_any="symver_asm_label symver_gnu_asm"
 
@@ -1414,8 +1394,8 @@ h263_encoder_select="aandct"
 h263_vaapi_hwaccel_select="vaapi h263_decoder"
 h263i_decoder_select="h263_decoder"
 h263p_encoder_select="h263_encoder"
+h264_decoder_select="golomb h264dsp h264pred"
 h264_crystalhd_decoder_select="crystalhd h264_mp4toannexb_bsf h264_parser"
-h264_decoder_select="golomb h264chroma h264dsp h264pred"
 h264_dxva2_hwaccel_deps="dxva2api_h"
 h264_dxva2_hwaccel_select="dxva2 h264_decoder"
 h264_vaapi_hwaccel_select="vaapi h264_decoder"
@@ -1430,34 +1410,32 @@ loco_decoder_select="golomb"
 mjpeg_encoder_select="aandct"
 mlp_decoder_select="mlp_parser"
 mp1_decoder_select="mpegaudiodsp"
-mp1float_decoder_select="mpegaudiodsp"
 mp2_decoder_select="mpegaudiodsp"
-mp2float_decoder_select="mpegaudiodsp"
-mp3_decoder_select="mpegaudiodsp"
 mp3adu_decoder_select="mpegaudiodsp"
+mp3_decoder_select="mpegaudiodsp"
+mp3on4_decoder_select="mpegaudiodsp"
+mp1float_decoder_select="mpegaudiodsp"
+mp2float_decoder_select="mpegaudiodsp"
 mp3adufloat_decoder_select="mpegaudiodsp"
 mp3float_decoder_select="mpegaudiodsp"
-mp3on4_decoder_select="mpegaudiodsp"
 mp3on4float_decoder_select="mpegaudiodsp"
-mpc7_decoder_select="mpegaudiodsp"
-mpc8_decoder_select="mpegaudiodsp"
+mpeg1video_encoder_select="aandct"
+mpeg2video_encoder_select="aandct"
+mpeg4_decoder_select="h263_decoder mpeg4video_parser"
+mpeg4_encoder_select="h263_encoder"
 mpeg_vdpau_decoder_select="vdpau mpegvideo_decoder"
-mpeg_xvmc_decoder_deps="X11_extensions_XvMClib_h"
-mpeg_xvmc_decoder_select="mpegvideo_decoder"
 mpeg1_vdpau_decoder_select="vdpau mpeg1video_decoder"
 mpeg1_vdpau_hwaccel_select="vdpau mpeg1video_decoder"
-mpeg1video_encoder_select="aandct"
 mpeg2_crystalhd_decoder_select="crystalhd"
 mpeg2_dxva2_hwaccel_deps="dxva2api_h"
 mpeg2_dxva2_hwaccel_select="dxva2 mpeg2video_decoder"
 mpeg2_vdpau_hwaccel_select="vdpau mpeg2video_decoder"
 mpeg2_vaapi_hwaccel_select="vaapi mpeg2video_decoder"
-mpeg2video_encoder_select="aandct"
 mpeg4_crystalhd_decoder_select="crystalhd"
-mpeg4_decoder_select="h263_decoder mpeg4video_parser"
-mpeg4_encoder_select="h263_encoder"
 mpeg4_vaapi_hwaccel_select="vaapi mpeg4_decoder"
 mpeg4_vdpau_decoder_select="vdpau mpeg4_decoder"
+mpeg_xvmc_decoder_deps="X11_extensions_XvMClib_h"
+mpeg_xvmc_decoder_select="mpegvideo_decoder"
 msmpeg4_crystalhd_decoder_select="crystalhd"
 msmpeg4v1_decoder_select="h263_decoder"
 msmpeg4v1_encoder_select="h263_encoder"
@@ -1476,8 +1454,8 @@ rv10_decoder_select="h263_decoder"
 rv10_encoder_select="h263_encoder"
 rv20_decoder_select="h263_decoder"
 rv20_encoder_select="h263_encoder"
-rv30_decoder_select="golomb h264chroma h264pred"
-rv40_decoder_select="golomb h264chroma h264pred"
+rv30_decoder_select="golomb h264pred"
+rv40_decoder_select="golomb h264pred"
 shorten_decoder_select="golomb"
 sipr_decoder_select="lsp"
 snow_decoder_select="dwt"
@@ -1486,7 +1464,7 @@ sonic_decoder_select="golomb"
 sonic_encoder_select="golomb"
 sonic_ls_encoder_select="golomb"
 svq1_encoder_select="aandct"
-svq3_decoder_select="golomb h264chroma h264dsp h264pred"
+svq3_decoder_select="golomb h264dsp h264pred"
 svq3_decoder_suggest="zlib"
 theora_decoder_select="vp3_decoder"
 tiff_decoder_suggest="zlib"
@@ -1494,8 +1472,8 @@ tiff_encoder_suggest="zlib"
 truehd_decoder_select="mlp_decoder"
 tscc_decoder_select="zlib"
 twinvq_decoder_select="mdct lsp sinewin"
+vc1_decoder_select="h263_decoder"
 vc1_crystalhd_decoder_select="crystalhd"
-vc1_decoder_select="h263_decoder h264chroma"
 vc1_dxva2_hwaccel_deps="dxva2api_h"
 vc1_dxva2_hwaccel_select="dxva2 vc1_decoder"
 vc1_vaapi_hwaccel_select="vaapi vc1_decoder"
@@ -1534,7 +1512,7 @@ vda_deps="VideoDecodeAcceleration_VDADecoder_h pthreads"
 vdpau_deps="vdpau_vdpau_h vdpau_vdpau_x11_h"
 
 # parsers
-h264_parser_select="golomb h264chroma h264dsp h264pred"
+h264_parser_select="golomb h264dsp h264pred"
 
 # external libraries
 libaacplus_encoder_deps="libaacplus"
@@ -1660,9 +1638,7 @@ mp_filter_deps="gpl avcodec"
 mptestsrc_filter_deps="gpl"
 negate_filter_deps="lut_filter"
 ocv_filter_deps="libopencv"
-pan_filter_deps="swresample"
 scale_filter_deps="swscale"
-tinterlace_filter_deps="gpl"
 yadif_filter_deps="gpl"
 
 # libraries
@@ -1671,13 +1647,15 @@ avformat_deps="avcodec"
 postproc_deps="gpl"
 
 # programs
+ffmpeg_deps="avcodec avformat swscale swresample"
+ffmpeg_select="buffer_filter buffersink_filter"
+avconv_deps="avcodec avformat swscale"
+avconv_select="buffer_filter"
 ffplay_deps="avcodec avformat swscale sdl"
 ffplay_select="buffersink_filter rdft"
 ffprobe_deps="avcodec avformat"
 ffserver_deps="avformat ffm_muxer fork rtp_protocol rtsp_demuxer"
 ffserver_extralibs='$ldl'
-ffmpeg_deps="avcodec avformat swscale swresample"
-ffmpeg_select="buffersink_filter"
 
 doc_deps="texi2html"
 
@@ -1700,6 +1678,7 @@ mxf_d10_test_deps="avfilter"
 seek_lavf_mxf_d10_test_deps="mxf_d10_test"
 
 test_deps _encoder _decoder                                             \
+    adpcm_g726=g726                                                     \
     adpcm_ima_qt                                                        \
     adpcm_ima_wav                                                       \
     adpcm_ms                                                            \
@@ -1715,7 +1694,6 @@ test_deps _encoder _decoder                                             \
     flac                                                                \
     flashsv                                                             \
     flv                                                                 \
-    adpcm_g726=g726                                                     \
     gif                                                                 \
     h261                                                                \
     h263="h263 h263p"                                                   \
@@ -1789,7 +1767,6 @@ incdir_default='${prefix}/include'
 libdir_default='${prefix}/lib'
 mandir_default='${prefix}/share/man'
 shlibdir_default="$libdir_default"
-postproc_version_default="current"
 
 # toolchain
 ar_default="ar"
@@ -1804,10 +1781,10 @@ objformat="elf"
 pkg_config_default=pkg-config
 ranlib="ranlib"
 strip_default="strip"
-yasmexe_default="yasm"
+yasmexe="yasm"
+nogas=":"
 
 nm_opts='-g'
-nogas=":"
 
 # machine
 arch_default=$(uname -m)
@@ -1817,33 +1794,29 @@ cpu="generic"
 target_os_default=$(tolower $(uname -s))
 host_os=$target_os_default
 
-# alternative libpostproc version
-ALT_PP_VER_MAJOR=51
-ALT_PP_VER_MINOR=2
-ALT_PP_VER_MICRO=101
-ALT_PP_VER=$ALT_PP_VER_MAJOR.$ALT_PP_VER_MINOR.$ALT_PP_VER_MICRO
-
 # configurable options
-enable $PROGRAM_LIST
-
 enable avcodec
 enable avdevice
 enable avfilter
 enable avformat
 enable avutil
-enable postproc
-enable stripping
-enable swresample
-enable swscale
-
 enable asm
 enable debug
 enable doc
 enable fastdiv
+enable ffmpeg
+enable ffplay
+enable ffprobe
+enable ffserver
 enable network
 enable optimizations
+enable postproc
+enable protocols
 enable safe_bitstream_reader
 enable static
+enable stripping
+enable swresample
+enable swscale
 enable swscale_alpha
 
 # build settings
@@ -1913,20 +1886,6 @@ INDEV_LIST=$(find_things    indev    _IN      libavdevice/alldevices.c)
 PROTOCOL_LIST=$(find_things protocol PROTOCOL libavformat/allformats.c)
 FILTER_LIST=$(find_things   filter   FILTER   libavfilter/allfilters.c)
 
-ALL_COMPONENTS="
-    $BSF_LIST
-    $DECODER_LIST
-    $DEMUXER_LIST
-    $ENCODER_LIST
-    $FILTER_LIST
-    $HWACCEL_LIST
-    $INDEV_LIST
-    $MUXER_LIST
-    $OUTDEV_LIST
-    $PARSER_LIST
-    $PROTOCOL_LIST
-"
-
 find_tests(){
     map "echo ${2}\${v}_test" $(ls "$source_path"/tests/ref/$1 | grep -v '[^-a-z0-9_]')
 }
@@ -1937,8 +1896,6 @@ LAVF_TESTS=$(find_tests lavf)
 LAVFI_TESTS=$(find_tests lavfi)
 SEEK_TESTS=$(find_tests seek seek_)
 
-ALL_TESTS="$ACODEC_TESTS $VCODEC_TESTS $LAVF_TESTS $LAVFI_TESTS $SEEK_TESTS"
-
 pcm_test_deps=$(map 'echo ${v%_*}_decoder $v' $(filter pcm_* $ENCODER_LIST))
 
 for n in $COMPONENT_LIST; do
@@ -1947,7 +1904,7 @@ for n in $COMPONENT_LIST; do
     eval ${n}_if_any="\$$v"
 done
 
-enable $ARCH_EXT_LIST $ALL_TESTS
+enable $ARCH_EXT_LIST $ACODEC_TESTS $VCODEC_TESTS $LAVF_TESTS $LAVFI_TESTS $SEEK_TESTS
 
 die_unknown(){
     echo "Unknown option \"$1\"."
@@ -2029,17 +1986,7 @@ if enabled cross_compile; then
         die "Must specify target arch and OS when cross-compiling"
 fi
 
-set_default arch target_os postproc_version
-
-# Check if we should build alternative libpostproc version instead of current
-if   test "$postproc_version" = $ALT_PP_VER; then
-  LIBPOSTPROC_VERSION=$ALT_PP_VER
-  LIBPOSTPROC_VERSION_MAJOR=$ALT_PP_VER_MAJOR
-  LIBPOSTPROC_VERSION_MINOR=$ALT_PP_VER_MINOR
-  LIBPOSTPROC_VERSION_MICRO=$ALT_PP_VER_MICRO
-elif test "$postproc_version" != current; then
-  die "Invalid argument to --postproc-version. See --help output."
-fi
+set_default arch target_os
 
 ar_default="${cross_prefix}${ar_default}"
 cc_default="${cross_prefix}${cc_default}"
@@ -2051,7 +1998,7 @@ strip_default="${cross_prefix}${strip_default}"
 
 sysinclude_default="${sysroot}/usr/include"
 
-set_default cc cxx nm pkg_config strip sysinclude yasmexe
+set_default cc cxx nm pkg_config strip sysinclude
 enabled cross_compile || host_cc_default=$cc
 set_default host_cc
 
@@ -2092,15 +2039,15 @@ tmpfile(){
 
 trap 'rm -f -- $TMPFILES' EXIT
 
-tmpfile TMPASM .asm
-tmpfile TMPC   .c
+tmpfile TMPC  .c
 tmpfile TMPCPP .cpp
-tmpfile TMPE   $EXESUF
-tmpfile TMPH   .h
-tmpfile TMPO   .o
-tmpfile TMPS   .S
-tmpfile TMPSH  .sh
-tmpfile TMPV   .ver
+tmpfile TMPE  $EXESUF
+tmpfile TMPH  .h
+tmpfile TMPO  .o
+tmpfile TMPS  .S
+tmpfile TMPV  .ver
+tmpfile TMPSH .sh
+tmpfile TMPASM .asm
 
 unset -f mktemp
 
@@ -2119,9 +2066,9 @@ EOF
     die "Sanity test failed."
 fi
 
-filter_asflags=echo
 filter_cflags=echo
 filter_cppflags=echo
+filter_asflags=echo
 
 if   $cc -v 2>&1 | grep -q '^gcc.*LLVM'; then
     cc_type=llvm_gcc
@@ -2944,6 +2891,8 @@ EOF
     enabled ssse3 && check_asm ssse3 '"pabsw %xmm0, %xmm0"'
     enabled mmx2  && check_asm mmx2  '"pmaxub %mm0, %mm1"'
 
+    check_asm bswap '"bswap %%eax" ::: "%eax"'
+
     if ! disabled_any asm mmx yasm; then
         if check_cmd $yasmexe --version; then
             enabled x86_64 && yasm_extra="-m amd64"
@@ -3033,26 +2982,22 @@ check_func  ${malloc_prefix}posix_memalign      && enable posix_memalign
 check_func  setrlimit
 check_func  strerror_r
 check_func  strptime
-check_func  sched_getaffinity
-check_func  sysconf
-check_func  sysctl
 check_func_headers conio.h kbhit
 check_func_headers windows.h PeekNamedPipe
 check_func_headers io.h setmode
 check_func_headers lzo/lzo1x.h lzo1x_999_compress
 check_lib2 "windows.h psapi.h" GetProcessMemoryInfo -lpsapi
-check_func_headers windows.h GetProcessAffinityMask
 check_func_headers windows.h GetProcessTimes
 check_func_headers windows.h MapViewOfFile
 check_func_headers windows.h VirtualAlloc
 
 check_header dlfcn.h
+check_header dxva.h
 check_header dxva2api.h -D_WIN32_WINNT=0x0600
 check_header libcrystalhd/libcrystalhd_if.h
 check_header malloc.h
 check_header poll.h
 check_header sys/mman.h
-check_header sys/param.h
 check_header sys/resource.h
 check_header sys/select.h
 check_header termios.h
@@ -3127,9 +3072,7 @@ enabled frei0r     && { check_header frei0r.h || die "ERROR: frei0r.h header not
 enabled gnutls     && require_pkg_config gnutls gnutls/gnutls.h gnutls_global_init
 enabled libaacplus && require  "libaacplus >= 2.0.0" aacplus.h aacplusEncOpen -laacplus
 enabled libass     && require_pkg_config libass ass/ass.h ass_library_init
-enabled libcelt    && require libcelt celt/celt.h celt_decode -lcelt0 &&
-                      { check_lib celt/celt.h celt_decoder_create_custom -lcelt0 ||
-                        die "ERROR: libcelt version must be >= 0.11.0."; }
+enabled libcelt    && require libcelt celt/celt.h celt_decode -lcelt0
 enabled libdc1394  && require_pkg_config libdc1394-2 dc1394/dc1394.h dc1394_new
 enabled libdirac   && require_pkg_config dirac                          \
     "libdirac_decoder/dirac_parser.h libdirac_encoder/dirac_encoder.h"  \
@@ -3196,13 +3139,10 @@ fi
 enabled sdl && add_cflags $sdl_cflags && add_extralibs $sdl_libs
 
 texi2html -version > /dev/null 2>&1 && enable texi2html || disable texi2html
-makeinfo --version > /dev/null 2>&1 && enable makeinfo  || disable makeinfo
 
 check_header linux/fb.h
 check_header linux/videodev.h
 check_header linux/videodev2.h
-check_struct linux/videodev2.h "struct v4l2_frmivalenum" discrete
-
 check_header sys/videoio.h
 
 check_func_headers "windows.h vfw.h" capCreateCaptureWindow "$vfwcap_indev_extralibs"
@@ -3265,7 +3205,6 @@ fi
 
 enabled debug && add_cflags -g"$debuglevel" && add_asflags -g"$debuglevel"
 enabled coverage && add_cflags "-fprofile-arcs -ftest-coverage" && add_ldflags "-fprofile-arcs -ftest-coverage"
-test -n "$valgrind" && target_exec="$valgrind --error-exitcode=1 --malloc-fill=0x2a --track-origins=yes --leak-check=full --gen-suppressions=all --suppressions=$source_path/tests/fate-valgrind.supp"
 
 # add some useful compiler flags if supported
 check_cflags -Wdeclaration-after-statement
@@ -3354,11 +3293,13 @@ elif enabled gcc; then
     check_cflags -fno-tree-vectorize
     check_cflags -Werror=implicit-function-declaration
     check_cflags -Werror=missing-prototypes
+    check_cflags -Werror=return-type
 elif enabled llvm_gcc; then
     check_cflags -mllvm -stack-alignment=16
 elif enabled clang; then
     check_cflags -mllvm -stack-alignment=16
     check_cflags -Qunused-arguments
+    check_cflags -Werror=return-type
 elif enabled armcc; then
     # 2523: use of inline assembler is deprecated
     add_cflags -W${armcc_opt},--diag_suppress=2523
@@ -3378,8 +3319,22 @@ enabled_any $THREADS_LIST      && enable threads
 check_deps $CONFIG_LIST       \
            $CONFIG_EXTRA      \
            $HAVE_LIST         \
-           $ALL_COMPONENTS    \
-           $ALL_TESTS         \
+           $DECODER_LIST      \
+           $ENCODER_LIST      \
+           $HWACCEL_LIST      \
+           $PARSER_LIST       \
+           $BSF_LIST          \
+           $DEMUXER_LIST      \
+           $MUXER_LIST        \
+           $FILTER_LIST       \
+           $INDEV_LIST        \
+           $OUTDEV_LIST       \
+           $PROTOCOL_LIST     \
+           $ACODEC_TESTS      \
+           $VCODEC_TESTS      \
+           $LAVF_TESTS        \
+           $LAVFI_TESTS       \
+           $SEEK_TESTS        \
 
 enabled asm || { arch=c; disable $ARCH_LIST $ARCH_EXT_LIST; }
 
@@ -3608,24 +3563,21 @@ EOF
 get_version(){
     name=$1
     file=$source_path/$2
-# This condition will be removed when we stop supporting old libpostproc versions
-if ! test "$name" = LIBPOSTPROC || test "$postproc_version" = current; then
     eval $(grep "#define ${name}_VERSION_M" "$file" | awk '{ print $2"="$3 }')
     eval ${name}_VERSION=\$${name}_VERSION_MAJOR.\$${name}_VERSION_MINOR.\$${name}_VERSION_MICRO
-fi
     lcname=$(tolower $name)
     eval echo "${lcname}_VERSION=\$${name}_VERSION" >> config.mak
     eval echo "${lcname}_VERSION_MAJOR=\$${name}_VERSION_MAJOR" >> config.mak
 }
 
+get_version LIBSWSCALE  libswscale/swscale.h
+get_version LIBSWRESAMPLE libswresample/swresample.h
+get_version LIBPOSTPROC libpostproc/postprocess.h
 get_version LIBAVCODEC  libavcodec/version.h
 get_version LIBAVDEVICE libavdevice/avdevice.h
-get_version LIBAVFILTER libavfilter/version.h
 get_version LIBAVFORMAT libavformat/version.h
 get_version LIBAVUTIL   libavutil/avutil.h
-get_version LIBPOSTPROC libpostproc/postprocess.h
-get_version LIBSWRESAMPLE libswresample/swresample.h
-get_version LIBSWSCALE  libswscale/swscale.h
+get_version LIBAVFILTER libavfilter/avfilter.h
 
 cat > $TMPH <<EOF
 /* Automatically generated by configure - do not modify! */
@@ -3664,7 +3616,17 @@ print_config ARCH_   "$config_files" $ARCH_LIST
 print_config HAVE_   "$config_files" $HAVE_LIST
 print_config CONFIG_ "$config_files" $CONFIG_LIST       \
                                      $CONFIG_EXTRA      \
-                                     $ALL_COMPONENTS    \
+                                     $DECODER_LIST      \
+                                     $ENCODER_LIST      \
+                                     $HWACCEL_LIST      \
+                                     $PARSER_LIST       \
+                                     $BSF_LIST          \
+                                     $DEMUXER_LIST      \
+                                     $MUXER_LIST        \
+                                     $FILTER_LIST       \
+                                     $PROTOCOL_LIST     \
+                                     $INDEV_LIST        \
+                                     $OUTDEV_LIST       \
 
 cat >>config.mak <<EOF
 ACODEC_TESTS=$(print_enabled -n _test $ACODEC_TESTS)
@@ -3689,12 +3651,6 @@ cat > $TMPH <<EOF
 #define AVUTIL_AVCONFIG_H
 EOF
 
-test "$postproc_version" != current && cat >> $TMPH <<EOF
-#define LIBPOSTPROC_VERSION_MAJOR $LIBPOSTPROC_VERSION_MAJOR
-#define LIBPOSTPROC_VERSION_MINOR $LIBPOSTPROC_VERSION_MINOR
-#define LIBPOSTPROC_VERSION_MICRO $LIBPOSTPROC_VERSION_MICRO
-EOF
-
 print_config AV_HAVE_ $TMPH $HAVE_LIST_PUB
 
 echo "#endif /* AVUTIL_AVCONFIG_H */" >> $TMPH

doc/APIchanges

@@ -13,26 +13,6 @@ libavutil:   2011-04-18
 
 API changes, most recent first:
 
-2012-01-24 - xxxxxxx - lavfi 2.60.100
-  Add avfilter_graph_dump.
-
-2012-01-25 - lavf 53.22.0
-  f1caf01 Allow doing av_write_frame(ctx, NULL) for flushing possible
-          buffered data within a muxer. Added AVFMT_ALLOW_FLUSH for
-          muxers supporting it (av_write_frame makes sure it is called
-          only for muxers with this flag).
-
-2012-01-15 - lavc 53.34.0
-  New audio encoding API:
-  b2c75b6 Add CODEC_CAP_VARIABLE_FRAME_SIZE capability for use by audio
-          encoders.
-  5ee5fa0 Add avcodec_fill_audio_frame() as a convenience function.
-  b2c75b6 Add avcodec_encode_audio2() and deprecate avcodec_encode_audio().
-          Add AVCodec.encode2().
-
-2012-01-12 - 3167dc9 - lavfi 2.15.0
-  Add a new installed header -- libavfilter/version.h -- with version macros.
-
 2011-12-08 - a502939 - lavfi 2.52.0
   Add av_buffersink_poll_frame() to buffersink.h.
 
@@ -51,37 +31,31 @@ API changes, most recent first:
 2011-10-20 - b35e9e1 - lavu 51.22.0
   Add av_strtok() to avstring.h.
 
-2011-01-03 - b73ec05 - lavu 51.21.0
-  Add av_popcount64
-
-2011-12-18 - 8400b12 - lavc 53.28.1
-  Deprecate AVFrame.age. The field is unused.
-
-2011-12-12 - 5266045 - lavf 53.17.0
-  Add avformat_close_input().
+2011-xx-xx - xxxxxxx - lavf 53.17.0
+  Add avformat_open_input().
   Deprecate av_close_input_file() and av_close_input_stream().
 
-2011-12-02 - 0eea212 - lavc 53.25.0
+2011-xx-xx - xxxxxxx - lavc 53.25.0
   Add nb_samples and extended_data fields to AVFrame.
   Deprecate AVCODEC_MAX_AUDIO_FRAME_SIZE.
   Deprecate avcodec_decode_audio3() in favor of avcodec_decode_audio4().
   avcodec_decode_audio4() writes output samples to an AVFrame, which allows
   audio decoders to use get_buffer().
 
-2011-12-04 - 560f773 - lavc 53.24.0
+2011-xx-xx - xxxxxxx - lavc 53.24.0
   Change AVFrame.data[4]/base[4]/linesize[4]/error[4] to [8] at next major bump.
   Change AVPicture.data[4]/linesize[4] to [8] at next major bump.
   Change AVCodecContext.error[4] to [8] at next major bump.
   Add AV_NUM_DATA_POINTERS to simplify the bump transition.
 
-2011-11-23 - bbb46f3 - lavu 51.18.0
+2011-11-23 - 8e576d5 / bbb46f3 - lavu 51.27.0 / 51.18.0
   Add av_samples_get_buffer_size(), av_samples_fill_arrays(), and
   av_samples_alloc(), to samplefmt.h.
 
-2011-11-23 - 8889cc4 - lavu 51.17.0
+2011-11-23 - 8e576d5 / 8889cc4 - lavu 51.27.0 / 51.17.0
   Add planar sample formats and av_sample_fmt_is_planar() to samplefmt.h.
 
-2011-11-19 - f3a29b7 - lavc 53.21.0
+2011-11-19 - dbb38bc / f3a29b7 - lavc 53.36.0 / 53.21.0
   Move some AVCodecContext fields to a new private struct, AVCodecInternal,
   which is accessed from a new field, AVCodecContext.internal.
   - fields moved:
@@ -89,55 +63,55 @@ API changes, most recent first:
       AVCodecContext.internal_buffer_count --> AVCodecInternal.buffer_count
       AVCodecContext.is_copy               --> AVCodecInternal.is_copy
 
-2011-11-16 - 6270671 - lavu 51.16.0
+2011-11-16 - 8709ba9 / 6270671 - lavu 51.26.0 / 51.16.0
   Add av_timegm()
 
-2011-11-13 - lavf 53.15.0
+2011-11-13 - lavf 53.21.0 / 53.15.0
   New interrupt callback API, allowing per-AVFormatContext/AVIOContext
   interrupt callbacks.
-  6aa0b98 Add AVIOInterruptCB struct and the interrupt_callback field to
+  5f268ca / 6aa0b98 Add AVIOInterruptCB struct and the interrupt_callback field to
           AVFormatContext.
-  1dee0ac Add avio_open2() with additional parameters. Those are
+  5f268ca / 1dee0ac Add avio_open2() with additional parameters. Those are
           an interrupt callback and an options AVDictionary.
           This will allow passing AVOptions to protocols after lavf
           54.0.
 
-2011-11-06 - ba04ecf - lavu 51.14.0
+2011-11-06 - 13b7781 / ba04ecf - lavu 51.24.0 / 51.14.0
   Add av_strcasecmp() and av_strncasecmp() to avstring.h.
 
-2011-11-06 - 07b172f - lavu 51.13.0
+2011-11-06 - 13b7781 / 07b172f - lavu 51.24.0 / 51.13.0
   Add av_toupper()/av_tolower()
 
-2011-11-05 - b6d08f4 - lavf 53.13.0
+2011-11-05 - d8cab5c / b6d08f4 - lavf 53.19.0 / 53.13.0
   Add avformat_network_init()/avformat_network_uninit()
 
-2011-10-27 - 512557b - lavc 53.15.0
+2011-10-27 - 6faf0a2 / 512557b - lavc 53.24.0 / 53.15.0
   Remove avcodec_parse_frame.
   Deprecate AVCodecContext.parse_only and CODEC_CAP_PARSE_ONLY.
 
-2011-10-19 - 569129a - lavf 53.10.0
+2011-10-19 - d049257 / 569129a - lavf 53.17.0 / 53.10.0
   Add avformat_new_stream(). Deprecate av_new_stream().
 
-2011-10-13 - b631fba - lavf 53.9.0
+2011-10-13 - 91eb1b1 / b631fba - lavf 53.16.0 / 53.9.0
   Add AVFMT_NO_BYTE_SEEK AVInputFormat flag.
 
-2011-10-12 - lavu 51.12.0
+2011-10-12 - lavu 51.21.0 / 51.12.0
   AVOptions API rewrite.
 
-  - 145f741 FF_OPT_TYPE* renamed to AV_OPT_TYPE_*
+  - f884ef0 / 145f741 FF_OPT_TYPE* renamed to AV_OPT_TYPE_*
   - new setting/getting functions with slightly different semantics:
-        dac66da av_set_string3 -> av_opt_set
+        f884ef0 / dac66da av_set_string3 -> av_opt_set
                 av_set_double  -> av_opt_set_double
                 av_set_q       -> av_opt_set_q
                 av_set_int     -> av_opt_set_int
 
-        41d9d51 av_get_string  -> av_opt_get
+        f884ef0 / 41d9d51 av_get_string  -> av_opt_get
                 av_get_double  -> av_opt_get_double
                 av_get_q       -> av_opt_get_q
                 av_get_int     -> av_opt_get_int
 
-  - 8c5dcaa trivial rename av_next_option -> av_opt_next
-  - 641c7af new functions - av_opt_child_next, av_opt_child_class_next
+  - f884ef0 / 8c5dcaa trivial rename av_next_option -> av_opt_next
+  - f884ef0 / 641c7af new functions - av_opt_child_next, av_opt_child_class_next
     and av_opt_find2()
 
 2011-09-22 - a70e787 - lavu 51.17.0
@@ -183,31 +157,27 @@ API changes, most recent first:
 2011-08-20 - 69e2c1a - lavu 51.13.0
   Add av_get_media_type_string().
 
-2011-09-03 - fb4ca26 - lavc 53.13.0
+2011-09-03 - 1889c67 / fb4ca26 - lavc 53.13.0
                        lavf 53.11.0
                        lsws  2.1.0
   Add {avcodec,avformat,sws}_get_class().
 
-2011-08-03 - c11fb82 - lavu 51.15.0
+2011-08-03 - 1889c67 / c11fb82 - lavu 51.15.0
   Add AV_OPT_SEARCH_FAKE_OBJ flag for av_opt_find() function.
 
 2011-08-14 - 323b930 - lavu 51.12.0
   Add av_fifo_peek2(), deprecate av_fifo_peek().
 
-2011-08-26 - lavu 51.9.0
-  - add41de..abc78a5 Do not include intfloat_readwrite.h,
-    mathematics.h, rational.h, pixfmt.h, or log.h from avutil.h.
-
-2011-08-16 - 48f9e45 - lavf 53.8.0
+2011-08-16 - 27fbe31 / 48f9e45 - lavf 53.11.0 / 53.8.0
   Add avformat_query_codec().
 
-2011-08-16 - bca06e7 - lavc 53.11.0
+2011-08-16 - 27fbe31 / bca06e7 - lavc 53.11.0
   Add avcodec_get_type().
 
-2011-08-06 - 2f63440 - lavf 53.7.0
+2011-08-06 - 0cb233c / 2f63440 - lavf 53.7.0
   Add error_recognition to AVFormatContext.
 
-2011-08-02 - 9d39cbf - lavc 53.9.1
+2011-08-02 - 1d186e9 / 9d39cbf - lavc 53.9.1
   Add AV_PKT_FLAG_CORRUPT AVPacket flag.
 
 2011-07-16 - b57df29 - lavfi 2.27.0
@@ -218,10 +188,10 @@ API changes, most recent first:
   avfilter_set_common_packing_formats()
   avfilter_all_packing_formats()
 
-2011-07-10 - a67c061 - lavf 53.6.0
+2011-07-10 - 3602ad7 / a67c061 - lavf 53.6.0
   Add avformat_find_stream_info(), deprecate av_find_stream_info().
 
-2011-07-10 - 0b950fe - lavc 53.8.0
+2011-07-10 - 3602ad7 / 0b950fe - lavc 53.8.0
   Add avcodec_open2(), deprecate avcodec_open().
 
 2011-07-01 - b442ca6 - lavf 53.5.0 - avformat.h
@@ -260,35 +230,35 @@ API changes, most recent first:
 2011-06-12 - 6119b23 - lavfi 2.16.0 - avfilter_graph_parse()
   Change avfilter_graph_parse() signature.
 
-2011-06-23 - 67e9ae1 - lavu 51.8.0 - attributes.h
+2011-06-23 - 686959e / 67e9ae1 - lavu 51.10.0 / 51.8.0 - attributes.h
   Add av_printf_format().
 
-2011-06-16 - 05e84c9, 25de595 - lavf 53.2.0 - avformat.h
+2011-06-16 - 2905e3f / 05e84c9, 2905e3f / 25de595 - lavf 53.4.0 / 53.2.0 - avformat.h
   Add avformat_open_input and avformat_write_header().
   Deprecate av_open_input_stream, av_open_input_file,
   AVFormatParameters and av_write_header.
 
-2011-06-16 - 7e83e1c, dc59ec5 - lavu 51.7.0 - opt.h
+2011-06-16 - 2905e3f / 7e83e1c, 2905e3f / dc59ec5 - lavu 51.9.0 / 51.7.0 - opt.h
   Add av_opt_set_dict() and av_opt_find().
   Deprecate av_find_opt().
   Add AV_DICT_APPEND flag.
 
-2011-06-10 - cb7c11c - lavu 51.6.0 - opt.h
+2011-06-10 - 45fb647 / cb7c11c - lavu 51.6.0 - opt.h
   Add av_opt_flag_is_set().
 
 2011-06-10 - c381960 - lavfi 2.15.0 - avfilter_get_audio_buffer_ref_from_arrays
   Add avfilter_get_audio_buffer_ref_from_arrays() to avfilter.h.
 
-2011-06-09 - d9f80ea - lavu 51.8.0 - AVMetadata
+2011-06-09 - f9ecb84 / d9f80ea - lavu 51.8.0 - AVMetadata
   Move AVMetadata from lavf to lavu and rename it to
   AVDictionary -- new installed header dict.h.
   All av_metadata_* functions renamed to av_dict_*.
 
-2011-06-07 - a6703fa - lavu 51.8.0 - av_get_bytes_per_sample()
+2011-06-07 - d552f61 / a6703fa - lavu 51.8.0 - av_get_bytes_per_sample()
   Add av_get_bytes_per_sample() in libavutil/samplefmt.h.
   Deprecate av_get_bits_per_sample_fmt().
 
-2011-06-05 - b39b062 - lavu 51.8.0 - opt.h
+2011-06-05 - f956924 / b39b062 - lavu 51.8.0 - opt.h
   Add av_opt_free convenience function.
 
 2011-06-06 - 95a0242 - lavfi 2.14.0 - AVFilterBufferRefAudioProps
@@ -318,7 +288,7 @@ API changes, most recent first:
   Add av_get_pix_fmt_name() in libavutil/pixdesc.h, and deprecate
   avcodec_get_pix_fmt_name() in libavcodec/avcodec.h in its favor.
 
-2011-05-25 - 30315a8 - lavf 53.3.0 - avformat.h
+2011-05-25 - 39e4206 / 30315a8 - lavf 53.3.0 - avformat.h
   Add fps_probe_size to AVFormatContext.
 
 2011-05-22 - 5ecdfd0 - lavf 53.2.0 - avformat.h
@@ -334,10 +304,10 @@ API changes, most recent first:
 2011-05-14 - 9fdf772 - lavfi 2.6.0 - avcodec.h
   Add avfilter_get_video_buffer_ref_from_frame() to libavfilter/avcodec.h.
 
-2011-05-18 - 64150ff - lavc 53.7.0 - AVCodecContext.request_sample_fmt
+2011-05-18 - 75a37b5 / 64150ff - lavc 53.7.0 - AVCodecContext.request_sample_fmt
   Add request_sample_fmt field to AVCodecContext.
 
-2011-05-10 - 188dea1 - lavc 53.6.0 - avcodec.h
+2011-05-10 - 59eb12f / 188dea1 - lavc 53.6.0 - avcodec.h
   Deprecate AVLPCType and the following fields in
   AVCodecContext: lpc_coeff_precision, prediction_order_method,
   min_partition_order, max_partition_order, lpc_type, lpc_passes.
@@ -367,81 +337,81 @@ API changes, most recent first:
   Add av_dynarray_add function for adding
   an element to a dynamic array.
 
-2011-04-26 - bebe72f - lavu 51.1.0 - avutil.h
+2011-04-26 - d7e5aeb / bebe72f - lavu 51.1.0 - avutil.h
   Add AVPictureType enum and av_get_picture_type_char(), deprecate
   FF_*_TYPE defines and av_get_pict_type_char() defined in
   libavcodec/avcodec.h.
 
-2011-04-26 - 10d3940 - lavfi 2.3.0 - avfilter.h
+2011-04-26 - d7e5aeb / 10d3940 - lavfi 2.3.0 - avfilter.h
   Add pict_type and key_frame fields to AVFilterBufferRefVideo.
 
-2011-04-26 - 7a11c82 - lavfi 2.2.0 - vsrc_buffer
+2011-04-26 - d7e5aeb / 7a11c82 - lavfi 2.2.0 - vsrc_buffer
   Add sample_aspect_ratio fields to vsrc_buffer arguments
 
-2011-04-21 - 94f7451 - lavc 53.1.0 - avcodec.h
+2011-04-21 - 8772156 / 94f7451 - lavc 53.1.0 - avcodec.h
   Add CODEC_CAP_SLICE_THREADS for codecs supporting sliced threading.
 
 2011-04-15 - lavc 52.120.0 - avcodec.h
   AVPacket structure got additional members for passing side information:
-    4de339e introduce side information for AVPacket
-    2d8591c make containers pass palette change in AVPacket
+    c407984 / 4de339e introduce side information for AVPacket
+    c407984 / 2d8591c make containers pass palette change in AVPacket
 
 2011-04-12 - lavf 52.107.0 - avio.h
   Avio cleanup, part II - deprecate the entire URLContext API:
-    175389c add avio_check as a replacement for url_exist
-    ff1ec0c add avio_pause and avio_seek_time as replacements
+    c55780d / 175389c add avio_check as a replacement for url_exist
+    9891004 / ff1ec0c add avio_pause and avio_seek_time as replacements
             for _av_url_read_fseek/fpause
-    cdc6a87 deprecate av_protocol_next(), avio_enum_protocols
+    d4d0932 / cdc6a87 deprecate av_protocol_next(), avio_enum_protocols
             should be used instead.
-    80c6e23 rename url_set_interrupt_cb->avio_set_interrupt_cb.
-    f87b1b3 rename open flags: URL_* -> AVIO_*
-    f8270bb add avio_enum_protocols.
-    5593f03 deprecate URLProtocol.
-    c486dad deprecate URLContext.
-    026e175 deprecate the typedef for URLInterruptCB
-    8e76a19 deprecate av_register_protocol2.
-    b840484 deprecate URL_PROTOCOL_FLAG_NESTED_SCHEME
-    1305d93 deprecate av_url_read_seek
-    fa104e1 deprecate av_url_read_pause
-    727c7aa deprecate url_get_filename().
-    5958df3 deprecate url_max_packet_size().
-    1869ea0 deprecate url_get_file_handle().
-    32a97d4 deprecate url_filesize().
-    e52a914 deprecate url_close().
-    58a48c6 deprecate url_seek().
-    925e908 deprecate url_write().
-    dce3756 deprecate url_read_complete().
-    bc371ac deprecate url_read().
-    0589da0 deprecate url_open().
-    62eaaea deprecate url_connect.
-    5652bb9 deprecate url_alloc.
-    333e894 deprecate url_open_protocol
-    e230705 deprecate url_poll and URLPollEntry
+    c88caa5 / 80c6e23 rename url_set_interrupt_cb->avio_set_interrupt_cb.
+    c88caa5 / f87b1b3 rename open flags: URL_* -> AVIO_*
+    d4d0932 / f8270bb add avio_enum_protocols.
+    d4d0932 / 5593f03 deprecate URLProtocol.
+    d4d0932 / c486dad deprecate URLContext.
+    d4d0932 / 026e175 deprecate the typedef for URLInterruptCB
+    c88caa5 / 8e76a19 deprecate av_register_protocol2.
+    11d7841 / b840484 deprecate URL_PROTOCOL_FLAG_NESTED_SCHEME
+    11d7841 / 1305d93 deprecate av_url_read_seek
+    11d7841 / fa104e1 deprecate av_url_read_pause
+    434f248 / 727c7aa deprecate url_get_filename().
+    434f248 / 5958df3 deprecate url_max_packet_size().
+    434f248 / 1869ea0 deprecate url_get_file_handle().
+    434f248 / 32a97d4 deprecate url_filesize().
+    434f248 / e52a914 deprecate url_close().
+    434f248 / 58a48c6 deprecate url_seek().
+    434f248 / 925e908 deprecate url_write().
+    434f248 / dce3756 deprecate url_read_complete().
+    434f248 / bc371ac deprecate url_read().
+    434f248 / 0589da0 deprecate url_open().
+    434f248 / 62eaaea deprecate url_connect.
+    434f248 / 5652bb9 deprecate url_alloc.
+    434f248 / 333e894 deprecate url_open_protocol
+    434f248 / e230705 deprecate url_poll and URLPollEntry
 
 2011-04-08 - lavf 52.106.0 - avformat.h
   Minor avformat.h cleanup:
-    a9bf9d8 deprecate av_guess_image2_codec
-    c3675df rename avf_sdp_create->av_sdp_create
+    d4d0932 / a9bf9d8 deprecate av_guess_image2_codec
+    d4d0932 / c3675df rename avf_sdp_create->av_sdp_create
 
 2011-04-03 - lavf 52.105.0 - avio.h
   Large-scale renaming/deprecating of AVIOContext-related functions:
-    724f6a0 deprecate url_fdopen
-    403ee83 deprecate url_open_dyn_packet_buf
-    6dc7d80 rename url_close_dyn_buf       -> avio_close_dyn_buf
-    b92c545 rename url_open_dyn_buf        -> avio_open_dyn_buf
-    8978fed introduce an AVIOContext.seekable field as a replacement for
+    2cae980 / 724f6a0 deprecate url_fdopen
+    2cae980 / 403ee83 deprecate url_open_dyn_packet_buf
+    2cae980 / 6dc7d80 rename url_close_dyn_buf       -> avio_close_dyn_buf
+    2cae980 / b92c545 rename url_open_dyn_buf        -> avio_open_dyn_buf
+    2cae980 / 8978fed introduce an AVIOContext.seekable field as a replacement for
             AVIOContext.is_streamed and url_is_streamed()
-    b64030f deprecate get_checksum()
-    4c4427a deprecate init_checksum()
-    4ec153b deprecate udp_set_remote_url/get_local_port
-    933e90a deprecate av_url_read_fseek/fpause
-    8d9769a deprecate url_fileno
-    b7f2fdd rename put_flush_packet -> avio_flush
-    35f1023 deprecate url_close_buf
-    83fddae deprecate url_open_buf
-    d9d86e0 rename url_fprintf -> avio_printf
-    59f65d9 deprecate url_setbufsize
-    3e68b3b deprecate url_ferror
+    1caa412 / b64030f deprecate get_checksum()
+    1caa412 / 4c4427a deprecate init_checksum()
+    2fd41c9 / 4ec153b deprecate udp_set_remote_url/get_local_port
+    4fa0e24 / 933e90a deprecate av_url_read_fseek/fpause
+    4fa0e24 / 8d9769a deprecate url_fileno
+    0fecf26 / b7f2fdd rename put_flush_packet -> avio_flush
+    0fecf26 / 35f1023 deprecate url_close_buf
+    0fecf26 / 83fddae deprecate url_open_buf
+    0fecf26 / d9d86e0 rename url_fprintf -> avio_printf
+    0fecf26 / 59f65d9 deprecate url_setbufsize
+    6947b0c / 3e68b3b deprecate url_ferror
     e8bb2e2 deprecate url_fget_max_packet_size
     76aa876 rename url_fsize -> avio_size
     e519753 deprecate url_fgetc
@@ -462,7 +432,7 @@ API changes, most recent first:
     b3db9ce deprecate get_partial_buffer
     8d9ac96 rename av_alloc_put_byte -> avio_alloc_context
 
-2011-03-25 - 34b47d7 - lavc 52.115.0 - AVCodecContext.audio_service_type
+2011-03-25 - 27ef7b1 / 34b47d7 - lavc 52.115.0 - AVCodecContext.audio_service_type
   Add audio_service_type field to AVCodecContext.
 
 2011-03-17 - e309fdc - lavu 50.40.0 - pixfmt.h
@@ -500,11 +470,11 @@ API changes, most recent first:
 2011-02-10 - 12c14cd - lavf 52.99.0 - AVStream.disposition
   Add AV_DISPOSITION_HEARING_IMPAIRED and AV_DISPOSITION_VISUAL_IMPAIRED.
 
-2011-02-09 - 5592734 - lavc 52.112.0 - avcodec_thread_init()
+2011-02-09 - c0b102c - lavc 52.112.0 - avcodec_thread_init()
   Deprecate avcodec_thread_init()/avcodec_thread_free() use; instead
   set thread_count before calling avcodec_open.
 
-2011-02-09 - 778b08a - lavc 52.111.0 - threading API
+2011-02-09 - 37b00b4 - lavc 52.111.0 - threading API
   Add CODEC_CAP_FRAME_THREADS with new restrictions on get_buffer()/
   release_buffer()/draw_horiz_band() callbacks for appropriate codecs.
   Add thread_type and active_thread_type fields to AVCodecContext.

doc/Makefile

@@ -9,24 +9,13 @@ HTMLPAGES   = $(PROGS-yes:%=doc/%.html)                                 \
               doc/libavfilter.html                                      \
               doc/platform.html                                         \
 
-TXTPAGES    = doc/fate.txt                                              \
-
-
 DOCS = $(HTMLPAGES) $(MANPAGES) $(PODPAGES)
-ifdef HAVE_MAKEINFO
-DOCS += $(TXTPAGES)
-endif
 
 all-$(CONFIG_DOC): documentation
 
 documentation: $(DOCS)
 
-TEXIDEP = awk '/^@(verbatim)?include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)
-
-doc/%.txt: TAG = TXT
-doc/%.txt: doc/%.texi
-	$(Q)$(TEXIDEP)
-	$(M)makeinfo --force --no-headers -o $@ $< 2>/dev/null
+TEXIDEP = awk '/^@include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)
 
 doc/%.html: TAG = HTML
 doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init
@@ -57,7 +46,7 @@ uninstall-man:
 	$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
 
 clean::
-	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 $(CLEANSUFFIXES:%=doc/%)
+	$(RM) doc/*.html doc/*.pod doc/*.1 $(CLEANSUFFIXES:%=doc/%)
 
 -include $(wildcard $(DOCS:%=%.d))
 

doc/RELEASE_NOTES

@@ -1,12 +1,11 @@
 Release Notes
 =============
 
-* 0.10 "Freedom"  January, 2012
+* 0.9 "Harmony"  December, 2011
 
 
 General notes
 -------------
-This release is binary compatible with 0.8 and 0.9.
 
 See the Changelog file for a list of significant changes. Note, there
 are many more new features and bugfixes than whats listed there.
@@ -16,34 +15,3 @@ accepted. If you are experiencing issues with any formally released version of
 FFmpeg, please try git master to check if the issue still exists. If it does,
 make your report against the development code following the usual bug reporting
 guidelines.
-
-
-API changes
------------
-
-A number of additional APIs have been introduced and some existing
-functions have been deprecated and are scheduled for removal in the next
-release. Significant API changes include:
-
-* new audio decoding API which decodes from an AVPacket to an AVFrame and
-is able to use AVCodecContext.get_buffer() in the similar way as video decoding.
-
-* new audio encoding API which encodes from an AVFrame to an AVPacket, thus
-allowing it to properly output timing information and side data.
-
-Please see the git history and the file doc/APIchanges for details.
-
-
-Other notable changes
----------------------
-
-Libavcodec and libavformat built as shared libraries now hide non-public
-symbols. This will break applications using those symbols. Possible solutions
-are, in order of preference:
-1) Try finding a way of accomplishing the same with public API.
-2) If there is no corresponding public API, but you think there should be,
-post a request on the developer mailing list or IRC channel.
-3) Finally if your program needs access to FFmpeg / libavcodec / libavformat
-internals for some special reason then the best solution is to link statically.
-
-Please see the Changelog file and git history for a more detailed list of changes.

doc/avtools-common-opts.texi

@@ -11,7 +11,6 @@ corresponding value to true. They can be set to false by prefixing
 with "no" the option name, for example using "-nofoo" in the
 command line will set to false the boolean option with name "foo".
 
-@anchor{Stream specifiers}
 @section Stream specifiers
 Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers
 are used to precisely specify which stream(s) does a given option belong to.
@@ -119,8 +118,8 @@ Set the logging level used by the library.
 By default the program logs to stderr, if coloring is supported by the
 terminal, colors are used to mark errors and warnings. Log coloring
 can be disabled setting the environment variable
-@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
-the environment variable @env{AV_LOG_FORCE_COLOR}.
+@env{FFMPEG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
+the environment variable @env{FFMPEG_FORCE_COLOR}.
 The use of the environment variable @env{NO_COLOR} is deprecated and
 will be dropped in a following FFmpeg version.
 

doc/bitstream_filters.texi

@@ -23,20 +23,6 @@ Below is a description of the currently available bitstream filters.
 
 @section h264_mp4toannexb
 
-Convert an H.264 bitstream from length prefixed mode to start code
-prefixed mode (as defined in the Annex B of the ITU-T H.264
-specification).
-
-This is required by some streaming formats, typically the MPEG-2
-transport stream format ("mpegts").
-
-For example to remux an MP4 file containing an H.264 stream to mpegts
-format with @command{ffmpeg}, you can use the command:
-
-@example
-ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
-@end example
-
 @section imx_dump_header
 
 @section mjpeg2jpeg

doc/decoders.texi

@@ -48,16 +48,3 @@ top-field-first is assumed
 @end table
 
 @c man end VIDEO DECODERS
-
-@chapter Audio Decoders
-@c man begin AUDIO DECODERS
-
-@section ffwavesynth
-
-Internal wave synthetizer.
-
-This decoder generates wave patterns according to predefined sequences. Its
-use is purely internal and the format of the data it accepts is not publicly
-documented.
-
-@c man end AUDIO DECODERS

doc/demuxers.texi

@@ -75,34 +75,4 @@ 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 sbg
-
-SBaGen script demuxer.
-
-This demuxer reads the script language used by SBaGen
-@url{http://uazu.net/sbagen/} to generate binaural beats sessions. A SBG
-script looks like that:
-@example
--SE
-a: 300-2.5/3 440+4.5/0
-b: 300-2.5/0 440+4.5/3
-off: -
-NOW      == a
-+0:07:00 == b
-+0:14:00 == a
-+0:21:00 == b
-+0:30:00    off
-@end example
-
-A SBG script can mix absolute and relative timestamps. If the script uses
-either only absolute timestamps (including the script start time) or only
-relative ones, then its layout is fixed, and the conversion is
-straightforward. On the other hand, if the script mixes both kind of
-timestamps, then the @var{NOW} reference for relative timestamps will be
-taken from the current time of day at the time the script is read, and the
-script layout will be frozen according to that reference. That means that if
-the script is directly played, the actual times will match the absolute
-timestamps up to the sound controller's clock accuracy, but if the user
-somehow pauses the playback or seeks, all times will be shifted accordingly.
-
 @c man end INPUT DEVICES

doc/developer.texi

@@ -345,7 +345,7 @@ for us and greatly increases your chances of getting your patch applied.
 Use the patcheck tool of FFmpeg to check your patch.
 The tool is located in the tools directory.
 
-Run the @ref{Regression tests} before submitting a patch in order to verify
+Run the @ref{Regression Tests} before submitting a patch in order to verify
 it does not cause unexpected problems.
 
 Patches should be posted as base64 encoded attachments (or any other
@@ -508,13 +508,12 @@ not related to the comments received during review. Such patches will
 be rejected. Instead, submit significant changes or new features as
 separate patches.
 
-@anchor{Regression tests}
 @section Regression tests
 
 Before submitting a patch (or committing to the repository), you should at least
 test that you did not break anything.
 
-Running 'make fate' accomplishes this, please see @url{fate.html} for details.
+Running 'make fate' accomplishes this, please see @file{doc/fate.txt} for details.
 
 [Of course, some patches may change the results of the regression tests. In
 this case, the reference results of the regression tests shall be modified

doc/doxy/footer.html

@@ -1,10 +0,0 @@
- </div>
-
- <div id="footer">
-   Generated on $datetime for $projectname by&#160;<a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion
- </div>
-
-</div>
-
-</body>
-</html>

doc/doxy/header.html

@@ -1,14 +0,0 @@
-<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
-<html xmlns="http://www.w3.org/1999/xhtml">
-<head>
-<meta http-equiv="Content-Type" content="text/xhtml;charset=UTF-8"/>
-<meta http-equiv="X-UA-Compatible" content="IE=9"/>
-<!--BEGIN PROJECT_NAME--><title>$projectname: $title</title><!--END PROJECT_NAME-->
-<!--BEGIN !PROJECT_NAME--><title>$title</title><!--END !PROJECT_NAME-->
-<link href="$relpath$doxy_stylesheet.css" rel="stylesheet" type="text/css" />
-</head>
-
-<div id="container">
-
-<div id="body">
- <div>

doc/encoders.texi

@@ -577,7 +577,7 @@ Allow to set any x264 option, see x264 --fullhelp for a list.
 ":".
 @end table
 
-For example to specify libx264 encoding options with @command{ffmpeg}:
+For example to specify libx264 encoding options with @file{ffmpeg}:
 @example
 ffmpeg -i foo.mpg -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv
 @end example

doc/eval.texi

@@ -98,14 +98,6 @@ point (@var{x}, @var{y}) from the origin.
 @item gcd(x, y)
 Return the greatest common divisor of @var{x} and @var{y}. If both @var{x} and
 @var{y} are 0 or either or both are less than zero then behavior is undefined.
-
-@item if(x, y)
-Evaluate @var{x}, and if the result is non-zero return the result of
-the evaluation of @var{y}, return 0 otherwise.
-
-@item ifnot(x, y)
-Evaluate @var{x}, and if the result is zero return the result of the
-evaluation of @var{y}, return 0 otherwise.
 @end table
 
 The following constants are available:
@@ -118,20 +110,19 @@ exp(1) (Euler's number), approximately 2.718
 golden ratio (1+sqrt(5))/2, approximately 1.618
 @end table
 
-Assuming that an expression is considered "true" if it has a non-zero
-value, note that:
+Note that:
 
 @code{*} works like AND
 
 @code{+} works like OR
 
-and the construct:
+thus
 @example
 if A then B else C
 @end example
 is equivalent to
 @example
-if(A,B) + ifnot(A,C)
+A*B + not(A)*C
 @end example
 
 In your C code, you can extend the list of unary and binary functions,

doc/faq.texi

@@ -272,44 +272,6 @@ ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
 rm temp[12].[av] all.[av]
 @end example
 
-@section -profile option fails when encoding H.264 video with AAC audio
-
-@command{ffmpeg} prints an error like
-
-@example
-Undefined constant or missing '(' in 'baseline'
-Unable to parse option value "baseline"
-Error setting option profile to value baseline.
-@end example
-
-Short answer: write @option{-profile:v} instead of @option{-profile}.
-
-Long answer: this happens because the @option{-profile} option can apply to both
-video and audio.  Specifically the AAC encoder also defines some profiles, none
-of which are named @var{baseline}.
-
-The solution is to apply the @option{-profile} option to the video stream only
-by using @url{http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1, Stream specifiers}.
-Appending @code{:v} to it will do exactly that.
-
-@section Using @option{-f lavfi}, audio becomes mono for no apparent reason.
-
-Use @option{-dumpgraph -} to find out exactly where the channel layout is
-lost.
-
-Most likely, it is through @code{auto-inserted aconvert}. Try to understand
-why the converting filter was needed at that place.
-
-Just before the output is a likely place, as @option{-f lavfi} currently
-only support packed S16.
-
-Then insert the correct @code{aconvert} explicitly in the filter graph,
-specifying the exact format.
-
-@example
-aconvert=s16:stereo:packed
-@end example
-
 @chapter Development
 
 @section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
@@ -415,16 +377,4 @@ wrong if it is larger than the average!
 For example, if you have mixed 25 and 30 fps content, then r_frame_rate
 will be 150.
 
-@section Why is @code{make fate} not running all tests?
-
-Make sure you have the fate-suite samples and the @code{SAMPLES} Make variable
-or @code{FATE_SAMPLES} environment variable or the @code{--samples}
-@command{configure} option is set to the right path.
-
-@section Why is @code{make fate} not finding the samples?
-
-Do you happen to have a @code{~} character in the samples path to indicate a
-home directory? The value is used in ways where the shell cannot expand it,
-causing FATE to not find files. Just replace @code{~} by the full path.
-
 @bye

doc/fate.texi

@@ -5,170 +5,131 @@
 @center @titlefont{FATE Automated Testing Environment}
 @end titlepage
 
-@node Top
 @top
 
 @contents
 
 @chapter Introduction
 
-  FATE is an extended regression suite on the client-side and a means
-for results aggregation and presentation on the server-side.
+FATE provides a regression testsuite embedded within the FFmpeg build system.
+It can be run locally and optionally configured to send reports to a web
+aggregator and viewer @url{http://fate.ffmpeg.org}.
 
-  The first part of this document explains how you can use FATE from
-your FFmpeg source directory to test your ffmpeg binary. The second
-part describes how you can run FATE to submit the results to FFmpeg's
-FATE server.
+It is advised to run FATE before submitting patches to the current codebase
+and provide new tests when submitting patches to add additional features.
 
-  In any way you can have a look at the publicly viewable FATE results
-by visiting this website:
+@chapter Running FATE
 
-  @url{http://fate.ffmpeg.org/}
+@section Samples and References
+In order to run, FATE needs a large amount of data (samples and references)
+that is provided separately from the actual source distribution.
 
-  This is especially recommended for all people contributing source
-code to FFmpeg, as it can be seen if some test on some platform broke
-with there recent contribution. This usually happens on the platforms
-the developers could not test on.
+To inform the build system about the testsuite location, pass
+@option{--samples=<path to the samples>} to @command{configure} or set the
+@var{SAMPLES} Make variable or the @var{FATE_SAMPLES} environment variable
+to a suitable value.
 
-  The second part of this document describes how you can run FATE to
-submit your results to FFmpeg's FATE server. If you want to submit your
-results be sure to check that your combination of CPU, OS and compiler
-is not already listed on the above mentioned website.
-
-  In the third part you can find a comprehensive listing of FATE makefile
-targets and variables.
-
-
-@chapter Using FATE from your FFmpeg source directory
-
-  If you want to run FATE on your machine you need to have the samples
-in place. You can get the samples via the build target fate-rsync.
-Use this command from the top-level source directory:
+The dataset is available through @command{rsync}, is possible to fetch
+the current sample using the straight rsync command or through a specific
+@ref{Makefile target}.
 
 @example
-make fate-rsync SAMPLES=fate-suite/
-make fate       SAMPLES=fate-suite/
+# rsync -aL rsync://fate.ffmpeg.org/fate-suite/ fate-suite
 @end example
 
-  The above commands set the samples location by passing a makefile
-variable via command line. It is also possible to set the samples
-location at source configuration time by invoking configure with
-`--samples=<path to the samples directory>'. Afterwards you can
-invoke the makefile targets without setting the SAMPLES makefile
-variable. This is illustrated by the following commands:
-
 @example
-./configure --samples=fate-suite/
-make fate-rsync
-make fate
+# make fate-rsync SAMPLES=fate-suite
 @end example
 
-  Yet another way to tell FATE about the location of the sample
-directory is by making sure the environment variable FATE_SAMPLES
-contains the path to your samples directory. This can be achieved
-by e.g. putting that variable in your shell profile or by setting
-it in your interactive session.
 
-@example
-FATE_SAMPLES=fate-suite/ make fate
-@end example
-
-@float NOTE
-Do not put a '~' character in the samples path to indicate a home
-directory. Because of shell nuances, this will cause FATE to fail.
-@end float
-
-
-@chapter Submitting the results to the FFmpeg result aggregation server
-
-  To submit your results to the server you should run fate through the
-shell script tests/fate.sh from the FFmpeg sources. This script needs
-to be invoked with a configuration file as its first argument.
-
-@example
-tests/fate.sh /path/to/fate_config
-@end example
-
-  A configuration file template with comments describing the individual
-configuration variables can be found at @file{tests/fate_config.sh.template}.
-
-@ifhtml
-  The mentioned configuration template is also available here:
-@verbatiminclude ../tests/fate_config.sh.template
-@end ifhtml
-
-  Create a configuration that suits your needs, based on the configuration
-template. The `slot' configuration variable can be any string that is not
-yet used, but it is suggested that you name it adhering to the following
-pattern <arch>-<os>-<compiler>-<compiler version>. The configuration file
-itself will be sourced in a shell script, therefore all shell features may
-be used. This enables you to setup the environment as you need it for your
-build.
-
-  For your first test runs the `fate_recv' variable should be empty or
-commented out. This will run everything as normal except that it will omit
-the submission of the results to the server. The following files should be
-present in $workdir as specified in the configuration file:
-
-@itemize
-    @item configure.log
-    @item compile.log
-    @item test.log
-    @item report
-    @item version
-@end itemize
-
-  When you have everything working properly you can create an SSH key and
-send its public part to the FATE server administrator.
-
-  Configure your SSH client to use public key authentication with that key
-when connecting to the FATE server. Also do not forget to check the identity
-of the server and to accept its host key. This can usually be achieved by
-running your SSH client manually and killing it after you accepted the key.
-The FATE server's fingerprint is:
-
-  b1:31:c8:79:3f:04:1d:f8:f2:23:26:5a:fd:55:fa:92
-
-  The only thing left is to automate the execution of the fate.sh script and
-the synchronisation of the samples directory.
-
-
-@chapter FATE makefile targets and variables
-
-@section Makefile targets
+@chapter Manual Run
+FATE regression test can be run through @command{make}.
+Specific Makefile targets and Makefile variables are available:
 
+@anchor{Makefile target}
+@section FATE Makefile targets
 @table @option
-@item fate-rsync
-    Download/synchronize sample files to the configured samples directory.
-
 @item fate-list
-    Will list all fate/regression test targets.
-
+List all fate/regression test targets.
+@item fate-rsync
+Shortcut to download the fate test samples to the specified testsuite location.
 @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
-
+@section Fate Makefile variables
 @table @option
 @item V
-    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
-        @item 2: show everything
-    @end itemize
-
+Verbosity level, can be set to 0, 1 or 2.
+@table @option
+    @item 0
+    show just the test arguments
+    @item 1
+    show just the command used in the test
+    @item 2
+    show everything
+@end table
 @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.
 @end table
 
-Example:
 @example
-make V=1 SAMPLES=/var/fate/samples THREADS=2 fate
+    make V=1 SAMPLES=/var/fate/samples THREADS=2 fate
 @end example
+
+@chapter Automated Tests
+In order to automatically testing specific configurations, e.g. multiple
+compilers, @command{tests/fate.sh} is provided.
+
+This shell script builds FFmpeg, runs the regression tests and prepares a
+report that can be sent to @url{fate.ffmpeg.org} or directly examined locally.
+
+@section Testing Profiles
+The configuration file passed to @command{fate.sh} is shell scripts as well.
+
+It must provide at least a @var{slot} identifier, the @var{repo} from
+which fetch the sources, the @var{samples} directory, a @var{workdir} with
+enough space to build and run all the tests.
+Optional submit command @var{fate_recv} and a @var{comment} to describe
+the testing profile are available.
+
+Additional optional parameter to tune the FFmpeg building and reporting process
+can be passed.
+
+@example
+slot=                                   # some unique identifier
+repo=git://source.ffmpeg.org/ffmpeg.git # the source repository
+samples=/path/to/fate/samples
+workdir=                                # directory in which to do all the work
+fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report
+comment=                                # optional description
+
+# the following are optional and map to configure options
+arch=
+cpu=
+cross_prefix=
+cc=
+target_os=
+sysroot=
+target_exec=
+target_path=
+extra_cflags=
+extra_ldflags=
+extra_libs=
+extra_conf=     # extra configure options not covered above
+
+#make=          # name of GNU make if not 'make'
+makeopts=       # extra options passed to 'make'
+#tar=           # command to create a tar archive from its arguments on
+                # stdout, defaults to 'tar c'
+@end example
+
+@section Submitting Reports
+In order to send reports you need to create an @command{ssh} key and send it
+to the fate server administrator.
+The current server fingerprint is @var{b1:31:c8:79:3f:04:1d:f8:f2:23:26:5a:fd:55:fa:92}
+

doc/fate.txt

@@ -0,0 +1,139 @@
+FATE Automated Testing Environment
+==================================
+
+  FATE is an extended regression suite on the client-side and a means
+for results aggregation and presentation on the server-side.
+
+  The first part of this document explains how you can use FATE from
+your FFmpeg source directory to test your ffmpeg binary. The second
+part describes how you can run FATE to submit the results to FFmpeg's
+FATE server.
+
+  In any way you can have a look at the publicly viewable FATE results
+by visiting this website:
+
+  http://fate.ffmpeg.org/
+
+  This is especially recommended for all people contributing source
+code to FFmpeg, as it can be seen if some test on some platform broke
+with there recent contribution. This usually happens on the platforms
+the developers could not test on.
+
+  The second part of this document describes how you can run FATE to
+submit your results to FFmpeg's FATE server. If you want to submit your
+results be sure to check that your combination of CPU, OS and compiler
+is not already listed on the above mentioned website.
+
+  In the third part you can find a comprehensive listing of FATE makefile
+targets and variables.
+
+
+1. Using FATE from your FFmpeg source directory
+-----------------------------------------------
+
+  If you want to run FATE on your machine you need to have the samples
+in place. You can get the samples via the build target fate-rsync.
+Use this command from the top-level source directory:
+
+# make fate-rsync SAMPLES=fate-suite/
+# make fate       SAMPLES=fate-suite/
+
+  The above commands set the samples location by passing a makefile
+variable via command line. It is also possible to set the samples
+location at source configuration time by invoking configure with
+`--samples=<path to the samples directory>'. Afterwards you can
+invoke the makefile targets without setting the SAMPLES makefile
+variable. This is illustrated by the following commands:
+
+# ./configure --samples=fate-suite/
+# make fate-rsync
+# make fate
+
+  Yet another way to tell FATE about the location of the sample
+directory is by making sure the environment variable FATE_SAMPLES
+contains the path to your samples directory. This can be achieved
+by e.g. putting that variable in your shell profile or by setting
+it in your interactive session.
+
+# FATE_SAMPLES=fate-suite/ make fate
+
+NOTE:
+Do not put a '~' character in the samples path to indicate a home
+directory. Because of shell nuances, this will cause FATE to fail.
+
+
+2. Submitting the results to the FFmpeg result aggregation server
+-----------------------------------------------------------------
+
+  To submit your results to the server you should run fate through the
+shell script tests/fate.sh from the FFmpeg sources. This script needs
+to be invoked with a configuration file as its first argument.
+
+# tests/fate.sh /path/to/fate_config
+
+  A configuration file template with comments describing the individual
+configuration variables can be found at tests/fate_config.sh.template .
+
+  Create a configuration that suits your needs, based on the configuration
+template. The `slot' configuration variable can be any string that is not
+yet used, but it is suggested that you name it adhering to the following
+pattern <arch>-<os>-<compiler>-<compiler version>. The configuration file
+itself will be sourced in a shell script, therefore all shell features may
+be used. This enables you to setup the environment as you need it for your
+build.
+
+  For your first test runs the `fate_recv' variable should be empty or
+commented out. This will run everything as normal except that it will omit
+the submission of the results to the server. The following files should be
+present in $workdir as specified in the configuration file:
+
+  - configure.log
+  - compile.log
+  - test.log
+  - report
+  - version
+
+  When you have everything working properly you can create an SSH key and
+send its public part to the FATE server administrator.
+
+  Configure your SSH client to use public key authentication with that key
+when connecting to the FATE server. Also do not forget to check the identity
+of the server and to accept its host key. This can usually be achieved by
+running your SSH client manually and killing it after you accepted the key.
+The FATE server's fingerprint is:
+
+  b1:31:c8:79:3f:04:1d:f8:f2:23:26:5a:fd:55:fa:92
+
+  The only thing left is to automate the execution of the fate.sh script and
+the synchronisation of the samples directory.
+
+
+3. FATE makefile targets and variables
+--------------------------------------
+
+FATE Makefile targets:
+
+fate-list
+    Will list all fate/regression test targets.
+
+fate
+    Run the FATE test suite (requires the fate-suite dataset).
+
+FATE Makefile variables:
+
+V
+    Verbosity level, can be set to 0, 1 or 2.
+    * 0: show just the test arguments
+    * 1: show just the command used in the test
+    * 2: show everything
+
+SAMPLES
+    Specify or override the path to the FATE samples at make time, it has a
+    meaning only while running the regression tests.
+
+THREADS
+    Specify how many threads to use while running regression tests, it is
+    quite useful to detect thread-related regressions.
+
+Example:
+    make V=1 SAMPLES=/var/fate/samples THREADS=2 fate

doc/ffmpeg.texi

@@ -29,7 +29,7 @@ rates and resize video on the fly with a high quality polyphase filter.
 ffmpeg reads from an arbitrary number of input "files" (which can be regular
 files, pipes, network streams, grabbing devices, etc.), specified by the
 @code{-i} option, and writes to an arbitrary number of output "files", which are
-specified by a plain output filename. Anything found on the command line which
+specified by a plain output filename. Anything found on the commandline which
 cannot be interpreted as an option is considered to be an output filename.
 
 Each input or output file can in principle contain any number of streams of
@@ -187,9 +187,9 @@ For example, for setting the title in the output file:
 ffmpeg -i in.avi -metadata title="my title" out.flv
 @end example
 
-To set the language of the first audio stream:
+To set the language of the second stream:
 @example
-ffmpeg -i INPUT -metadata:s:a:1 language=eng OUTPUT
+ffmpeg -i INPUT -metadata:s:1 language=eng OUTPUT
 @end example
 
 @item -target @var{type} (@emph{output})
@@ -407,6 +407,10 @@ prefix is ``ffmpeg2pass''. The complete file name will be
 @file{PREFIX-N.log}, where N is a number specific to the output
 stream
 
+Note that this option is overwritten by a local option of the same name
+when using @code{-vcodec libx264}. That option maps to the x264 option stats
+which has a different syntax.
+
 @item -vlang @var{code}
 Set the ISO 639 language code (3 letters) of the current video stream.
 
@@ -825,36 +829,18 @@ The following example split the channels of a stereo input into streams:
 ffmpeg -i stereo.wav -map 0:0 -map 0:0 -map_channel 0.0.0:0.0 -map_channel 0.0.1:0.1 -y out.ogg
 @end example
 
-Note that currently each output stream can only contain channels from a single
-input stream; you can't for example use "-map_channel" to pick multiple input
-audio channels contained in different streams (from the same or different files)
-and merge them into a single output stream. It is therefore not currently
-possible, for example, to turn two separate mono streams into a single stereo
-stream. However spliting a stereo stream into two single channel mono streams
-is possible.
+Note that "-map_channel" is currently limited to the scope of one input for
+each output; you can't for example use it to pick multiple input audio files
+and mix them into one single output.
 
-@item -map_metadata[:@var{metadata_spec_out}] @var{infile}[:@var{metadata_spec_in}] (@emph{output,per-metadata})
+@item -map_metadata[:@var{metadata_type}][:@var{index}] @var{infile}[:@var{metadata_type}][:@var{index}] (@emph{output,per-metadata})
 Set metadata information of the next output file from @var{infile}. Note that
 those are file indices (zero-based), not filenames.
-Optional @var{metadata_spec_in/out} parameters specify, which metadata to copy.
-A metadata specifier can have the following forms:
-@table @option
-@item @var{g}
-global metadata, i.e. metadata that applies to the whole file
-
-@item @var{s}[:@var{stream_spec}]
-per-stream metadata. @var{stream_spec} is a stream specifier as described
-in the @ref{Stream specifiers} chapter. In an input metadata specifier, the first
-matching stream is copied from. In an output metadata specifier, all matching
-streams are copied to.
-
-@item @var{c}:@var{chapter_index}
-per-chapter metadata. @var{chapter_index} is the zero-based chapter index.
-
-@item @var{p}:@var{program_index}
-per-program metadata. @var{program_index} is the zero-based program index.
-@end table
-If metadata specifier is omitted, it defaults to global.
+Optional @var{metadata_type} parameters specify, which metadata to copy - (g)lobal
+(i.e. metadata that applies to the whole file), per-(s)tream, per-(c)hapter or
+per-(p)rogram. All metadata specifiers other than global must be followed by the
+stream/chapter/program index. If metadata specifier is omitted, it defaults to
+global.
 
 By default, global metadata is copied from the first input file,
 per-stream and per-chapter metadata is copied along with streams/chapters. These
@@ -866,14 +852,6 @@ of the output file:
 @example
 ffmpeg -i in.ogg -map_metadata 0:s:0 out.mp3
 @end example
-
-To do the reverse, i.e. copy global metadata to all audio streams:
-@example
-ffmpeg -i in.mkv -map_metadata:s:a 0:g out.mkv
-@end example
-Note that simple @code{0} would work as well in this example, since global
-metadata is assumed by default.
-
 @item -map_chapters @var{input_file_index} (@emph{output})
 Copy chapters from input file with index @var{input_file_index} to the next
 output file. If no chapter mapping is specified, then chapters are copied from
@@ -941,15 +919,15 @@ Thread count.
 Video sync method.
 
 @table @option
-@item 0, passthrough
+@item 0
 Each frame is passed with its timestamp from the demuxer to the muxer.
-@item 1, cfr
+@item 1
 Frames will be duplicated and dropped to achieve exactly the requested
 constant framerate.
-@item 2, vfr
+@item 2
 Frames are passed through with their timestamp or dropped so as to
 prevent 2 frames from having the same timestamp.
-@item -1, auto
+@item -1
 Chooses between 1 and 2 depending on muxer capabilities. This is the
 default method.
 @end table
@@ -1000,13 +978,6 @@ ffmpeg -i file.mov -an -vn -sbsf mov2textsub -c:s copy -f rawvideo sub.txt
 
 @item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{per-stream})
 Force a tag/fourcc for matching streams.
-
-@item -timecode @var{hh}:@var{mm}:@var{ss}SEP@var{ff}
-Specify Timecode for writing. @var{SEP} is ':' for non drop timecode and ';'
-(or '.') for drop.
-@example
-ffmpeg -i input.mpg -timecode 01:02:03.04 -r 30000/1001 -s ntsc output.mpg
-@end example
 @end table
 
 @section Preset files

doc/ffprobe.texi

@@ -94,11 +94,6 @@ For example for printing the output in JSON format, specify:
 For more details on the available output printing formats, see the
 Writers section below.
 
-@item -show_error
-Show information about the error found when trying to probe the input.
-
-The error information is printed within a section with name "ERROR".
-
 @item -show_format
 Show information about the container format of the input multimedia
 stream.
@@ -113,13 +108,6 @@ stream.
 The information for each single packet is printed within a dedicated
 section with name "PACKET".
 
-@item -show_frames
-Show information about each frame contained in the input multimedia
-stream.
-
-The information for each single frame is printed within a dedicated
-section with name "FRAME".
-
 @item -show_streams
 Show information about each media stream contained in the input
 multimedia stream.
@@ -127,29 +115,6 @@ multimedia stream.
 Each media stream information is printed within a dedicated section
 with name "STREAM".
 
-@item -show_private_data, -private
-Show private data, that is data depending on the format of the
-particular shown element.
-This option is enabled by default, but you may need to disable it
-for specific uses, for example when creating XSD-compliant XML output.
-
-@item -show_program_version
-Show information related to program version.
-
-Version information is printed within a section with name
-"PROGRAM_VERSION".
-
-@item -show_library_versions
-Show information related to library versions.
-
-Version information for each library is printed within a section with
-name "LIBRARY_VERSION".
-
-@item -show_versions
-Show information related to program and library versions. This is the
-equivalent of setting both @option{-show_program_version} and
-@option{-show_library_versions} options.
-
 @item -i @var{input_file}
 Read @var{input_file}.
 
@@ -159,7 +124,7 @@ Read @var{input_file}.
 @chapter Writers
 @c man begin WRITERS
 
-A writer defines the output format adopted by @command{ffprobe}, and will be
+A writer defines the output format adopted by @file{ffprobe}, and will be
 used for printing all the parts of the output.
 
 A writer may accept one or more arguments, which specify the options to
@@ -245,70 +210,8 @@ JSON based format.
 
 Each section is printed using JSON notation.
 
-This writer accepts options as a list of @var{key}=@var{value} pairs,
-separated by ":".
-
-The description of the accepted options follows.
-
-@table @option
-
-@item compact, c
-If set to 1 enable compact output, that is each section will be
-printed on a single line. Default value is 0.
-@end table
-
 For more information about JSON, see @url{http://www.json.org/}.
 
-@section xml
-XML based format.
-
-The XML output is described in the XML schema description file
-@file{ffprobe.xsd} installed in the FFmpeg datadir.
-
-Note that the output issued will be compliant to the
-@file{ffprobe.xsd} schema only when no special global output options
-(@option{unit}, @option{prefix}, @option{byte_binary_prefix},
-@option{sexagesimal} etc.) are specified.
-
-This writer accepts options as a list of @var{key}=@var{value} pairs,
-separated by ":".
-
-The description of the accepted options follows.
-
-@table @option
-
-@item fully_qualified, q
-If set to 1 specify if the output should be fully qualified. Default
-value is 0.
-This is required for generating an XML file which can be validated
-through an XSD file.
-
-@item xsd_compliant, x
-If set to 1 perform more checks for ensuring that the output is XSD
-compliant. Default value is 0.
-This option automatically sets @option{fully_qualified} to 1.
-@end table
-
-For more information about the XML format, see
-@url{http://www.w3.org/XML/}.
-
-@chapter Timecode
-
-@command{ffprobe} supports Timecode extraction:
-
-@itemize
-
-@item MPEG1/2 timecode is extracted from the GOP, and is available in the video
-stream details (@option{-show_streams}, see @var{timecode}).
-
-@item MOV timecode is extracted from tmcd track, so is available in the tmcd
-stream metadata (@option{-show_streams}, see @var{TAG:timecode}).
-
-@item DV and GXF timecodes are available in format metadata
-(@option{-show_format}, see @var{TAG:timecode}).
-
-@end itemize
-
 @c man end WRITERS
 
 @include decoders.texi

doc/ffprobe.xsd

@@ -1,164 +0,0 @@
-<?xml version="1.0" encoding="UTF-8"?>
-
-<xsd:schema xmlns:xsd="http://www.w3.org/2001/XMLSchema"
-    targetNamespace="http://www.ffmpeg.org/schema/ffprobe"
-    xmlns:ffprobe="http://www.ffmpeg.org/schema/ffprobe">
-
-    <xsd:element name="ffprobe" type="ffprobe:ffprobeType"/>
-
-    <xsd:complexType name="ffprobeType">
-        <xsd:sequence>
-            <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="format"   type="ffprobe:formatType"  minOccurs="0" maxOccurs="1" />
-            <xsd:element name="error"    type="ffprobe:errorType"   minOccurs="0" maxOccurs="1" />
-            <xsd:element name="program_version"  type="ffprobe:programVersionType"  minOccurs="0" maxOccurs="1" />
-            <xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="packetsType">
-        <xsd:sequence>
-            <xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="framesType">
-        <xsd:sequence>
-            <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="packetType">
-      <xsd:attribute name="codec_type"    type="xsd:string" use="required" />
-      <xsd:attribute name="stream_index"  type="xsd:int" use="required" />
-      <xsd:attribute name="pts"           type="xsd:long"  />
-      <xsd:attribute name="pts_time"      type="xsd:float" />
-      <xsd:attribute name="dts"           type="xsd:long"  />
-      <xsd:attribute name="dts_time"      type="xsd:float" />
-      <xsd:attribute name="duration"      type="xsd:long"  />
-      <xsd:attribute name="duration_time" type="xsd:float" />
-      <xsd:attribute name="size"          type="xsd:long" use="required" />
-      <xsd:attribute name="pos"           type="xsd:long"  />
-      <xsd:attribute name="flags"         type="xsd:string" use="required" />
-    </xsd:complexType>
-
-    <xsd:complexType name="frameType">
-      <xsd:attribute name="media_type"    type="xsd:string" use="required"/>
-      <xsd:attribute name="key_frame"     type="xsd:int"    use="required"/>
-      <xsd:attribute name="pts"           type="xsd:long" />
-      <xsd:attribute name="pts_time"      type="xsd:float"/>
-      <xsd:attribute name="pkt_pts"       type="xsd:long" />
-      <xsd:attribute name="pkt_pts_time"  type="xsd:float"/>
-      <xsd:attribute name="pkt_dts"       type="xsd:long" />
-      <xsd:attribute name="pkt_dts_time"  type="xsd:float"/>
-      <xsd:attribute name="pkt_pos"       type="xsd:long" />
-
-      <!-- audio attributes -->
-      <xsd:attribute name="sample_fmt"             type="xsd:string"/>
-      <xsd:attribute name="nb_samples"             type="xsd:long"  />
-
-      <!-- video attributes -->
-      <xsd:attribute name="width"                  type="xsd:long"  />
-      <xsd:attribute name="height"                 type="xsd:long"  />
-      <xsd:attribute name="pix_fmt"                type="xsd:string"/>
-      <xsd:attribute name="sample_aspect_ratio"    type="xsd:string"/>
-      <xsd:attribute name="pict_type"              type="xsd:string"/>
-      <xsd:attribute name="coded_picture_number"   type="xsd:long"  />
-      <xsd:attribute name="display_picture_number" type="xsd:long"  />
-      <xsd:attribute name="interlaced_frame"       type="xsd:int"   />
-      <xsd:attribute name="top_field_first"        type="xsd:int"   />
-      <xsd:attribute name="repeat_pict"            type="xsd:int"   />
-      <xsd:attribute name="reference"              type="xsd:int"   />
-    </xsd:complexType>
-
-    <xsd:complexType name="streamsType">
-        <xsd:sequence>
-            <xsd:element name="stream" type="ffprobe:streamType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="streamType">
-      <xsd:attribute name="index"            type="xsd:int" use="required"/>
-      <xsd:attribute name="codec_name"       type="xsd:string" />
-      <xsd:attribute name="codec_long_name"  type="xsd:string" />
-      <xsd:attribute name="codec_type"       type="xsd:string" />
-      <xsd:attribute name="codec_time_base"  type="xsd:string" use="required"/>
-      <xsd:attribute name="codec_tag"        type="xsd:string" use="required"/>
-      <xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
-
-      <!-- video attributes -->
-      <xsd:attribute name="width"                type="xsd:int"/>
-      <xsd:attribute name="height"               type="xsd:int"/>
-      <xsd:attribute name="has_b_frames"         type="xsd:int"/>
-      <xsd:attribute name="sample_aspect_ratio"  type="xsd:string"/>
-      <xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
-      <xsd:attribute name="pix_fmt"              type="xsd:string"/>
-      <xsd:attribute name="level"                type="xsd:int"/>
-      <xsd:attribute name="timecode"             type="xsd:string"/>
-
-      <!-- audio attributes -->
-      <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="bits_per_sample"  type="xsd:int"/>
-
-      <xsd:attribute name="id"               type="xsd:string"/>
-      <xsd:attribute name="r_frame_rate"     type="xsd:string" use="required"/>
-      <xsd:attribute name="avg_frame_rate"   type="xsd:string" use="required"/>
-      <xsd:attribute name="time_base"        type="xsd:string" use="required"/>
-      <xsd:attribute name="start_time"       type="xsd:float"/>
-      <xsd:attribute name="duration"         type="xsd:float"/>
-      <xsd:attribute name="nb_frames"        type="xsd:int"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="formatType">
-      <xsd:sequence>
-        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-      </xsd:sequence>
-
-      <xsd:attribute name="filename"         type="xsd:string" use="required"/>
-      <xsd:attribute name="nb_streams"       type="xsd:int"    use="required"/>
-      <xsd:attribute name="format_name"      type="xsd:string" use="required"/>
-      <xsd:attribute name="format_long_name" type="xsd:string" use="required"/>
-      <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:complexType>
-
-    <xsd:complexType name="tagType">
-      <xsd:attribute name="key"   type="xsd:string" use="required"/>
-      <xsd:attribute name="value" type="xsd:string" use="required"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="errorType">
-      <xsd:attribute name="code"   type="xsd:int"    use="required"/>
-      <xsd:attribute name="string" type="xsd:string" use="required"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="programVersionType">
-      <xsd:attribute name="version"          type="xsd:string" use="required"/>
-      <xsd:attribute name="copyright"        type="xsd:string" use="required"/>
-      <xsd:attribute name="build_date"       type="xsd:string" use="required"/>
-      <xsd:attribute name="build_time"       type="xsd:string" use="required"/>
-      <xsd:attribute name="compiler_type"    type="xsd:string" use="required"/>
-      <xsd:attribute name="compiler_version" type="xsd:string" use="required"/>
-      <xsd:attribute name="configuration"    type="xsd:string" use="required"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="libraryVersionType">
-      <xsd:attribute name="name"        type="xsd:string" use="required"/>
-      <xsd:attribute name="major"       type="xsd:int"    use="required"/>
-      <xsd:attribute name="minor"       type="xsd:int"    use="required"/>
-      <xsd:attribute name="micro"       type="xsd:int"    use="required"/>
-      <xsd:attribute name="version"     type="xsd:int"    use="required"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="libraryVersionsType">
-        <xsd:sequence>
-          <xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-</xsd:schema>

doc/ffserver.texi

@@ -147,7 +147,7 @@ that only captures in stereo and also requires that one channel be flipped.
 If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before
 starting ffmpeg.
 
-@subsection The audio and video lose sync after a while.
+@subsection The audio and video loose sync after a while.
 
 Yes, they do.
 

doc/filters.texi

@@ -93,7 +93,7 @@ Follows a BNF description for the filtergraph syntax:
 @c man begin AUDIO FILTERS
 
 When you configure your FFmpeg build, you can disable any of the
-existing filters using @code{--disable-filters}.
+existing filters using --disable-filters.
 The configure output will show the audio filters included in your
 build.
 
@@ -156,39 +156,6 @@ aformat=u8\\,s16:mono:packed
 aformat=s16:mono\\,stereo:all
 @end example
 
-@section amerge
-
-Merge two audio streams into a single multi-channel stream.
-
-This filter does not need any argument.
-
-If the channel layouts of the inputs are disjoint, and therefore compatible,
-the channel layout of the output will be set accordingly and the channels
-will be reordered as necessary. If the channel layouts of the inputs are not
-disjoint, the output will have all the channels of the first input then all
-the channels of the second input, in that order, and the channel layout of
-the output will be the default value corresponding to the total number of
-channels.
-
-For example, if the first input is in 2.1 (FL+FR+LF) and the second input
-is FC+BL+BR, then the output will be in 5.1, with the channels in the
-following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of the
-first input, b1 is the first channel of the second input).
-
-On the other hand, if both input are in stereo, the output channels will be
-in the default order: a1, a2, b1, b2, and the channel layout will be
-arbitrarily set to 4.0, which may or may not be the expected value.
-
-Both inputs must have the same sample rate, format and packing.
-
-If inputs do not have the same duration, the output will stop with the
-shortest.
-
-Example: merge two mono files into a stereo stream:
-@example
-amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge
-@end example
-
 @section anull
 
 Pass the audio source unchanged to the output.
@@ -257,48 +224,6 @@ expressed in the form "[@var{c0} @var{c1} @var{c2} @var{c3} @var{c4} @var{c5}
 @var{c6} @var{c7}]"
 @end table
 
-@section asplit
-
-Pass on the input audio to two outputs. Both outputs are identical to
-the input audio.
-
-For example:
-@example
-[in] asplit[out0], showaudio[out1]
-@end example
-
-will create two separate outputs from the same input, one cropped and
-one padded.
-
-@section astreamsync
-
-Forward two audio streams and control the order the buffers are forwarded.
-
-The argument to the filter is an expression deciding which stream should be
-forwarded next: if the result is negative, the first stream is forwarded; if
-the result is positive or zero, the second stream is forwarded. It can use
-the following variables:
-
-@table @var
-@item b1 b2
-number of buffers forwarded so far on each stream
-@item s1 s2
-number of samples forwarded so far on each stream
-@item t1 t2
-current timestamp of each stream
-@end table
-
-The default value is @code{t1-t2}, which means to always forward the stream
-that has a smaller timestamp.
-
-Example: stress-test @code{amerge} by randomly sending buffers on the wrong
-input, while avoiding too much of a desynchronization:
-@example
-amovie=file.ogg [a] ; amovie=file.mp3 [b] ;
-[a] [b] astreamsync=(2*random(1))-1+tanh(5*(t1-t2)) [a2] [b2] ;
-[a2] [b2] amerge
-@end example
-
 @section earwax
 
 Make audio easier to listen to on headphones.
@@ -315,9 +240,6 @@ Ported from SoX.
 Mix channels with specific gain levels. The filter accepts the output
 channel layout followed by a set of channels definitions.
 
-This filter is also designed to remap efficiently the channels of an audio
-stream.
-
 The filter accepts parameters of the form:
 "@var{l}:@var{outdef}:@var{outdef}:..."
 
@@ -345,8 +267,6 @@ If the `=' in a channel specification is replaced by `<', then the gains for
 that specification will be renormalized so that the total is 1, thus
 avoiding clipping noise.
 
-@subsection Mixing examples
-
 For example, if you want to down-mix from stereo to mono, but with a bigger
 factor for the left channel:
 @example
@@ -359,80 +279,10 @@ A customized down-mix to stereo that works automatically for 3-, 4-, 5- and
 pan=stereo: FL < FL + 0.5*FC + 0.6*BL + 0.6*SL : FR < FR + 0.5*FC + 0.6*BR + 0.6*SR
 @end example
 
-Note that @command{ffmpeg} integrates a default down-mix (and up-mix) system
+Note that @file{ffmpeg} integrates a default down-mix (and up-mix) system
 that should be preferred (see "-ac" option) unless you have very specific
 needs.
 
-@subsection Remapping examples
-
-The channel remapping will be effective if, and only if:
-
-@itemize
-@item gain coefficients are zeroes or ones,
-@item only one input per channel output,
-@item the number of output channels is supported by libswresample (16 at the
-      moment)
-@c if SWR_CH_MAX changes, fix the line above.
-@end itemize
-
-If all these conditions are satisfied, the filter will notify the user ("Pure
-channel mapping detected"), and use an optimized and lossless method to do the
-remapping.
-
-For example, if you have a 5.1 source and want a stereo audio stream by
-dropping the extra channels:
-@example
-pan="stereo: c0=FL : c1=FR"
-@end example
-
-Given the same source, you can also switch front left and front right channels
-and keep the input channel layout:
-@example
-pan="5.1: c0=c1 : c1=c0 : c2=c2 : c3=c3 : c4=c4 : c5=c5"
-@end example
-
-If the input is a stereo audio stream, you can mute the front left channel (and
-still keep the stereo channel layout) with:
-@example
-pan="stereo:c1=c1"
-@end example
-
-Still with a stereo audio stream input, you can copy the right channel in both
-front left and right:
-@example
-pan="stereo: c0=FR : c1=FR"
-@end example
-
-@section silencedetect
-
-Detect silence in an audio stream.
-
-This filter logs a message when it detects that the input audio volume is less
-or equal to a noise tolerance value for a duration greater or equal to the
-minimum detected noise duration.
-
-The printed times and duration are expressed in seconds.
-
-@table @option
-@item duration, d
-Set silence duration until notification (default is 2 seconds).
-
-@item noise, n
-Set noise tolerance. Can be specified in dB (in case "dB" is appended to the
-specified value) or amplitude ratio. Default is -60dB, or 0.001.
-@end table
-
-Detect 5 seconds of silence with -50dB noise tolerance:
-@example
-silencedetect=n=-50dB:d=5
-@end example
-
-Complete example with @command{ffmpeg} to detect silence with 0.0001 noise
-tolerance in @file{silence.mp3}:
-@example
-ffmpeg -f lavfi -i amovie=silence.mp3,silencedetect=noise=0.0001 -f null -
-@end example
-
 @section volume
 
 Adjust the input audio volume.
@@ -442,7 +292,7 @@ how the audio volume will be increased or decreased.
 
 Output values are clipped to the maximum value.
 
-If @var{vol} is expressed as a decimal number, the output audio
+If @var{vol} is expressed as a decimal number, and the output audio
 volume is given by the relation:
 @example
 @var{output_volume} = @var{vol} * @var{input_volume}
@@ -728,7 +578,7 @@ tools.
 @c man begin VIDEO FILTERS
 
 When you configure your FFmpeg build, you can disable any of the
-existing filters using @code{--disable-filters}.
+existing filters using --disable-filters.
 The configure output will show the video filters included in your
 build.
 
@@ -1298,15 +1148,6 @@ the number of input frame, starting from 0
 
 @item t
 timestamp expressed in seconds, NAN if the input timestamp is unknown
-
-@item timecode
-initial timecode representation in "hh:mm:ss[:;.]ff" format. It can be used
-with or without text parameter. @var{rate} option must be specified.
-Note that timecode options are @emph{not} effective if FFmpeg is build with
-@code{--disable-avcodec}.
-
-@item r, rate
-frame rate (timecode only)
 @end table
 
 Some examples follow.
@@ -1488,7 +1329,7 @@ format=yuv420p:yuv444p:yuv410p
 Apply a frei0r effect to the input video.
 
 To enable compilation of this filter you need to install the frei0r
-header and configure FFmpeg with @code{--enable-frei0r}.
+header and configure FFmpeg with --enable-frei0r.
 
 The filter supports the syntax:
 @example
@@ -1847,7 +1688,7 @@ Pass the video source unchanged to the output.
 Apply video transform using libopencv.
 
 To enable this filter install libopencv library and headers and
-configure FFmpeg with @code{--enable-libopencv}.
+configure FFmpeg with --enable-libopencv.
 
 The filter takes the parameters: @var{filter_name}@{:=@}@var{filter_params}.
 
@@ -2157,6 +1998,9 @@ input sample aspect ratio
 @item dar
 input display aspect ratio, it is the same as (@var{iw} / @var{ih}) * @var{sar}
 
+@item sar
+input sample aspect ratio
+
 @item hsub, vsub
 horizontal and vertical chroma subsample values. For example for the
 pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1.
@@ -2549,64 +2393,6 @@ For example:
 will create two separate outputs from the same input, one cropped and
 one padded.
 
-@section thumbnail
-Select the most representative frame in a given sequence of consecutive frames.
-
-It accepts as argument the frames batch size to analyze (default @var{N}=100);
-in a set of @var{N} frames, the filter will pick one of them, and then handle
-the next batch of @var{N} frames until the end.
-
-Since the filter keeps track of the whole frames sequence, a bigger @var{N}
-value will result in a higher memory usage, so a high value is not recommended.
-
-The following example extract one picture each 50 frames:
-@example
-thumbnail=50
-@end example
-
-Complete example of a thumbnail creation with @command{ffmpeg}:
-@example
-ffmpeg -i in.avi -vf thumbnail,scale=300:200 -frames:v 1 out.png
-@end example
-
-@section tinterlace
-
-Perform various types of temporal field interlacing.
-
-Frames are counted starting from 1, so the first input frame is
-considered odd.
-
-This filter accepts a single parameter specifying the mode. Available
-modes are:
-
-@table @samp
-@item 0
-Move odd frames into the upper field, even into the lower field,
-generating a double height frame at half framerate.
-
-@item 1
-Only output even frames, odd frames are dropped, generating a frame with
-unchanged height at half framerate.
-
-@item 2
-Only output odd frames, even frames are dropped, generating a frame with
-unchanged height at half framerate.
-
-@item 3
-Expand each frame to full height, but pad alternate lines with black,
-generating a frame with double height at the same input framerate.
-
-@item 4
-Interleave the upper field from odd frames with the lower field from
-even frames, generating a frame with unchanged height at half framerate.
-
-@item 5
-Interleave the lower field from odd frames with the upper field from
-even frames, generating a frame with unchanged height at half framerate.
-@end table
-
-Default mode is 0.
-
 @section transpose
 
 Transpose rows with columns in the input video and optionally flip it.
@@ -3089,7 +2875,7 @@ will generate a "dc_luma" test pattern.
 Provide a frei0r source.
 
 To enable compilation of this filter you need to install the frei0r
-header and configure FFmpeg with @code{--enable-frei0r}.
+header and configure FFmpeg with --enable-frei0r.
 
 The source supports the syntax:
 @example
@@ -3272,7 +3058,7 @@ number or a valid video frame rate abbreviation. The default value is
 @item sar
 Set the sample aspect ratio of the sourced video.
 
-@item duration, d
+@item duration
 Set the video duration of the sourced video. The accepted syntax is:
 @example
 [-]HH[:MM[:SS[.m...]]]
@@ -3282,14 +3068,6 @@ See also the function @code{av_parse_time()}.
 
 If not specified, or the expressed duration is negative, the video is
 supposed to be generated forever.
-
-@item decimals, n
-Set the number of decimals to show in the timestamp, only used in the
-@code{testsrc} source.
-
-The displayed timestamp value will correspond to the original
-timestamp value multiplied by the power of 10 of the specified
-value. Default value is 0.
 @end table
 
 For example the following:

doc/general.texi

@@ -94,7 +94,7 @@ details), you must upgrade FFmpeg's license to GPL in order to use it.
 
 
 
-@chapter Supported File Formats, Codecs or Features
+@chapter Supported File Formats and Codecs
 
 You can use the @code{-formats} and @code{-codecs} options to have an exhaustive list.
 
@@ -134,7 +134,7 @@ library:
 @item Brute Force & Ignorance   @tab   @tab X
     @tab Used in the game Flash Traffic: City of Angels.
 @item BWF                       @tab X @tab X
-@item CRI ADX                   @tab X @tab X
+@item CRI ADX                   @tab   @tab X
     @tab Audio-only format used in console video games.
 @item Discworld II BMV          @tab   @tab X
 @item Interplay C93             @tab   @tab X
@@ -307,7 +307,6 @@ library:
 @item RTP                       @tab X @tab X
 @item RTSP                      @tab X @tab X
 @item SAP                       @tab X @tab X
-@item SBG                       @tab   @tab X
 @item SDP                       @tab   @tab X
 @item Sega FILM/CPK             @tab   @tab X
     @tab Used in many Sega Saturn console games.
@@ -317,9 +316,8 @@ library:
     @tab Used in Sierra CD-ROM games.
 @item Smacker                   @tab   @tab X
     @tab Multimedia format used by many games.
-@item SMJPEG                    @tab X @tab X
-    @tab Used in certain Loki game ports.
-@item Sony OpenMG (OMA)         @tab X @tab X
+@item SMJPEG                    @tab   @tab X
+@item Sony OpenMG (OMA)         @tab   @tab X
     @tab Audio format used in Sony Sonic Stage and Sony Vegas.
 @item Sony PlayStation STR      @tab   @tab X
 @item Sony Wave64 (W64)         @tab   @tab X
@@ -399,8 +397,6 @@ 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 XWD  @tab X @tab X
-    @tab X Window Dump image format
 @end multitable
 
 @code{X} means that encoding (resp. decoding) is supported.
@@ -440,8 +436,6 @@ following image formats are supported:
 @item Autodesk Animator Flic video  @tab     @tab  X
 @item Autodesk RLE           @tab     @tab  X
     @tab fourcc: AASC
-@item Avid 1:1 10-bit RGB Packer  @tab  X  @tab  X
-    @tab fourcc: AVrp
 @item AVS (Audio Video Standard) video  @tab     @tab  X
     @tab Video encoding used by the Creature Shock game.
 @item Beam Software VB       @tab     @tab  X
@@ -577,8 +571,8 @@ following image formats are supported:
     @tab fourcc: 'smc '
 @item QuickTime video (RPZA) @tab     @tab  X
     @tab fourcc: rpza
-@item R10K AJA Kona 10-bit RGB Codec     @tab  X  @tab  X
-@item R210 Quicktime Uncompressed RGB 10-bit     @tab  X  @tab  X
+@item R10K AJA Kona 10-bit RGB Codec     @tab     @tab  X
+@item R210 Quicktime Uncompressed RGB 10-bit     @tab     @tab  X
 @item Raw Video              @tab  X  @tab  X
 @item RealVideo 1.0          @tab  X  @tab  X
 @item RealVideo 2.0          @tab  X  @tab  X
@@ -610,9 +604,8 @@ following image formats are supported:
 @item Tiertex Limited SEQ video  @tab     @tab  X
     @tab Codec used in DOS CD-ROM FlashBack game.
 @item Ut Video               @tab     @tab  X
-@item v210 QuickTime uncompressed 4:2:2 10-bit     @tab  X  @tab  X
-@item v308 QuickTime uncompressed 4:4:4            @tab  X  @tab  X
-@item v410 QuickTime uncompressed 4:4:4 10-bit     @tab  X  @tab  X
+@item V210 Quicktime Uncompressed 4:2:2 10-bit     @tab  X  @tab  X
+@item v410 Quicktime Uncompressed 4:4:4 10-bit     @tab  X  @tab  X
 @item VBLE Lossless Codec    @tab     @tab  X
 @item VMware Screen Codec / VMware Video  @tab     @tab  X
     @tab Codec used in videos captured by VMware.
@@ -630,8 +623,6 @@ following image formats are supported:
 @item WMV7                   @tab  X  @tab  X
 @item YAMAHA SMAF            @tab  X  @tab  X
 @item Psygnosis YOP Video    @tab     @tab  X
-@item yuv4                   @tab  X  @tab  X
-    @tab libquicktime uncompressed packed 4:2:0
 @item ZLIB                   @tab  X  @tab  X
     @tab part of LCL, encoder experimental
 @item Zip Motion Blocks Video  @tab   X @tab  X
@@ -866,15 +857,4 @@ performance on systems without hardware floating point support).
 
 @code{X} means that input/output is supported.
 
-@section Timecode
-
-@multitable @columnfractions .4 .1 .1
-@item Codec/format      @tab Read   @tab Write
-@item DV                @tab X      @tab X
-@item GXF               @tab X      @tab X
-@item MOV               @tab X      @tab
-@item MPEG1/2           @tab X      @tab X
-@item MXF               @tab        @tab X
-@end multitable
-
 @bye

doc/indevs.texi

@@ -196,12 +196,12 @@ device.
 Once you have created one or more JACK readable clients, you need to
 connect them to one or more JACK writable clients.
 
-To connect or disconnect JACK clients you can use the @command{jack_connect}
-and @command{jack_disconnect} programs, or do it through a graphical interface,
-for example with @command{qjackctl}.
+To connect or disconnect JACK clients you can use the
+@file{jack_connect} and @file{jack_disconnect} programs, or do it
+through a graphical interface, for example with @file{qjackctl}.
 
 To list the JACK clients and their properties you can invoke the command
-@command{jack_lsp}.
+@file{jack_lsp}.
 
 Follows an example which shows how to capture a JACK readable client
 with @command{ffmpeg}.
@@ -260,7 +260,7 @@ device.
 
 @itemize
 @item
-Create a color video stream and play it back with @command{ffplay}:
+Create a color video stream and play it back with @file{ffplay}:
 @example
 ffplay -f lavfi -graph "color=pink [out0]" dummy
 @end example
@@ -280,14 +280,14 @@ ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [ou
 
 @item
 Read an audio stream from a file using the amovie source and play it
-back with @command{ffplay}:
+back with @file{ffplay}:
 @example
 ffplay -f lavfi "amovie=test.wav"
 @end example
 
 @item
 Read an audio stream and a video stream and play it back with
-@command{ffplay}:
+@file{ffplay}:
 @example
 ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"
 @end example
@@ -380,7 +380,7 @@ $ ffmpeg -f openal -i '' out.ogg
 @end example
 
 Capture from two devices simultaneously, writing to two different files,
-within the same @command{ffmpeg} command:
+within the same @file{ffmpeg} command:
 @example
 $ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out1.ogg -f openal -i 'ALSA Default' out2.ogg
 @end example
@@ -415,7 +415,7 @@ The filename to provide to the input device is a source device or the
 string "default"
 
 To list the pulse source devices and their properties you can invoke
-the command @command{pactl list sources}.
+the command @file{pactl list sources}.
 
 @example
 ffmpeg -f pulse -i default /tmp/pulse.wav
@@ -515,9 +515,9 @@ kind @file{/dev/video@var{N}}, where @var{N} is a number associated to
 the device.
 
 Video4Linux and Video4Linux2 devices only support a limited set of
-@var{width}x@var{height} sizes and framerates. You can check which are
-supported for example with the command @command{dov4l} for Video4Linux
-devices and using @command{-list_formats all} for Video4Linux2 devices.
+@var{width}x@var{height} sizes and frame rates. You can check which are
+supported for example with the command @file{dov4l} for Video4Linux
+devices and the command @file{v4l-info} for Video4Linux2 devices.
 
 If the size for the device is set to 0x0, the input device will
 try to auto-detect the size to use.
@@ -579,7 +579,7 @@ default to 0.
 
 Check the X11 documentation (e.g. man X) for more detailed information.
 
-Use the @command{dpyinfo} program for getting basic information about the
+Use the @file{dpyinfo} program for getting basic information about the
 properties of your X11 display (e.g. grep for "name" or "dimensions").
 
 For example to grab from @file{:0.0} using @command{ffmpeg}:

doc/issue_tracker.txt

@@ -24,7 +24,7 @@ a mail for every change to every issue.
 The subscription URL for the ffmpeg-trac list is:
 http(s)://ffmpeg.org/mailman/listinfo/ffmpeg-trac
 The URL of the webinterface of the tracker is:
-http(s)://ffmpeg.org/trac/ffmpeg
+http(s)://trac.ffmpeg.org
 
 Type:
 -----

doc/libavfilter.texi

@@ -43,13 +43,13 @@ The result will be that in output the top half of the video is mirrored
 onto the bottom half.
 
 Video filters are loaded using the @var{-vf} option passed to
-@command{ffmpeg} or to @command{ffplay}. Filters in the same linear
-chain are separated by commas. In our example, @var{split, fifo,
-overlay} are in one linear chain, and @var{fifo, crop, vflip} are in
-another. The points where the linear chains join are labeled by names
-enclosed in square brackets. In our example, that is @var{[T1]} and
-@var{[T2]}. The magic labels @var{[in]} and @var{[out]} are the points
-where video is input and output.
+ffmpeg or to ffplay. Filters in the same linear chain are separated by
+commas. In our example, @var{split, fifo, overlay} are in one linear
+chain, and @var{fifo, crop, vflip} are in another. The points where
+the linear chains join are labeled by names enclosed in square
+brackets. In our example, that is @var{[T1]} and @var{[T2]}. The magic
+labels @var{[in]} and @var{[out]} are the points where video is input
+and output.
 
 Some filters take in input a list of parameters: they are specified
 after the filter name and an equal sign, and are separated each other

doc/muxers.texi

@@ -90,7 +90,6 @@ ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc -
 
 See also the @ref{crc} muxer.
 
-@anchor{image2}
 @section image2
 
 Image file muxer.
@@ -286,35 +285,4 @@ For example a 3D WebM clip can be created using the following command line:
 ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm
 @end example
 
-@section segment
-
-Basic stream segmenter.
-
-The segmenter muxer outputs streams to a number of separate files of nearly
-fixed duration. Output filename pattern can be set in a fashion similar to
-@ref{image2}.
-
-Every segment starts with a video keyframe, if a video stream is present.
-The segment muxer works best with a single constant frame rate video.
-
-Optionally it can generate a flat list of the created segments, one segment
-per line.
-
-@table @option
-@item segment_format @var{format}
-Override the inner container format, by default it is guessed by the filename
-extension.
-@item segment_time @var{t}
-Set segment duration to @var{t} seconds.
-@item segment_list @var{name}
-Generate also a listfile named @var{name}.
-@item segment_list_size @var{size}
-Overwrite the listfile once it reaches @var{size} entries.
-@end table
-
-@example
-ffmpeg -i in.mkv -c copy -map 0 -f segment -list out.list out%03d.nut
-@end example
-
-
 @c man end MUXERS

doc/outdevs.texi

@@ -60,7 +60,7 @@ If not specified it defaults to the size of the input video.
 
 @subsection Examples
 
-The following command shows the @command{ffmpeg} output is an
+The following command shows the @file{ffmpeg} output is an
 SDL window, forcing its size to the qcif format:
 @example
 ffmpeg -i INPUT -vcodec rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output"

doc/platform.texi

@@ -235,8 +235,6 @@ make install
 Your install path (@file{/usr/local/} by default) should now have the
 necessary DLL and LIB files under the @file{bin} directory.
 
-@end enumerate
-
 Alternatively, build the libraries with a cross compiler, according to
 the instructions below in @ref{Cross compilation for Windows with Linux}.
 

doc/protocols.texi

@@ -52,7 +52,7 @@ resource to be concatenated, each one possibly specifying a distinct
 protocol.
 
 For example to read a sequence of files @file{split1.mpeg},
-@file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
+@file{split2.mpeg}, @file{split3.mpeg} with @file{ffplay} use the
 command:
 @example
 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
@@ -155,8 +155,8 @@ be seekable, so they will fail with the pipe output protocol.
 
 Real-Time Messaging Protocol.
 
-The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
-content across a TCP/IP network.
+The Real-Time Messaging Protocol (RTMP) is used for streaming
+multimedia content across a TCP/IP network.
 
 The required syntax is:
 @example
@@ -183,7 +183,7 @@ application specified in @var{app}, may be prefixed by "mp4:".
 
 @end table
 
-For example to read with @command{ffplay} a multimedia resource named
+For example to read with @file{ffplay} a multimedia resource named
 "sample" from the application "vod" from an RTMP server "myserver":
 @example
 ffplay rtmp://myserver/vod/sample
@@ -224,7 +224,7 @@ For example, to stream a file in real-time to an RTMP server using
 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
 @end example
 
-To play the same stream using @command{ffplay}:
+To play the same stream using @file{ffplay}:
 @example
 ffplay "rtmp://myserver/live/mystream live=1"
 @end example
@@ -249,7 +249,7 @@ The required syntax for a RTSP url is:
 rtsp://@var{hostname}[:@var{port}]/@var{path}
 @end example
 
-The following options (set on the @command{ffmpeg}/@command{ffplay} command
+The following options (set on the @command{ffmpeg}/@file{ffplay} command
 line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
 are supported:
 
@@ -288,7 +288,7 @@ When receiving data over UDP, the demuxer tries to reorder received packets
 order for this to be enabled, a maximum delay must be specified in the
 @code{max_delay} field of AVFormatContext.
 
-When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
+When watching multi-bitrate Real-RTSP streams with @file{ffplay}, the
 streams to display can be chosen with @code{-vst} @var{n} and
 @code{-ast} @var{n} for video and audio respectively, and can be switched
 on the fly by pressing @code{v} and @code{a}.
@@ -365,13 +365,13 @@ To broadcast a stream on the local subnet, for watching in VLC:
 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
 @end example
 
-Similarly, for watching in @command{ffplay}:
+Similarly, for watching in ffplay:
 
 @example
 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
 @end example
 
-And for watching in @command{ffplay}, over IPv6:
+And for watching in ffplay, over IPv6:
 
 @example
 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]

doc/soc.txt

@@ -18,7 +18,7 @@ essential that changes to their codebase are publicly visible, clean and
 easy reviewable that again leads us to:
 * use of a revision control system like git
 * separation of cosmetic from non-cosmetic changes (this is almost entirely
-  ignored by mentors and students in soc 2006 which might lead to a surprise
+  ignored by mentors and students in soc 2006 which might lead to a suprise
   when the code will be reviewed at the end before a possible inclusion in
   FFmpeg, individual changes were generally not reviewable due to cosmetics).
 * frequent commits, so that comments can be provided early

ffserver.c

@@ -1,4 +1,5 @@
 /*
+ * Multiple format streaming server
  * Copyright (c) 2000, 2001, 2002 Fabrice Bellard
  *
  * This file is part of FFmpeg.
@@ -18,11 +19,6 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-/**
- * @file
- * multiple format streaming server based on the FFmpeg libraries
- */
-
 #include "config.h"
 #if !HAVE_CLOSESOCKET
 #define closesocket close
@@ -30,16 +26,13 @@
 #include <string.h>
 #include <stdlib.h>
 #include "libavformat/avformat.h"
-// FIXME those are internal headers, avserver _really_ shouldn't use them
 #include "libavformat/ffm.h"
 #include "libavformat/network.h"
 #include "libavformat/os_support.h"
 #include "libavformat/rtpdec.h"
 #include "libavformat/rtsp.h"
+// XXX for ffio_open_dyn_packet_buffer, to be removed
 #include "libavformat/avio_internal.h"
-#include "libavformat/internal.h"
-#include "libavformat/url.h"
-
 #include "libavutil/avstring.h"
 #include "libavutil/lfg.h"
 #include "libavutil/dict.h"
@@ -482,7 +475,7 @@ static void start_children(FFStream *feed)
                     slash++;
                 strcpy(slash, "ffmpeg");
 
-                http_log("Launch command line: ");
+                http_log("Launch commandline: ");
                 http_log("%s ", pathname);
                 for (i = 1; feed->child_argv[i] && feed->child_argv[i][0]; i++)
                     http_log("%s ", feed->child_argv[i]);
@@ -502,10 +495,7 @@ static void start_children(FFStream *feed)
                 }
 
                 /* This is needed to make relative pathnames work */
-                if (chdir(my_program_dir) < 0) {
-                    http_log("chdir failed\n");
-                    exit(1);
-                }
+                chdir(my_program_dir);
 
                 signal(SIGPIPE, SIG_DFL);
 
@@ -859,7 +849,7 @@ static void close_connection(HTTPContext *c)
             if (st->codec->codec)
                 avcodec_close(st->codec);
         }
-        avformat_close_input(&c->fmt_in);
+        av_close_input_file(c->fmt_in);
     }
 
     /* free RTP output streams if any */
@@ -877,7 +867,7 @@ static void close_connection(HTTPContext *c)
         }
         h = c->rtp_handles[i];
         if (h)
-            ffurl_close(h);
+            url_close(h);
     }
 
     ctx = &c->fmt_ctx;
@@ -2122,6 +2112,22 @@ static void compute_status(HTTPContext *c)
     c->buffer_end = c->pb_buffer + len;
 }
 
+/* check if the parser needs to be opened for stream i */
+static void open_parser(AVFormatContext *s, int i)
+{
+    AVStream *st = s->streams[i];
+    AVCodec *codec;
+
+    if (!st->codec->codec) {
+        codec = avcodec_find_decoder(st->codec->codec_id);
+        if (codec && (codec->capabilities & CODEC_CAP_PARSE_ONLY)) {
+            st->codec->parse_only = 1;
+            if (avcodec_open2(st->codec, codec, NULL) < 0)
+                st->codec->parse_only = 0;
+        }
+    }
+}
+
 static int open_input_stream(HTTPContext *c, const char *info)
 {
     char buf[128];
@@ -2163,10 +2169,14 @@ static int open_input_stream(HTTPContext *c, const char *info)
     c->fmt_in = s;
     if (strcmp(s->iformat->name, "ffm") && avformat_find_stream_info(c->fmt_in, NULL) < 0) {
         http_log("Could not find stream info '%s'\n", input_filename);
-        avformat_close_input(&s);
+        av_close_input_file(s);
         return -1;
     }
 
+    /* open each parser */
+    for(i=0;i<s->nb_streams;i++)
+        open_parser(s, i);
+
     /* choose stream as clock source (we favorize video stream if
        present) for packet sending */
     c->pts_stream_index = 0;
@@ -2258,6 +2268,7 @@ static int http_prepare_data(HTTPContext *c)
          * Default value from FFmpeg
          * Try to set it use configuration option
          */
+        c->fmt_ctx.preload   = (int)(0.5*AV_TIME_BASE);
         c->fmt_ctx.max_delay = (int)(0.7*AV_TIME_BASE);
 
         if (avformat_write_header(&c->fmt_ctx, NULL) < 0) {
@@ -2300,7 +2311,8 @@ static int http_prepare_data(HTTPContext *c)
                     return 0;
                 } else {
                     if (c->stream->loop) {
-                        avformat_close_input(&c->fmt_in);
+                        av_close_input_file(c->fmt_in);
+                        c->fmt_in = NULL;
                         if (open_input_stream(c, "") < 0)
                             goto no_loop;
                         goto redo;
@@ -2376,7 +2388,7 @@ static int http_prepare_data(HTTPContext *c)
                         if (c->rtp_protocol == RTSP_LOWER_TRANSPORT_TCP)
                             max_packet_size = RTSP_TCP_MAX_PACKET_SIZE;
                         else
-                            max_packet_size = c->rtp_handles[c->packet_stream_index]->max_packet_size;
+                            max_packet_size = url_get_max_packet_size(c->rtp_handles[c->packet_stream_index]);
                         ret = ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size);
                     } else {
                         ret = avio_open_dyn_buf(&ctx->pb);
@@ -2529,8 +2541,8 @@ static int http_send_data(HTTPContext *c)
                 } else {
                     /* send RTP packet directly in UDP */
                     c->buffer_ptr += 4;
-                    ffurl_write(c->rtp_handles[c->packet_stream_index],
-                                c->buffer_ptr, len);
+                    url_write(c->rtp_handles[c->packet_stream_index],
+                              c->buffer_ptr, len);
                     c->buffer_ptr += len;
                     /* here we continue as we can send several packets per 10 ms slot */
                 }
@@ -2724,7 +2736,7 @@ static int http_receive_data(HTTPContext *c)
 
             /* Now we have the actual streams */
             if (s->nb_streams != feed->nb_streams) {
-                avformat_close_input(&s);
+                av_close_input_stream(s);
                 av_free(pb);
                 http_log("Feed '%s' stream number does not match registered feed\n",
                          c->stream->feed_filename);
@@ -2737,7 +2749,7 @@ static int http_receive_data(HTTPContext *c)
                 avcodec_copy_context(fst->codec, st->codec);
             }
 
-            avformat_close_input(&s);
+            av_close_input_stream(s);
             av_free(pb);
         }
         c->buffer_ptr = c->buffer;
@@ -3413,10 +3425,10 @@ static int rtp_new_av_stream(HTTPContext *c,
                      "rtp://%s:%d", ipaddr, ntohs(dest_addr->sin_port));
         }
 
-        if (ffurl_open(&h, ctx->filename, AVIO_FLAG_WRITE, NULL, NULL) < 0)
+        if (url_open(&h, ctx->filename, AVIO_FLAG_WRITE) < 0)
             goto fail;
         c->rtp_handles[stream_index] = h;
-        max_packet_size = h->max_packet_size;
+        max_packet_size = url_get_max_packet_size(h);
         break;
     case RTSP_LOWER_TRANSPORT_TCP:
         /* RTP/TCP case */
@@ -3439,7 +3451,7 @@ static int rtp_new_av_stream(HTTPContext *c,
     if (avformat_write_header(ctx, NULL) < 0) {
     fail:
         if (h)
-            ffurl_close(h);
+            url_close(h);
         av_free(ctx);
         return -1;
     }
@@ -3476,7 +3488,7 @@ static AVStream *add_av_stream1(FFStream *stream, AVCodecContext *codec, int cop
     }
     fst->priv_data = av_mallocz(sizeof(FeedData));
     fst->index = stream->nb_streams;
-    avpriv_set_pts_info(fst, 33, 1, 90000);
+    av_set_pts_info(fst, 33, 1, 90000);
     fst->sample_aspect_ratio = codec->sample_aspect_ratio;
     stream->streams[stream->nb_streams++] = fst;
     return fst;
@@ -3617,7 +3629,7 @@ static void build_file_streams(void)
                 if (avformat_find_stream_info(infile, NULL) < 0) {
                     http_log("Could not find codec parameters from '%s'\n",
                              stream->feed_filename);
-                    avformat_close_input(&infile);
+                    av_close_input_file(infile);
                     goto fail;
                 }
                 extract_mpeg4_header(infile);
@@ -3625,7 +3637,7 @@ static void build_file_streams(void)
                 for(i=0;i<infile->nb_streams;i++)
                     add_av_stream1(stream, infile->streams[i]->codec, 1);
 
-                avformat_close_input(&infile);
+                av_close_input_file(infile);
             }
         }
     }
@@ -3715,7 +3727,7 @@ static void build_feed_streams(void)
                     http_log("Deleting feed file '%s' as stream counts differ (%d != %d)\n",
                         feed->feed_filename, s->nb_streams, feed->nb_streams);
 
-                avformat_close_input(&s);
+                av_close_input_file(s);
             } else
                 http_log("Deleting feed file '%s' as it appears to be corrupt\n",
                         feed->feed_filename);

libavcodec/4xm.c

@@ -132,8 +132,10 @@ typedef struct FourXContext{
     AVFrame current_picture, last_picture;
     GetBitContext pre_gb;          ///< ac/dc prefix
     GetBitContext gb;
-    GetByteContext g;
-    GetByteContext g2;
+    const uint8_t *bytestream;
+    const uint8_t *bytestream_end;
+    const uint16_t *wordstream;
+    const uint16_t *wordstream_end;
     int mv[256];
     VLC pre_vlc;
     int last_dc;
@@ -328,11 +330,11 @@ static void decode_p_block(FourXContext *f, uint16_t *dst, uint16_t *src, int lo
     assert(code>=0 && code<=6);
 
     if(code == 0){
-        if (f->g.buffer_end - f->g.buffer < 1){
+        if (f->bytestream_end - f->bytestream < 1){
             av_log(f->avctx, AV_LOG_ERROR, "bytestream overread\n");
             return;
         }
-        src += f->mv[ *f->g.buffer++ ];
+        src += f->mv[ *f->bytestream++ ];
         if(start > src || src > end){
             av_log(f->avctx, AV_LOG_ERROR, "mv out of pic\n");
             return;
@@ -349,37 +351,37 @@ static void decode_p_block(FourXContext *f, uint16_t *dst, uint16_t *src, int lo
     }else if(code == 3 && f->version<2){
         mcdc(dst, src, log2w, h, stride, 1, 0);
     }else if(code == 4){
-        if (f->g.buffer_end - f->g.buffer < 1){
+        if (f->bytestream_end - f->bytestream < 1){
             av_log(f->avctx, AV_LOG_ERROR, "bytestream overread\n");
             return;
         }
-        src += f->mv[ *f->g.buffer++ ];
+        src += f->mv[ *f->bytestream++ ];
         if(start > src || src > end){
             av_log(f->avctx, AV_LOG_ERROR, "mv out of pic\n");
             return;
         }
-        if (f->g2.buffer_end - f->g2.buffer < 1){
+        if (f->wordstream_end - f->wordstream < 1){
             av_log(f->avctx, AV_LOG_ERROR, "wordstream overread\n");
             return;
         }
-        mcdc(dst, src, log2w, h, stride, 1, bytestream2_get_le16(&f->g2));
+        mcdc(dst, src, log2w, h, stride, 1, av_le2ne16(*f->wordstream++));
     }else if(code == 5){
-        if (f->g2.buffer_end - f->g2.buffer < 1){
+        if (f->wordstream_end - f->wordstream < 1){
             av_log(f->avctx, AV_LOG_ERROR, "wordstream overread\n");
             return;
         }
-        mcdc(dst, src, log2w, h, stride, 0, bytestream2_get_le16(&f->g2));
+        mcdc(dst, src, log2w, h, stride, 0, av_le2ne16(*f->wordstream++));
     }else if(code == 6){
-        if (f->g2.buffer_end - f->g2.buffer < 2){
+        if (f->wordstream_end - f->wordstream < 2){
             av_log(f->avctx, AV_LOG_ERROR, "wordstream overread\n");
             return;
         }
         if(log2w){
-            dst[0] = bytestream2_get_le16(&f->g2);
-            dst[1] = bytestream2_get_le16(&f->g2);
+            dst[0] = av_le2ne16(*f->wordstream++);
+            dst[1] = av_le2ne16(*f->wordstream++);
         }else{
-            dst[0     ] = bytestream2_get_le16(&f->g2);
-            dst[stride] = bytestream2_get_le16(&f->g2);
+            dst[0     ] = av_le2ne16(*f->wordstream++);
+            dst[stride] = av_le2ne16(*f->wordstream++);
         }
     }
 }
@@ -391,7 +393,7 @@ static int decode_p_frame(FourXContext *f, const uint8_t *buf, int length){
     uint16_t *src= (uint16_t*)f->last_picture.data[0];
     uint16_t *dst= (uint16_t*)f->current_picture.data[0];
     const int stride= f->current_picture.linesize[0]>>1;
-    unsigned int bitstream_size, bytestream_size, wordstream_size, extra, bytestream_offset, wordstream_offset;
+    unsigned int bitstream_size, bytestream_size, wordstream_size, extra;
 
     if(f->version>1){
         extra=20;
@@ -423,10 +425,10 @@ static int decode_p_frame(FourXContext *f, const uint8_t *buf, int length){
     memset((uint8_t*)f->bitstream_buffer + bitstream_size, 0, FF_INPUT_BUFFER_PADDING_SIZE);
     init_get_bits(&f->gb, f->bitstream_buffer, 8*bitstream_size);
 
-    wordstream_offset = extra + bitstream_size;
-    bytestream_offset = extra + bitstream_size + wordstream_size;
-    bytestream2_init(&f->g2, buf + wordstream_offset, length - wordstream_offset);
-    bytestream2_init(&f->g,  buf + bytestream_offset, length - bytestream_offset);
+    f->wordstream= (const uint16_t*)(buf + extra + bitstream_size);
+    f->wordstream_end= f->wordstream + wordstream_size/2;
+    f->bytestream= buf + extra + bitstream_size + wordstream_size;
+    f->bytestream_end = f->bytestream + bytestream_size;
 
     init_mv(f);
 
@@ -438,6 +440,15 @@ static int decode_p_frame(FourXContext *f, const uint8_t *buf, int length){
         dst += 8*stride;
     }
 
+    if(   bitstream_size != (get_bits_count(&f->gb)+31)/32*4
+       || (((const char*)f->wordstream - (const char*)buf + 2)&~2) != extra + bitstream_size + wordstream_size
+       || (((const char*)f->bytestream - (const char*)buf + 3)&~3) != extra + bitstream_size + wordstream_size + bytestream_size)
+        av_log(f->avctx, AV_LOG_ERROR, " %d %td %td bytes left\n",
+            bitstream_size - (get_bits_count(&f->gb)+31)/32*4,
+            -(((const char*)f->bytestream - (const char*)buf + 3)&~3) + (extra + bitstream_size + wordstream_size + bytestream_size),
+            -(((const char*)f->wordstream - (const char*)buf + 2)&~2) + (extra + bitstream_size + wordstream_size)
+        );
+
     return 0;
 }
 
@@ -643,17 +654,9 @@ static int decode_i2_frame(FourXContext *f, const uint8_t *buf, int length){
     int x, y, x2, y2;
     const int width= f->avctx->width;
     const int height= f->avctx->height;
-    const int mbs = (FFALIGN(width, 16) >> 4) * (FFALIGN(height, 16) >> 4);
     uint16_t *dst= (uint16_t*)f->current_picture.data[0];
     const int stride= f->current_picture.linesize[0]>>1;
     const uint8_t *buf_end = buf + length;
-    GetByteContext g3;
-
-    if(length < mbs * 8) {
-        av_log(f->avctx, AV_LOG_ERROR, "packet size too small\n");
-        return AVERROR_INVALIDDATA;
-    }
-    bytestream2_init(&g3, buf, length);
 
     for(y=0; y<height; y+=16){
         for(x=0; x<width; x+=16){
@@ -662,8 +665,8 @@ static int decode_i2_frame(FourXContext *f, const uint8_t *buf, int length){
                 return -1;
             memset(color, 0, sizeof(color));
 //warning following is purely guessed ...
-            color[0]= bytestream2_get_le16u(&g3);
-            color[1]= bytestream2_get_le16u(&g3);
+            color[0]= bytestream_get_le16(&buf);
+            color[1]= bytestream_get_le16(&buf);
 
             if(color[0]&0x8000) av_log(NULL, AV_LOG_ERROR, "unk bit 1\n");
             if(color[1]&0x8000) av_log(NULL, AV_LOG_ERROR, "unk bit 2\n");
@@ -671,7 +674,7 @@ static int decode_i2_frame(FourXContext *f, const uint8_t *buf, int length){
             color[2]= mix(color[0], color[1]);
             color[3]= mix(color[1], color[0]);
 
-            bits= bytestream2_get_le32u(&g3);
+            bits= bytestream_get_le32(&buf);
             for(y2=0; y2<16; y2++){
                 for(x2=0; x2<16; x2++){
                     int index= 2*(x2>>2) + 8*(y2>>2);
@@ -680,7 +683,7 @@ static int decode_i2_frame(FourXContext *f, const uint8_t *buf, int length){
             }
             dst+=16;
         }
-        dst += 16 * stride - x;
+        dst += 16*stride - width;
     }
 
     return 0;
@@ -690,17 +693,16 @@ static int decode_i_frame(FourXContext *f, const uint8_t *buf, int length){
     int x, y;
     const int width= f->avctx->width;
     const int height= f->avctx->height;
+    uint16_t *dst= (uint16_t*)f->current_picture.data[0];
+    const int stride= f->current_picture.linesize[0]>>1;
     const unsigned int bitstream_size= AV_RL32(buf);
     unsigned int prestream_size;
     const uint8_t *prestream;
 
-    if (bitstream_size > (1<<26) || length < bitstream_size + 12) {
-        av_log(f->avctx, AV_LOG_ERROR, "packet size too small\n");
-        return AVERROR_INVALIDDATA;
-    }
-
-    prestream_size = 4 * AV_RL32(buf + bitstream_size + 4);
-    prestream      = buf + bitstream_size + 12;
+    if (bitstream_size > (1<<26) || length < bitstream_size + 12)
+        return -1;
+    prestream_size = 4*AV_RL32(buf + bitstream_size + 4);
+    prestream = buf + bitstream_size + 12;
 
     if (prestream_size > (1<<26) ||
         prestream_size != length - (bitstream_size + 12)){
@@ -732,6 +734,7 @@ static int decode_i_frame(FourXContext *f, const uint8_t *buf, int length){
 
             idct_put(f, x, y);
         }
+        dst += 16*stride;
     }
 
     if(get_vlc2(&f->pre_gb, f->pre_vlc.table, ACDC_VLC_BITS, 3) != 256)
@@ -831,7 +834,7 @@ static int decode_frame(AVCodecContext *avctx,
 
     if(frame_4cc == AV_RL32("ifr2")){
         p->pict_type= AV_PICTURE_TYPE_I;
-        if(decode_i2_frame(f, buf-4, frame_size + 4) < 0) {
+        if(decode_i2_frame(f, buf-4, frame_size+4) < 0){
             av_log(f->avctx, AV_LOG_ERROR, "decode i2 frame failed\n");
             return -1;
         }

libavcodec/8svx.c

@@ -106,11 +106,12 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
 {
     EightSvxContext *esc = avctx->priv_data;
     int n, out_data_size, ret;
+    uint8_t *out_date;
     uint8_t *src, *dst;
 
     /* decode and interleave the first packet */
     if (!esc->samples && avpkt) {
-        uint8_t *deinterleaved_samples, *p = NULL;
+        uint8_t *deinterleaved_samples;
 
         esc->samples_size = avctx->codec->id == CODEC_ID_8SVX_RAW || avctx->codec->id ==CODEC_ID_PCM_S8_PLANAR?
             avpkt->size : avctx->channels + (avpkt->size-avctx->channels) * 2;
@@ -129,7 +130,6 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
             }
             if (!(deinterleaved_samples = av_mallocz(n)))
                 return AVERROR(ENOMEM);
-            p = deinterleaved_samples;
 
             /* the uncompressed starting value is contained in the first byte */
             if (avctx->channels == 2) {
@@ -146,7 +146,6 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
             interleave_stereo(esc->samples, deinterleaved_samples, esc->samples_size);
         else
             memcpy(esc->samples, deinterleaved_samples, esc->samples_size);
-        av_freep(&p);
     }
 
     /* get output buffer */

libavcodec/Makefile

@@ -91,8 +91,6 @@ OBJS-$(CONFIG_ATRAC1_DECODER)          += atrac1.o atrac.o
 OBJS-$(CONFIG_ATRAC3_DECODER)          += atrac3.o atrac.o
 OBJS-$(CONFIG_AURA_DECODER)            += cyuv.o
 OBJS-$(CONFIG_AURA2_DECODER)           += aura.o
-OBJS-$(CONFIG_AVRP_DECODER)            += r210dec.o
-OBJS-$(CONFIG_AVRP_ENCODER)            += r210enc.o
 OBJS-$(CONFIG_AVS_DECODER)             += avs.o
 OBJS-$(CONFIG_BETHSOFTVID_DECODER)     += bethsoftvideo.o
 OBJS-$(CONFIG_BFI_DECODER)             += bfi.o
@@ -158,7 +156,6 @@ OBJS-$(CONFIG_FFV1_DECODER)            += ffv1.o rangecoder.o
 OBJS-$(CONFIG_FFV1_ENCODER)            += ffv1.o rangecoder.o
 OBJS-$(CONFIG_FFVHUFF_DECODER)         += huffyuv.o
 OBJS-$(CONFIG_FFVHUFF_ENCODER)         += huffyuv.o
-OBJS-$(CONFIG_FFWAVESYNTH_DECODER)     += ffwavesynth.o
 OBJS-$(CONFIG_FLAC_DECODER)            += flacdec.o flacdata.o flac.o vorbis_data.o
 OBJS-$(CONFIG_FLAC_ENCODER)            += flacenc.o flacdata.o flac.o vorbis_data.o
 OBJS-$(CONFIG_FLASHSV_DECODER)         += flashsv.o
@@ -340,9 +337,7 @@ OBJS-$(CONFIG_QPEG_DECODER)            += qpeg.o
 OBJS-$(CONFIG_QTRLE_DECODER)           += qtrle.o
 OBJS-$(CONFIG_QTRLE_ENCODER)           += qtrleenc.o
 OBJS-$(CONFIG_R10K_DECODER)            += r210dec.o
-OBJS-$(CONFIG_R10K_ENCODER)            += r210enc.o
 OBJS-$(CONFIG_R210_DECODER)            += r210dec.o
-OBJS-$(CONFIG_R210_ENCODER)            += r210enc.o
 OBJS-$(CONFIG_RA_144_DECODER)          += ra144dec.o ra144.o celp_filters.o
 OBJS-$(CONFIG_RA_144_ENCODER)          += ra144enc.o ra144.o celp_filters.o
 OBJS-$(CONFIG_RA_288_DECODER)          += ra288.o celp_math.o celp_filters.o
@@ -419,8 +414,6 @@ OBJS-$(CONFIG_ULTI_DECODER)            += ulti.o
 OBJS-$(CONFIG_UTVIDEO_DECODER)         += utvideo.o
 OBJS-$(CONFIG_V210_DECODER)            += v210dec.o
 OBJS-$(CONFIG_V210_ENCODER)            += v210enc.o
-OBJS-$(CONFIG_V308_DECODER)            += v308dec.o
-OBJS-$(CONFIG_V308_ENCODER)            += v308enc.o
 OBJS-$(CONFIG_V410_DECODER)            += v410dec.o
 OBJS-$(CONFIG_V410_ENCODER)            += v410enc.o
 OBJS-$(CONFIG_V210X_DECODER)           += v210x.o
@@ -473,13 +466,9 @@ OBJS-$(CONFIG_XBIN_DECODER)            += bintext.o cga_data.o
 OBJS-$(CONFIG_XL_DECODER)              += xl.o
 OBJS-$(CONFIG_XSUB_DECODER)            += xsubdec.o
 OBJS-$(CONFIG_XSUB_ENCODER)            += xsubenc.o
-OBJS-$(CONFIG_XWD_DECODER)             += xwddec.o
-OBJS-$(CONFIG_XWD_ENCODER)             += xwdenc.o
 OBJS-$(CONFIG_Y41P_DECODER)            += y41pdec.o
 OBJS-$(CONFIG_Y41P_ENCODER)            += y41penc.o
 OBJS-$(CONFIG_YOP_DECODER)             += yop.o
-OBJS-$(CONFIG_YUV4_DECODER)            += yuv4dec.o
-OBJS-$(CONFIG_YUV4_ENCODER)            += yuv4enc.o
 OBJS-$(CONFIG_ZLIB_DECODER)            += lcldec.o
 OBJS-$(CONFIG_ZLIB_ENCODER)            += lclenc.o
 OBJS-$(CONFIG_ZMBV_DECODER)            += zmbv.o
@@ -538,20 +527,19 @@ OBJS-$(CONFIG_PCM_ZORK_DECODER)           += pcm.o
 
 OBJS-$(CONFIG_ADPCM_4XM_DECODER)          += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_ADX_DECODER)          += adxdec.o adx.o
-OBJS-$(CONFIG_ADPCM_ADX_ENCODER)          += adxenc.o adx.o
+OBJS-$(CONFIG_ADPCM_ADX_ENCODER)          += adxenc.o
 OBJS-$(CONFIG_ADPCM_CT_DECODER)           += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_EA_DECODER)           += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_EA_MAXIS_XA_DECODER)  += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_EA_R1_DECODER)        += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_EA_R2_DECODER)        += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_EA_R3_DECODER)        += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_EA_XAS_DECODER)       += adpcm.o adpcm_data.o
+OBJS-$(CONFIG_ADPCM_EA_DECODER)           += adpcm.o
+OBJS-$(CONFIG_ADPCM_EA_MAXIS_XA_DECODER)  += adpcm.o
+OBJS-$(CONFIG_ADPCM_EA_R1_DECODER)        += adpcm.o
+OBJS-$(CONFIG_ADPCM_EA_R2_DECODER)        += adpcm.o
+OBJS-$(CONFIG_ADPCM_EA_R3_DECODER)        += adpcm.o
+OBJS-$(CONFIG_ADPCM_EA_XAS_DECODER)       += adpcm.o
 OBJS-$(CONFIG_ADPCM_G722_DECODER)         += g722.o g722dec.o
 OBJS-$(CONFIG_ADPCM_G722_ENCODER)         += g722.o g722enc.o
 OBJS-$(CONFIG_ADPCM_G726_DECODER)         += g726.o
 OBJS-$(CONFIG_ADPCM_G726_ENCODER)         += g726.o
 OBJS-$(CONFIG_ADPCM_IMA_AMV_DECODER)      += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_IMA_APC_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_DK3_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_DK4_DECODER)      += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_EA_EACS_DECODER)  += adpcm.o adpcm_data.o
@@ -565,13 +553,13 @@ OBJS-$(CONFIG_ADPCM_IMA_WAV_ENCODER)      += adpcmenc.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_IMA_WS_DECODER)       += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_MS_DECODER)           += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_MS_ENCODER)           += adpcmenc.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_SBPRO_2_DECODER)      += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_SBPRO_3_DECODER)      += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_SBPRO_4_DECODER)      += adpcm.o adpcm_data.o
+OBJS-$(CONFIG_ADPCM_SBPRO_2_DECODER)      += adpcm.o
+OBJS-$(CONFIG_ADPCM_SBPRO_3_DECODER)      += adpcm.o
+OBJS-$(CONFIG_ADPCM_SBPRO_4_DECODER)      += adpcm.o
 OBJS-$(CONFIG_ADPCM_SWF_DECODER)          += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_SWF_ENCODER)          += adpcmenc.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_THP_DECODER)          += adpcm.o adpcm_data.o
-OBJS-$(CONFIG_ADPCM_XA_DECODER)           += adpcm.o adpcm_data.o
+OBJS-$(CONFIG_ADPCM_THP_DECODER)          += adpcm.o
+OBJS-$(CONFIG_ADPCM_XA_DECODER)           += adpcm.o
 OBJS-$(CONFIG_ADPCM_YAMAHA_DECODER)       += adpcm.o adpcm_data.o
 OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER)       += adpcmenc.o adpcm_data.o
 
@@ -594,7 +582,7 @@ OBJS-$(CONFIG_MATROSKA_MUXER)          += xiph.o mpeg4audio.o \
                                           flacdec.o flacdata.o flac.o \
                                           mpegaudiodata.o vorbis_data.o
 OBJS-$(CONFIG_MP3_MUXER)               += mpegaudiodata.o mpegaudiodecheader.o
-OBJS-$(CONFIG_MOV_DEMUXER)             += mpeg4audio.o mpegaudiodata.o timecode.o
+OBJS-$(CONFIG_MOV_DEMUXER)             += mpeg4audio.o mpegaudiodata.o
 OBJS-$(CONFIG_MOV_MUXER)               += mpeg4audio.o mpegaudiodata.o
 OBJS-$(CONFIG_MPEGTS_MUXER)            += mpegvideo.o mpeg4audio.o
 OBJS-$(CONFIG_MPEGTS_DEMUXER)          += mpeg4audio.o mpegaudiodata.o
@@ -661,7 +649,6 @@ OBJS-$(CONFIG_DVBSUB_PARSER)           += dvbsub_parser.o
 OBJS-$(CONFIG_DVDSUB_PARSER)           += dvdsub_parser.o
 OBJS-$(CONFIG_FLAC_PARSER)             += flac_parser.o flacdata.o flac.o \
                                           vorbis_data.o
-OBJS-$(CONFIG_GSM_PARSER)              += gsm_parser.o
 OBJS-$(CONFIG_H261_PARSER)             += h261_parser.o
 OBJS-$(CONFIG_H263_PARSER)             += h263_parser.o
 OBJS-$(CONFIG_H264_PARSER)             += h264_parser.o h264.o            \
@@ -721,6 +708,8 @@ OBJS-$(CONFIG_MLIB)                    += mlib/dsputil_mlib.o           \
 # well.
 OBJS-$(!CONFIG_SMALL)                  += inverse.o
 
+-include $(SRC_PATH)/$(SUBDIR)$(ARCH)/Makefile
+
 SKIPHEADERS                            += %_tablegen.h                  \
                                           %_tables.h                    \
                                           aac_tablegen_decl.h           \
@@ -748,6 +737,8 @@ DIRS = alpha arm bfin mlib ppc ps2 sh4 sparc x86
 
 CLEANFILES = *_tables.c *_tables.h *_tablegen$(HOSTEXESUF)
 
+include $(SRC_PATH)/subdir.mak
+
 $(SUBDIR)dct-test$(EXESUF): $(SUBDIR)dctref.o
 
 TRIG_TABLES  = cos cos_fixed sin

libavcodec/aac.h

@@ -84,7 +84,6 @@ enum BandType {
 #define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10)
 
 enum ChannelPosition {
-    AAC_CHANNEL_OFF   = 0,
     AAC_CHANNEL_FRONT = 1,
     AAC_CHANNEL_SIDE  = 2,
     AAC_CHANNEL_BACK  = 3,

libavcodec/aaccoder.c

@@ -110,15 +110,14 @@ static av_always_inline float quantize_and_encode_band_cost_template(
                                 int *bits, int BT_ZERO, int BT_UNSIGNED,
                                 int BT_PAIR, int BT_ESC)
 {
-    const int q_idx = POW_SF2_ZERO - scale_idx + SCALE_ONE_POS - SCALE_DIV_512;
-    const float Q   = ff_aac_pow2sf_tab [q_idx];
-    const float Q34 = ff_aac_pow34sf_tab[q_idx];
-    const float IQ  = ff_aac_pow2sf_tab [POW_SF2_ZERO + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
+    const float IQ = ff_aac_pow2sf_tab[POW_SF2_ZERO + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
+    const float  Q = ff_aac_pow2sf_tab[POW_SF2_ZERO - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
     const float CLIPPED_ESCAPE = 165140.0f*IQ;
     int i, j;
     float cost = 0;
     const int dim = BT_PAIR ? 2 : 4;
     int resbits = 0;
+    const float  Q34 = sqrtf(Q * sqrtf(Q));
     const int range  = aac_cb_range[cb];
     const int maxval = aac_cb_maxval[cb];
     int off;
@@ -421,7 +420,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
     const int run_esc  = (1 << run_bits) - 1;
     int idx, ppos, count;
     int stackrun[120], stackcb[120], stack_len;
-    float next_minbits = INFINITY;
+    float next_minrd = INFINITY;
     int next_mincb = 0;
 
     abs_pow34_v(s->scoefs, sce->coeffs, 1024);
@@ -435,7 +434,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
         size = sce->ics.swb_sizes[swb];
         if (sce->zeroes[win*16 + swb]) {
             float cost_stay_here = path[swb][0].cost;
-            float cost_get_here  = next_minbits + run_bits + 4;
+            float cost_get_here  = next_minrd + run_bits + 4;
             if (   run_value_bits[sce->ics.num_windows == 8][path[swb][0].run]
                 != run_value_bits[sce->ics.num_windows == 8][path[swb][0].run+1])
                 cost_stay_here += run_bits;
@@ -448,7 +447,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
                 path[swb+1][0].cost     = cost_stay_here;
                 path[swb+1][0].run      = path[swb][0].run + 1;
             }
-            next_minbits = path[swb+1][0].cost;
+            next_minrd = path[swb+1][0].cost;
             next_mincb = 0;
             for (cb = 1; cb < 12; cb++) {
                 path[swb+1][cb].cost = 61450;
@@ -456,10 +455,10 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
                 path[swb+1][cb].run = 0;
             }
         } else {
-            float minbits = next_minbits;
+            float minrd = next_minrd;
             int mincb = next_mincb;
             int startcb = sce->band_type[win*16+swb];
-            next_minbits = INFINITY;
+            next_minrd = INFINITY;
             next_mincb = 0;
             for (cb = 0; cb < startcb; cb++) {
                 path[swb+1][cb].cost = 61450;
@@ -468,15 +467,15 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
             }
             for (cb = startcb; cb < 12; cb++) {
                 float cost_stay_here, cost_get_here;
-                float bits = 0.0f;
+                float rd = 0.0f;
                 for (w = 0; w < group_len; w++) {
-                    bits += quantize_band_cost(s, sce->coeffs + start + w*128,
-                                               s->scoefs + start + w*128, size,
-                                               sce->sf_idx[(win+w)*16+swb], cb,
-                                               0, INFINITY, NULL);
+                    rd += quantize_band_cost(s, sce->coeffs + start + w*128,
+                                             s->scoefs + start + w*128, size,
+                                             sce->sf_idx[(win+w)*16+swb], cb,
+                                             0, INFINITY, NULL);
                 }
-                cost_stay_here = path[swb][cb].cost + bits;
-                cost_get_here  = minbits            + bits + run_bits + 4;
+                cost_stay_here = path[swb][cb].cost + rd;
+                cost_get_here  = minrd              + rd + run_bits + 4;
                 if (   run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run]
                     != run_value_bits[sce->ics.num_windows == 8][path[swb][cb].run+1])
                     cost_stay_here += run_bits;
@@ -489,8 +488,8 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
                     path[swb+1][cb].cost     = cost_stay_here;
                     path[swb+1][cb].run      = path[swb][cb].run + 1;
                 }
-                if (path[swb+1][cb].cost < next_minbits) {
-                    next_minbits = path[swb+1][cb].cost;
+                if (path[swb+1][cb].cost < next_minrd) {
+                    next_minrd = path[swb+1][cb].cost;
                     next_mincb = cb;
                 }
             }

libavcodec/aacdec.c

@@ -98,7 +98,6 @@
 #include "aacsbr.h"
 #include "mpeg4audio.h"
 #include "aacadtsdec.h"
-#include "libavutil/intfloat.h"
 
 #include <assert.h>
 #include <errno.h>
@@ -109,6 +108,11 @@
 #   include "arm/aac.h"
 #endif
 
+union float754 {
+    float f;
+    uint32_t i;
+};
+
 static VLC vlc_scalefactors;
 static VLC vlc_spectral[11];
 
@@ -163,19 +167,6 @@ static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
     }
 }
 
-static int count_channels(enum ChannelPosition che_pos[4][MAX_ELEM_ID])
-{
-    int i, type, sum = 0;
-    for (i = 0; i < MAX_ELEM_ID; i++) {
-        for (type = 0; type < 4; type++) {
-            sum += (1 + (type == TYPE_CPE)) *
-                (che_pos[type][i] != AAC_CHANNEL_OFF &&
-                 che_pos[type][i] != AAC_CHANNEL_CC);
-        }
-    }
-    return sum;
-}
-
 /**
  * Check for the channel element in the current channel position configuration.
  * If it exists, make sure the appropriate element is allocated and map the
@@ -450,12 +441,6 @@ static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
         if ((ret = set_default_channel_config(avctx, new_che_pos, channel_config)))
             return ret;
     }
-
-    if (count_channels(new_che_pos) > 1) {
-        m4ac->ps = 0;
-    } else if (m4ac->sbr == 1 && m4ac->ps == -1)
-        m4ac->ps = 1;
-
     if (ac && (ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR)))
         return ret;
 
@@ -514,6 +499,8 @@ static int decode_audio_specific_config(AACContext *ac,
         av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index);
         return -1;
     }
+    if (m4ac->sbr == 1 && m4ac->ps == -1)
+        m4ac->ps = 1;
 
     skip_bits_long(&gb, i);
 
@@ -739,13 +726,16 @@ static void decode_ltp(AACContext *ac, LongTermPrediction *ltp,
 
 /**
  * Decode Individual Channel Stream info; reference: table 4.6.
+ *
+ * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
  */
 static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
-                           GetBitContext *gb)
+                           GetBitContext *gb, int common_window)
 {
     if (get_bits1(gb)) {
         av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
-        return AVERROR_INVALIDDATA;
+        memset(ics, 0, sizeof(IndividualChannelStream));
+        return -1;
     }
     ics->window_sequence[1] = ics->window_sequence[0];
     ics->window_sequence[0] = get_bits(gb, 2);
@@ -780,11 +770,13 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
         if (ics->predictor_present) {
             if (ac->m4ac.object_type == AOT_AAC_MAIN) {
                 if (decode_prediction(ac, ics, gb)) {
-                    return AVERROR_INVALIDDATA;
+                    memset(ics, 0, sizeof(IndividualChannelStream));
+                    return -1;
                 }
             } else if (ac->m4ac.object_type == AOT_AAC_LC) {
                 av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
-                return AVERROR_INVALIDDATA;
+                memset(ics, 0, sizeof(IndividualChannelStream));
+                return -1;
             } else {
                 if ((ics->ltp.present = get_bits(gb, 1)))
                     decode_ltp(ac, &ics->ltp, gb, ics->max_sfb);
@@ -796,7 +788,8 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
         av_log(ac->avctx, AV_LOG_ERROR,
                "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
                ics->max_sfb, ics->num_swb);
-        return AVERROR_INVALIDDATA;
+        memset(ics, 0, sizeof(IndividualChannelStream));
+        return -1;
     }
 
     return 0;
@@ -1030,7 +1023,7 @@ static inline float *VMUL4(float *dst, const float *v, unsigned idx,
 static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
                             unsigned sign, const float *scale)
 {
-    union av_intfloat32 s0, s1;
+    union float754 s0, s1;
 
     s0.f = s1.f = *scale;
     s0.i ^= sign >> 1 << 31;
@@ -1048,8 +1041,8 @@ static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
                             unsigned sign, const float *scale)
 {
     unsigned nz = idx >> 12;
-    union av_intfloat32 s = { .f = *scale };
-    union av_intfloat32 t;
+    union float754 s = { .f = *scale };
+    union float754 t;
 
     t.i = s.i ^ (sign & 1U<<31);
     *dst++ = v[idx    & 3] * t.f;
@@ -1298,7 +1291,7 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
 
 static av_always_inline float flt16_round(float pf)
 {
-    union av_intfloat32 tmp;
+    union float754 tmp;
     tmp.f = pf;
     tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
     return tmp.f;
@@ -1306,7 +1299,7 @@ static av_always_inline float flt16_round(float pf)
 
 static av_always_inline float flt16_even(float pf)
 {
-    union av_intfloat32 tmp;
+    union float754 tmp;
     tmp.f = pf;
     tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
     return tmp.f;
@@ -1314,7 +1307,7 @@ static av_always_inline float flt16_even(float pf)
 
 static av_always_inline float flt16_trunc(float pf)
 {
-    union av_intfloat32 pun;
+    union float754 pun;
     pun.f = pf;
     pun.i &= 0xFFFF0000U;
     return pun.f;
@@ -1401,8 +1394,8 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
     global_gain = get_bits(gb, 8);
 
     if (!common_window && !scale_flag) {
-        if (decode_ics_info(ac, ics, gb) < 0)
-            return AVERROR_INVALIDDATA;
+        if (decode_ics_info(ac, ics, gb, 0) < 0)
+            return -1;
     }
 
     if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
@@ -1518,8 +1511,8 @@ static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
 
     common_window = get_bits1(gb);
     if (common_window) {
-        if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
-            return AVERROR_INVALIDDATA;
+        if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
+            return -1;
         i = cpe->ch[1].ics.use_kb_window[0];
         cpe->ch[1].ics = cpe->ch[0].ics;
         cpe->ch[1].ics.use_kb_window[1] = i;
@@ -2110,7 +2103,7 @@ static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
 
     size = avpriv_aac_parse_header(gb, &hdr_info);
     if (size > 0) {
-        if (hdr_info.chan_config) {
+        if (hdr_info.chan_config && (hdr_info.chan_config!=ac->m4ac.chan_config || ac->m4ac.sample_rate!=hdr_info.sample_rate)) {
             enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
             memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
             ac->m4ac.chan_config = hdr_info.chan_config;
@@ -2218,11 +2211,10 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
             if ((err = decode_pce(avctx, &ac->m4ac, new_che_pos, gb)))
                 break;
             if (ac->output_configured > OC_TRIAL_PCE)
-                av_log(avctx, AV_LOG_INFO,
-                       "Evaluating a further program_config_element.\n");
-            err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
-            if (!err)
-                ac->m4ac.chan_config = 0;
+                av_log(avctx, AV_LOG_ERROR,
+                       "Not evaluating a further program_config_element as this construct is dubious at best.\n");
+            else
+                err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
             break;
         }
 
@@ -2294,31 +2286,12 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
 static int aac_decode_frame(AVCodecContext *avctx, void *data,
                             int *got_frame_ptr, AVPacket *avpkt)
 {
-    AACContext *ac = avctx->priv_data;
     const uint8_t *buf = avpkt->data;
     int buf_size = avpkt->size;
     GetBitContext gb;
     int buf_consumed;
     int buf_offset;
     int err;
-    int new_extradata_size;
-    const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
-                                       AV_PKT_DATA_NEW_EXTRADATA,
-                                       &new_extradata_size);
-
-    if (new_extradata) {
-        av_free(avctx->extradata);
-        avctx->extradata = av_mallocz(new_extradata_size +
-                                      FF_INPUT_BUFFER_PADDING_SIZE);
-        if (!avctx->extradata)
-            return AVERROR(ENOMEM);
-        avctx->extradata_size = new_extradata_size;
-        memcpy(avctx->extradata, new_extradata, new_extradata_size);
-        if (decode_audio_specific_config(ac, ac->avctx, &ac->m4ac,
-                                         avctx->extradata,
-                                         avctx->extradata_size*8, 1) < 0)
-            return AVERROR_INVALIDDATA;
-    }
 
     init_get_bits(&gb, buf, buf_size * 8);
 
@@ -2393,8 +2366,6 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
                                "config not byte aligned.\n", 1);
         return AVERROR_INVALIDDATA;
     }
-    if (asclen <= 0)
-        return AVERROR_INVALIDDATA;
     bits_consumed = decode_audio_specific_config(NULL, avctx, &m4ac,
                                          gb->buffer + (config_start_bit / 8),
                                          asclen, sync_extension);

libavcodec/aacenc.c

@@ -46,14 +46,6 @@
 
 #define AAC_MAX_CHANNELS 6
 
-#define ERROR_IF(cond, ...) \
-    if (cond) { \
-        av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
-        return AVERROR(EINVAL); \
-    }
-
-float ff_aac_pow34sf_tab[428];
-
 static const uint8_t swb_size_1024_96[] = {
     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
     12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
@@ -143,10 +135,7 @@ static const uint8_t aac_chan_configs[6][5] = {
  {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
 };
 
-/**
- * Table to remap channels from Libav's default order to AAC order.
- */
-static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
+static const uint8_t channel_maps[][AAC_MAX_CHANNELS] = {
     { 0 },
     { 0, 1 },
     { 2, 0, 1 },
@@ -167,7 +156,7 @@ static void put_audio_specific_config(AVCodecContext *avctx)
     init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
     put_bits(&pb, 5, 2); //object type - AAC-LC
     put_bits(&pb, 4, s->samplerate_index); //sample rate index
-    put_bits(&pb, 4, s->channels);
+    put_bits(&pb, 4, avctx->channels);
     //GASpecificConfig
     put_bits(&pb, 1, 0); //frame length - 1024 samples
     put_bits(&pb, 1, 0); //does not depend on core coder
@@ -180,80 +169,117 @@ static void put_audio_specific_config(AVCodecContext *avctx)
     flush_put_bits(&pb);
 }
 
-#define WINDOW_FUNC(type) \
-static void apply_ ##type ##_window(DSPContext *dsp, SingleChannelElement *sce, const float *audio)
-
-WINDOW_FUNC(only_long)
-{
-    const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
-    const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
-    float *out = sce->ret;
-
-    dsp->vector_fmul        (out,        audio,        lwindow, 1024);
-    dsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
-}
-
-WINDOW_FUNC(long_start)
-{
-    const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
-    const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
-    float *out = sce->ret;
-
-    dsp->vector_fmul(out, audio, lwindow, 1024);
-    memcpy(out + 1024, audio, sizeof(out[0]) * 448);
-    dsp->vector_fmul_reverse(out + 1024 + 448, audio, swindow, 128);
-    memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
-}
-
-WINDOW_FUNC(long_stop)
-{
-    const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
-    const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
-    float *out = sce->ret;
-
-    memset(out, 0, sizeof(out[0]) * 448);
-    dsp->vector_fmul(out + 448, audio + 448, swindow, 128);
-    memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
-    dsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
-}
-
-WINDOW_FUNC(eight_short)
-{
-    const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
-    const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
-    const float *in = audio + 448;
-    float *out = sce->ret;
-
-    for (int w = 0; w < 8; w++) {
-        dsp->vector_fmul        (out, in, w ? pwindow : swindow, 128);
-        out += 128;
-        in  += 128;
-        dsp->vector_fmul_reverse(out, in, swindow, 128);
-        out += 128;
-    }
-}
-
-static void (*const apply_window[4])(DSPContext *dsp, SingleChannelElement *sce, const float *audio) = {
-    [ONLY_LONG_SEQUENCE]   = apply_only_long_window,
-    [LONG_START_SEQUENCE]  = apply_long_start_window,
-    [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
-    [LONG_STOP_SEQUENCE]   = apply_long_stop_window
-};
-
-static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
-                                  float *audio)
+static av_cold int aac_encode_init(AVCodecContext *avctx)
 {
+    AACEncContext *s = avctx->priv_data;
     int i;
+    const uint8_t *sizes[2];
+    uint8_t grouping[AAC_MAX_CHANNELS];
+    int lengths[2];
+
+    avctx->frame_size = 1024;
+
+    for (i = 0; i < 16; i++)
+        if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
+            break;
+    if (i == 16) {
+        av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
+        return -1;
+    }
+    if (avctx->channels > AAC_MAX_CHANNELS) {
+        av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
+        return -1;
+    }
+    if (avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW) {
+        av_log(avctx, AV_LOG_ERROR, "Unsupported profile %d\n", avctx->profile);
+        return -1;
+    }
+    if (1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * avctx->channels) {
+        av_log(avctx, AV_LOG_ERROR, "Too many bits per frame requested\n");
+        return -1;
+    }
+    s->samplerate_index = i;
+
+    dsputil_init(&s->dsp, avctx);
+    ff_mdct_init(&s->mdct1024, 11, 0, 1.0);
+    ff_mdct_init(&s->mdct128,   8, 0, 1.0);
+    // window init
+    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
+    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
+    ff_init_ff_sine_windows(10);
+    ff_init_ff_sine_windows(7);
+
+    s->chan_map           = aac_chan_configs[avctx->channels-1];
+    s->samples            = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
+    s->cpe                = av_mallocz(sizeof(ChannelElement) * s->chan_map[0]);
+    avctx->extradata      = av_mallocz(5 + FF_INPUT_BUFFER_PADDING_SIZE);
+    avctx->extradata_size = 5;
+    put_audio_specific_config(avctx);
+
+    sizes[0]   = swb_size_1024[i];
+    sizes[1]   = swb_size_128[i];
+    lengths[0] = ff_aac_num_swb_1024[i];
+    lengths[1] = ff_aac_num_swb_128[i];
+    for (i = 0; i < s->chan_map[0]; i++)
+        grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
+    ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping);
+    s->psypp = ff_psy_preprocess_init(avctx);
+    s->coder = &ff_aac_coders[s->options.aac_coder];
+
+    s->lambda = avctx->global_quality ? avctx->global_quality : 120;
+
+    ff_aac_tableinit();
+
+    return 0;
+}
+
+static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
+                                  SingleChannelElement *sce, short *audio)
+{
+    int i, k;
+    const int chans = avctx->channels;
+    const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
+    const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
+    const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
     float *output = sce->ret;
 
-    apply_window[sce->ics.window_sequence[0]](&s->dsp, sce, audio);
-
-    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
+    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
+        memcpy(output, sce->saved, sizeof(float)*1024);
+        if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) {
+            memset(output, 0, sizeof(output[0]) * 448);
+            for (i = 448; i < 576; i++)
+                output[i] = sce->saved[i] * pwindow[i - 448];
+            for (i = 576; i < 704; i++)
+                output[i] = sce->saved[i];
+        }
+        if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
+            for (i = 0; i < 1024; i++) {
+                output[i+1024]         = audio[i * chans] * lwindow[1024 - i - 1];
+                sce->saved[i] = audio[i * chans] * lwindow[i];
+            }
+        } else {
+            for (i = 0; i < 448; i++)
+                output[i+1024]         = audio[i * chans];
+            for (; i < 576; i++)
+                output[i+1024]         = audio[i * chans] * swindow[576 - i - 1];
+            memset(output+1024+576, 0, sizeof(output[0]) * 448);
+            for (i = 0; i < 1024; i++)
+                sce->saved[i] = audio[i * chans];
+        }
         s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
-    else
-        for (i = 0; i < 1024; i += 128)
-            s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2);
-    memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
+    } else {
+        for (k = 0; k < 1024; k += 128) {
+            for (i = 448 + k; i < 448 + k + 256; i++)
+                output[i - 448 - k] = (i < 1024)
+                                         ? sce->saved[i]
+                                         : audio[(i-1024)*chans];
+            s->dsp.vector_fmul        (output,     output, k ?  swindow : pwindow, 128);
+            s->dsp.vector_fmul_reverse(output+128, output+128, swindow, 128);
+            s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + k, output);
+        }
+        for (i = 0; i < 1024; i++)
+            sce->saved[i] = audio[i * chans];
+    }
 }
 
 /**
@@ -462,46 +488,20 @@ static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
     put_bits(&s->pb, 3, TYPE_FIL);
     put_bits(&s->pb, 4, FFMIN(namelen, 15));
     if (namelen >= 15)
-        put_bits(&s->pb, 8, namelen - 14);
+        put_bits(&s->pb, 8, namelen - 16);
     put_bits(&s->pb, 4, 0); //extension type - filler
-    padbits = -put_bits_count(&s->pb) & 7;
+    padbits = 8 - (put_bits_count(&s->pb) & 7);
     avpriv_align_put_bits(&s->pb);
     for (i = 0; i < namelen - 2; i++)
         put_bits(&s->pb, 8, name[i]);
     put_bits(&s->pb, 12 - padbits, 0);
 }
 
-/*
- * Deinterleave input samples.
- * Channels are reordered from Libav's default order to AAC order.
- */
-static void deinterleave_input_samples(AACEncContext *s,
-                                       const float *samples)
-{
-    int ch, i;
-    const int sinc = s->channels;
-    const uint8_t *channel_map = aac_chan_maps[sinc - 1];
-
-    /* deinterleave and remap input samples */
-    for (ch = 0; ch < sinc; ch++) {
-        const float *sptr = samples + channel_map[ch];
-
-        /* copy last 1024 samples of previous frame to the start of the current frame */
-        memcpy(&s->planar_samples[ch][0], &s->planar_samples[ch][1024], 1024 * sizeof(s->planar_samples[0][0]));
-
-        /* deinterleave */
-        for (i = 1024; i < 1024 * 2; i++) {
-            s->planar_samples[ch][i] = *sptr;
-            sptr += sinc;
-        }
-    }
-}
-
 static int aac_encode_frame(AVCodecContext *avctx,
                             uint8_t *frame, int buf_size, void *data)
 {
     AACEncContext *s = avctx->priv_data;
-    float **samples = s->planar_samples, *samples2, *la, *overlap;
+    int16_t *samples = s->samples, *samples2, *la;
     ChannelElement *cpe;
     int i, ch, w, g, chans, tag, start_ch;
     int chan_el_counter[4];
@@ -509,15 +509,36 @@ static int aac_encode_frame(AVCodecContext *avctx,
 
     if (s->last_frame)
         return 0;
-
     if (data) {
-        deinterleave_input_samples(s, data);
-        if (s->psypp)
-            ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
+        if (!s->psypp) {
+            if (avctx->channels <= 2) {
+                memcpy(s->samples + 1024 * avctx->channels, data,
+                       1024 * avctx->channels * sizeof(s->samples[0]));
+            } else {
+                for (i = 0; i < 1024; i++)
+                    for (ch = 0; ch < avctx->channels; ch++)
+                        s->samples[(i + 1024) * avctx->channels + ch] =
+                            ((int16_t*)data)[i * avctx->channels +
+                                             channel_maps[avctx->channels-1][ch]];
+            }
+        } else {
+            start_ch = 0;
+            samples2 = s->samples + 1024 * avctx->channels;
+            for (i = 0; i < s->chan_map[0]; i++) {
+                tag = s->chan_map[i+1];
+                chans = tag == TYPE_CPE ? 2 : 1;
+                ff_psy_preprocess(s->psypp,
+                                  (uint16_t*)data + channel_maps[avctx->channels-1][start_ch],
+                                  samples2 + start_ch, start_ch, chans);
+                start_ch += chans;
+            }
+        }
     }
-
-    if (!avctx->frame_number)
+    if (!avctx->frame_number) {
+        memcpy(s->samples, s->samples + 1024 * avctx->channels,
+               1024 * avctx->channels * sizeof(s->samples[0]));
         return 0;
+    }
 
     start_ch = 0;
     for (i = 0; i < s->chan_map[0]; i++) {
@@ -528,9 +549,8 @@ static int aac_encode_frame(AVCodecContext *avctx,
         for (ch = 0; ch < chans; ch++) {
             IndividualChannelStream *ics = &cpe->ch[ch].ics;
             int cur_channel = start_ch + ch;
-            overlap  = &samples[cur_channel][0];
-            samples2 = overlap + 1024;
-            la       = samples2 + (448+64);
+            samples2 = samples + cur_channel;
+            la       = samples2 + (448+64) * avctx->channels;
             if (!data)
                 la = NULL;
             if (tag == TYPE_LFE) {
@@ -558,7 +578,7 @@ static int aac_encode_frame(AVCodecContext *avctx,
             for (w = 0; w < ics->num_windows; w++)
                 ics->group_len[w] = wi[ch].grouping[w];
 
-            apply_window_and_mdct(s, &cpe->ch[ch], overlap);
+            apply_window_and_mdct(avctx, s, &cpe->ch[ch], samples2);
         }
         start_ch += chans;
     }
@@ -624,8 +644,8 @@ static int aac_encode_frame(AVCodecContext *avctx,
         }
 
         frame_bits = put_bits_count(&s->pb);
-        if (frame_bits <= 6144 * s->channels - 3) {
-            s->psy.bitres.bits = frame_bits / s->channels;
+        if (frame_bits <= 6144 * avctx->channels - 3) {
+            s->psy.bitres.bits = frame_bits / avctx->channels;
             break;
         }
 
@@ -646,7 +666,8 @@ static int aac_encode_frame(AVCodecContext *avctx,
 
     if (!data)
         s->last_frame = 1;
-
+    memcpy(s->samples, s->samples + 1024 * avctx->channels,
+           1024 * avctx->channels * sizeof(s->samples[0]));
     return put_bits_count(&s->pb)>>3;
 }
 
@@ -657,109 +678,12 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)
     ff_mdct_end(&s->mdct1024);
     ff_mdct_end(&s->mdct128);
     ff_psy_end(&s->psy);
-    if (s->psypp)
-        ff_psy_preprocess_end(s->psypp);
-    av_freep(&s->buffer.samples);
+    ff_psy_preprocess_end(s->psypp);
+    av_freep(&s->samples);
     av_freep(&s->cpe);
     return 0;
 }
 
-static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
-{
-    int ret = 0;
-
-    dsputil_init(&s->dsp, avctx);
-
-    // window init
-    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
-    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
-    ff_init_ff_sine_windows(10);
-    ff_init_ff_sine_windows(7);
-
-    if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0))
-        return ret;
-    if (ret = ff_mdct_init(&s->mdct128,   8, 0, 32768.0))
-        return ret;
-
-    return 0;
-}
-
-static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
-{
-    FF_ALLOCZ_OR_GOTO(avctx, s->buffer.samples, 3 * 1024 * s->channels * sizeof(s->buffer.samples[0]), alloc_fail);
-    FF_ALLOCZ_OR_GOTO(avctx, s->cpe, sizeof(ChannelElement) * s->chan_map[0], alloc_fail);
-    FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
-
-    for(int ch = 0; ch < s->channels; ch++)
-        s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
-
-    return 0;
-alloc_fail:
-    return AVERROR(ENOMEM);
-}
-
-static av_cold int aac_encode_init(AVCodecContext *avctx)
-{
-    AACEncContext *s = avctx->priv_data;
-    int i, ret = 0;
-    const uint8_t *sizes[2];
-    uint8_t grouping[AAC_MAX_CHANNELS];
-    int lengths[2];
-
-    avctx->frame_size = 1024;
-
-    for (i = 0; i < 16; i++)
-        if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
-            break;
-
-    s->channels = avctx->channels;
-
-    ERROR_IF(i == 16,
-             "Unsupported sample rate %d\n", avctx->sample_rate);
-    ERROR_IF(s->channels > AAC_MAX_CHANNELS,
-             "Unsupported number of channels: %d\n", s->channels);
-    ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW,
-             "Unsupported profile %d\n", avctx->profile);
-    ERROR_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
-             "Too many bits per frame requested\n");
-
-    s->samplerate_index = i;
-
-    s->chan_map = aac_chan_configs[s->channels-1];
-
-    if (ret = dsp_init(avctx, s))
-        goto fail;
-
-    if (ret = alloc_buffers(avctx, s))
-        goto fail;
-
-    avctx->extradata_size = 5;
-    put_audio_specific_config(avctx);
-
-    sizes[0]   = swb_size_1024[i];
-    sizes[1]   = swb_size_128[i];
-    lengths[0] = ff_aac_num_swb_1024[i];
-    lengths[1] = ff_aac_num_swb_128[i];
-    for (i = 0; i < s->chan_map[0]; i++)
-        grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
-    if (ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping))
-        goto fail;
-    s->psypp = ff_psy_preprocess_init(avctx);
-    s->coder = &ff_aac_coders[s->options.aac_coder];
-
-    s->lambda = avctx->global_quality ? avctx->global_quality : 120;
-
-    ff_aac_tableinit();
-
-    for (i = 0; i < 428; i++)
-        ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));
-
-    return 0;
-fail:
-    aac_encode_end(avctx);
-    return ret;
-}
-
 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
 static const AVOption aacenc_options[] = {
     {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.dbl = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},
@@ -786,7 +710,7 @@ AVCodec ff_aac_encoder = {
     .encode         = aac_encode_frame,
     .close          = aac_encode_end,
     .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
-    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
+    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
     .priv_class = &aacenc_class,
 };

libavcodec/aacenc.h

@@ -61,10 +61,9 @@ typedef struct AACEncContext {
     FFTContext mdct1024;                         ///< long (1024 samples) frame transform context
     FFTContext mdct128;                          ///< short (128 samples) frame transform context
     DSPContext  dsp;
-    float *planar_samples[6];                    ///< saved preprocessed input
+    int16_t *samples;                            ///< saved preprocessed input
 
     int samplerate_index;                        ///< MPEG-4 samplerate index
-    int channels;                                ///< channel count
     const uint8_t *chan_map;                     ///< channel configuration map
 
     ChannelElement *cpe;                         ///< channel elements
@@ -76,12 +75,6 @@ typedef struct AACEncContext {
     float lambda;
     DECLARE_ALIGNED(16, int,   qcoefs)[96];      ///< quantized coefficients
     DECLARE_ALIGNED(32, float, scoefs)[1024];    ///< scaled coefficients
-
-    struct {
-        float *samples;
-    } buffer;
 } AACEncContext;
 
-extern float ff_aac_pow34sf_tab[428];
-
 #endif /* AVCODEC_AACENC_H */

libavcodec/aacps.c

@@ -223,7 +223,7 @@ int ff_ps_read_data(AVCodecContext *avctx, GetBitContext *gb_host, PSContext *ps
             cnt -= 2 + ps_read_extension_data(gb, ps, ps_extension_id);
         }
         if (cnt < 0) {
-            av_log(avctx, AV_LOG_ERROR, "ps extension overflow %d\n", cnt);
+            av_log(avctx, AV_LOG_ERROR, "ps extension overflow %d", cnt);
             goto err;
         }
         skip_bits(gb, cnt);

libavcodec/aacpsy.c

@@ -216,7 +216,7 @@ static const float psy_fir_coeffs[] = {
 };
 
 /**
- * Calculate the ABR attack threshold from the above LAME psymodel table.
+ * calculates the attack threshold for ABR from the above table for the LAME psy model
  */
 static float lame_calc_attack_threshold(int bitrate)
 {
@@ -400,7 +400,7 @@ static av_unused FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx,
         int stay_short = 0;
         for (i = 0; i < 8; i++) {
             for (j = 0; j < 128; j++) {
-                v = iir_filter(la[i*128+j], pch->iir_state);
+                v = iir_filter(la[(i*128+j)*ctx->avctx->channels], pch->iir_state);
                 sum += v*v;
             }
             s[i]  = sum;
@@ -776,8 +776,9 @@ static void lame_apply_block_type(AacPsyChannel *ctx, FFPsyWindowInfo *wi, int u
     ctx->next_window_seq = blocktype;
 }
 
-static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio,
-                                       const float *la, int channel, int prev_type)
+static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx,
+                                       const int16_t *audio, const int16_t *la,
+                                       int channel, int prev_type)
 {
     AacPsyContext *pctx = (AacPsyContext*) ctx->model_priv_data;
     AacPsyChannel *pch  = &pctx->ch[channel];
@@ -794,20 +795,20 @@ static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio,
         float attack_intensity[(AAC_NUM_BLOCKS_SHORT + 1) * PSY_LAME_NUM_SUBBLOCKS];
         float energy_subshort[(AAC_NUM_BLOCKS_SHORT + 1) * PSY_LAME_NUM_SUBBLOCKS];
         float energy_short[AAC_NUM_BLOCKS_SHORT + 1] = { 0 };
-        const float *firbuf = la + (AAC_BLOCK_SIZE_SHORT/4 - PSY_LAME_FIR_LEN);
+        int chans = ctx->avctx->channels;
+        const int16_t *firbuf = la + (AAC_BLOCK_SIZE_SHORT/4 - PSY_LAME_FIR_LEN) * chans;
         int j, att_sum = 0;
 
         /* LAME comment: apply high pass filter of fs/4 */
         for (i = 0; i < AAC_BLOCK_SIZE_LONG; i++) {
             float sum1, sum2;
-            sum1 = firbuf[i + (PSY_LAME_FIR_LEN - 1) / 2];
+            sum1 = firbuf[(i + ((PSY_LAME_FIR_LEN - 1) / 2)) * chans];
             sum2 = 0.0;
             for (j = 0; j < ((PSY_LAME_FIR_LEN - 1) / 2) - 1; j += 2) {
-                sum1 += psy_fir_coeffs[j] * (firbuf[i + j] + firbuf[i + PSY_LAME_FIR_LEN - j]);
-                sum2 += psy_fir_coeffs[j + 1] * (firbuf[i + j + 1] + firbuf[i + PSY_LAME_FIR_LEN - j - 1]);
+                sum1 += psy_fir_coeffs[j] * (firbuf[(i + j) * chans] + firbuf[(i + PSY_LAME_FIR_LEN - j) * chans]);
+                sum2 += psy_fir_coeffs[j + 1] * (firbuf[(i + j + 1) * chans] + firbuf[(i + PSY_LAME_FIR_LEN - j - 1) * chans]);
             }
-            /* NOTE: The LAME psymodel expects it's input in the range -32768 to 32768. Tuning this for normalized floats would be difficult. */
-            hpfsmpl[i] = (sum1 + sum2) * 32768.0f;
+            hpfsmpl[i] = sum1 + sum2;
         }
 
         /* Calculate the energies of each sub-shortblock */
@@ -822,15 +823,16 @@ static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio,
             float const *const pfe = pf + AAC_BLOCK_SIZE_LONG / (AAC_NUM_BLOCKS_SHORT * PSY_LAME_NUM_SUBBLOCKS);
             float p = 1.0f;
             for (; pf < pfe; pf++)
-                p = FFMAX(p, fabsf(*pf));
+                if (p < fabsf(*pf))
+                    p = fabsf(*pf);
             pch->prev_energy_subshort[i] = energy_subshort[i + PSY_LAME_NUM_SUBBLOCKS] = p;
             energy_short[1 + i / PSY_LAME_NUM_SUBBLOCKS] += p;
-            /* NOTE: The indexes below are [i + 3 - 2] in the LAME source.
-             *       Obviously the 3 and 2 have some significance, or this would be just [i + 1]
-             *       (which is what we use here). What the 3 stands for is ambiguous, as it is both
-             *       number of short blocks, and the number of sub-short blocks.
-             *       It seems that LAME is comparing each sub-block to sub-block + 1 in the
-             *       previous block.
+            /* FIXME: The indexes below are [i + 3 - 2] in the LAME source.
+             *          Obviously the 3 and 2 have some significance, or this would be just [i + 1]
+             *          (which is what we use here). What the 3 stands for is ambigious, as it is both
+             *          number of short blocks, and the number of sub-short blocks.
+             *          It seems that LAME is comparing each sub-block to sub-block + 1 in the
+             *          previous block.
              */
             if (p > energy_subshort[i + 1])
                 p = p / energy_subshort[i + 1];

libavcodec/aacsbr.c

@@ -1185,15 +1185,14 @@ static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
 {
     int i, n;
     const float *sbr_qmf_window = div ? sbr_qmf_window_ds : sbr_qmf_window_us;
-    const int step = 128 >> div;
     float *v;
     for (i = 0; i < 32; i++) {
-        if (*v_off < step) {
+        if (*v_off < 128 >> div) {
             int saved_samples = (1280 - 128) >> div;
             memcpy(&v0[SBR_SYNTHESIS_BUF_SIZE - saved_samples], v0, saved_samples * sizeof(float));
-            *v_off = SBR_SYNTHESIS_BUF_SIZE - saved_samples - step;
+            *v_off = SBR_SYNTHESIS_BUF_SIZE - saved_samples - (128 >> div);
         } else {
-            *v_off -= step;
+            *v_off -= 128 >> div;
         }
         v = v0 + *v_off;
         if (div) {

libavcodec/ac3_parser.c

@@ -34,18 +34,6 @@ static const uint8_t eac3_blocks[4] = {
     1, 2, 3, 6
 };
 
-/**
- * Table for center mix levels
- * reference: Section 5.4.2.4 cmixlev
- */
-static const uint8_t center_levels[4] = { 4, 5, 6, 5 };
-
-/**
- * Table for surround mix levels
- * reference: Section 5.4.2.5 surmixlev
- */
-static const uint8_t surround_levels[4] = { 4, 6, 7, 6 };
-
 
 int avpriv_ac3_parse_header(GetBitContext *gbc, AC3HeaderInfo *hdr)
 {
@@ -65,8 +53,8 @@ int avpriv_ac3_parse_header(GetBitContext *gbc, AC3HeaderInfo *hdr)
     hdr->num_blocks = 6;
 
     /* set default mix levels */
-    hdr->center_mix_level   = 5;  // -4.5dB
-    hdr->surround_mix_level = 6;  // -6.0dB
+    hdr->center_mix_level   = 1;  // -4.5dB
+    hdr->surround_mix_level = 1;  // -6.0dB
 
     if(hdr->bitstream_id <= 10) {
         /* Normal AC-3 */
@@ -88,9 +76,9 @@ int avpriv_ac3_parse_header(GetBitContext *gbc, AC3HeaderInfo *hdr)
             skip_bits(gbc, 2); // skip dsurmod
         } else {
             if((hdr->channel_mode & 1) && hdr->channel_mode != AC3_CHMODE_MONO)
-                hdr->  center_mix_level =   center_levels[get_bits(gbc, 2)];
+                hdr->center_mix_level = get_bits(gbc, 2);
             if(hdr->channel_mode & 4)
-                hdr->surround_mix_level = surround_levels[get_bits(gbc, 2)];
+                hdr->surround_mix_level = get_bits(gbc, 2);
         }
         hdr->lfe_on = get_bits1(gbc);
 

libavcodec/ac3dec.c

@@ -76,6 +76,18 @@ static const float gain_levels[9] = {
     LEVEL_MINUS_9DB
 };
 
+/**
+ * Table for center mix levels
+ * reference: Section 5.4.2.4 cmixlev
+ */
+static const uint8_t center_levels[4] = { 4, 5, 6, 5 };
+
+/**
+ * Table for surround mix levels
+ * reference: Section 5.4.2.5 surmixlev
+ */
+static const uint8_t surround_levels[4] = { 4, 6, 7, 6 };
+
 /**
  * Table for default stereo downmixing coefficients
  * reference: Section 7.8.2 Downmixing Into Two Channels
@@ -211,7 +223,7 @@ static int ac3_parse_header(AC3DecodeContext *s)
     int i;
 
     /* read the rest of the bsi. read twice for dual mono mode. */
-    i = !s->channel_mode;
+    i = !(s->channel_mode);
     do {
         skip_bits(gbc, 5); // skip dialog normalization
         if (get_bits1(gbc))
@@ -308,8 +320,8 @@ static int parse_frame_header(AC3DecodeContext *s)
 static void set_downmix_coeffs(AC3DecodeContext *s)
 {
     int i;
-    float cmix = gain_levels[s->  center_mix_level];
-    float smix = gain_levels[s->surround_mix_level];
+    float cmix = gain_levels[center_levels[s->center_mix_level]];
+    float smix = gain_levels[surround_levels[s->surround_mix_level]];
     float norm0, norm1;
 
     for (i = 0; i < s->fbw_channels; i++) {
@@ -780,7 +792,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
     }
 
     /* dynamic range */
-    i = !s->channel_mode;
+    i = !(s->channel_mode);
     do {
         if (get_bits1(gbc)) {
             s->dynamic_range[i] = ((dynamic_range_tab[get_bits(gbc, 8)] - 1.0) *
@@ -1388,8 +1400,8 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
         avctx->channels       = s->out_channels;
         avctx->channel_layout = s->channel_layout;
 
-        s->loro_center_mix_level   = gain_levels[s->  center_mix_level];
-        s->loro_surround_mix_level = gain_levels[s->surround_mix_level];
+        s->loro_center_mix_level   = gain_levels[  center_levels[s->  center_mix_level]];
+        s->loro_surround_mix_level = gain_levels[surround_levels[s->surround_mix_level]];
         s->ltrt_center_mix_level   = LEVEL_MINUS_3DB;
         s->ltrt_surround_mix_level = LEVEL_MINUS_3DB;
         /* set downmixing coefficients if needed */

libavcodec/acelp_vectors.c

@@ -238,11 +238,11 @@ void ff_set_fixed_vector(float *out, const AMRFixed *in, float scale, int size)
         float y = in->y[i] * scale;
 
         if (in->pitch_lag > 0)
-            do {
-                out[x] += y;
-                y *= in->pitch_fac;
-                x += in->pitch_lag;
-            } while (x < size && repeats);
+        do {
+            out[x] += y;
+            y *= in->pitch_fac;
+            x += in->pitch_lag;
+        } while (x < size && repeats);
     }
 }
 
@@ -254,9 +254,9 @@ void ff_clear_fixed_vector(float *out, const AMRFixed *in, int size)
         int x  = in->x[i], repeats = !((in->no_repeat_mask >> i) & 1);
 
         if (in->pitch_lag > 0)
-            do {
-                out[x] = 0.0;
-                x += in->pitch_lag;
-            } while (x < size && repeats);
+        do {
+            out[x] = 0.0;
+            x += in->pitch_lag;
+        } while (x < size && repeats);
     }
 }

libavcodec/adpcm.c

@@ -86,19 +86,14 @@ static const int swf_index_tables[4][16] = {
 typedef struct ADPCMDecodeContext {
     AVFrame frame;
     ADPCMChannelStatus status[6];
-    int vqa_version;                /**< VQA version. Used for ADPCM_IMA_WS */
 } ADPCMDecodeContext;
 
 static av_cold int adpcm_decode_init(AVCodecContext * avctx)
 {
     ADPCMDecodeContext *c = avctx->priv_data;
-    unsigned int min_channels = 1;
     unsigned int max_channels = 2;
 
     switch(avctx->codec->id) {
-    case CODEC_ID_ADPCM_EA:
-        min_channels = 2;
-        break;
     case CODEC_ID_ADPCM_EA_R1:
     case CODEC_ID_ADPCM_EA_R2:
     case CODEC_ID_ADPCM_EA_R3:
@@ -106,7 +101,7 @@ static av_cold int adpcm_decode_init(AVCodecContext * avctx)
         max_channels = 6;
         break;
     }
-    if (avctx->channels < min_channels || avctx->channels > max_channels) {
+    if (avctx->channels <= 0 || avctx->channels > max_channels) {
         av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
         return AVERROR(EINVAL);
     }
@@ -121,16 +116,12 @@ static av_cold int adpcm_decode_init(AVCodecContext * avctx)
             return -1;
         }
         break;
-    case CODEC_ID_ADPCM_IMA_APC:
-        if (avctx->extradata && avctx->extradata_size >= 8) {
+    case CODEC_ID_ADPCM_IMA_WS:
+        if (avctx->extradata && avctx->extradata_size == 2 * 4) {
             c->status[0].predictor = AV_RL32(avctx->extradata);
             c->status[1].predictor = AV_RL32(avctx->extradata + 4);
         }
         break;
-    case CODEC_ID_ADPCM_IMA_WS:
-        if (avctx->extradata && avctx->extradata_size >= 42)
-            c->vqa_version = AV_RL16(avctx->extradata);
-        break;
     default:
         break;
     }
@@ -367,7 +358,6 @@ static int get_nb_samples(AVCodecContext *avctx, const uint8_t *buf,
         break;
     /* simple 4-bit adpcm */
     case CODEC_ID_ADPCM_CT:
-    case CODEC_ID_ADPCM_IMA_APC:
     case CODEC_ID_ADPCM_IMA_EA_SEAD:
     case CODEC_ID_ADPCM_IMA_WS:
     case CODEC_ID_ADPCM_YAMAHA:
@@ -782,37 +772,13 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
             *samples++ = adpcm_ima_expand_nibble(&c->status[st], v2, 3);
         }
         break;
-    case CODEC_ID_ADPCM_IMA_APC:
+    case CODEC_ID_ADPCM_IMA_WS:
         while (src < buf + buf_size) {
             uint8_t v = *src++;
             *samples++ = adpcm_ima_expand_nibble(&c->status[0],  v >> 4  , 3);
             *samples++ = adpcm_ima_expand_nibble(&c->status[st], v & 0x0F, 3);
         }
         break;
-    case CODEC_ID_ADPCM_IMA_WS:
-        for (channel = 0; channel < avctx->channels; channel++) {
-            const uint8_t *src0;
-            int src_stride;
-            int16_t *smp = samples + channel;
-
-            if (c->vqa_version == 3) {
-                src0 = src + channel * buf_size / 2;
-                src_stride = 1;
-            } else {
-                src0 = src + channel;
-                src_stride = avctx->channels;
-            }
-            for (n = nb_samples / 2; n > 0; n--) {
-                uint8_t v = *src0;
-                src0 += src_stride;
-                *smp = adpcm_ima_expand_nibble(&c->status[channel], v >> 4  , 3);
-                smp += avctx->channels;
-                *smp = adpcm_ima_expand_nibble(&c->status[channel], v & 0x0F, 3);
-                smp += avctx->channels;
-            }
-        }
-        src = buf + buf_size;
-        break;
     case CODEC_ID_ADPCM_XA:
         while (buf_size >= 128) {
             xa_decode(samples, src, &c->status[0], &c->status[1],
@@ -1256,7 +1222,6 @@ ADPCM_DECODER(CODEC_ID_ADPCM_EA_R2, adpcm_ea_r2, "ADPCM Electronic Arts R2");
 ADPCM_DECODER(CODEC_ID_ADPCM_EA_R3, adpcm_ea_r3, "ADPCM Electronic Arts R3");
 ADPCM_DECODER(CODEC_ID_ADPCM_EA_XAS, adpcm_ea_xas, "ADPCM Electronic Arts XAS");
 ADPCM_DECODER(CODEC_ID_ADPCM_IMA_AMV, adpcm_ima_amv, "ADPCM IMA AMV");
-ADPCM_DECODER(CODEC_ID_ADPCM_IMA_APC, adpcm_ima_apc, "ADPCM IMA CRYO APC");
 ADPCM_DECODER(CODEC_ID_ADPCM_IMA_DK3, adpcm_ima_dk3, "ADPCM IMA Duck DK3");
 ADPCM_DECODER(CODEC_ID_ADPCM_IMA_DK4, adpcm_ima_dk4, "ADPCM IMA Duck DK4");
 ADPCM_DECODER(CODEC_ID_ADPCM_IMA_EA_EACS, adpcm_ima_ea_eacs, "ADPCM IMA Electronic Arts EACS");

libavcodec/adx.c

@@ -58,7 +58,7 @@ int avpriv_adx_decode_header(AVCodecContext *avctx, const uint8_t *buf,
 
     /* channels */
     avctx->channels = buf[7];
-    if (avctx->channels <= 0 || avctx->channels > 2)
+    if (avctx->channels > 2)
         return AVERROR_INVALIDDATA;
 
     /* sample rate */

libavcodec/adx_parser.c

@@ -45,31 +45,27 @@ static int adx_parse(AVCodecParserContext *s1,
     ParseContext *pc = &s->pc;
     int next = END_NOT_FOUND;
     int i;
-    uint64_t state = pc->state64;
+    uint64_t state= pc->state64;
 
-    if (!s->header_size) {
-        for (i = 0; i < buf_size; i++) {
-            state = (state << 8) | buf[i];
-            /* check for fixed fields in ADX header for possible match */
-            if ((state & 0xFFFF0000FFFFFF00) == 0x8000000003120400ULL) {
-                int channels    = state & 0xFF;
-                int header_size = ((state >> 32) & 0xFFFF) + 4;
-                if (channels > 0 && header_size >= 8) {
-                    s->header_size = header_size;
-                    s->block_size  = BLOCK_SIZE * channels;
-                    s->remaining   = i - 7 + s->header_size + s->block_size;
-                    break;
-                }
+    if(!s->header_size){
+        for(i=0; i<buf_size; i++){
+            state= (state<<8) | buf[i];
+            if((state&0xFFFF0000FFFFFF00) == 0x8000000003120400ULL && (state&0xFF) && ((state>>32)&0xFFFF)>=4){
+                s->header_size= ((state>>32)&0xFFFF) + 4;
+                s->block_size = BLOCK_SIZE * (state&0xFF);
+                s->remaining  = i - 7 + s->header_size + s->block_size;
+                break;
             }
         }
-        pc->state64 = state;
+        pc->state64= state;
     }
 
     if (s->header_size) {
-        if (!s->remaining)
+        if (!s->remaining) {
             s->remaining = s->block_size;
-        if (s->remaining <= buf_size) {
-            next = s->remaining;
+        }
+        if (s->remaining<=buf_size) {
+            next= s->remaining;
             s->remaining = 0;
         } else
             s->remaining -= buf_size;

libavcodec/adxdec.c

@@ -45,8 +45,7 @@ static av_cold int adx_decode_init(AVCodecContext *avctx)
             av_log(avctx, AV_LOG_ERROR, "error parsing ADX header\n");
             return AVERROR_INVALIDDATA;
         }
-        c->channels      = avctx->channels;
-        c->header_parsed = 1;
+        c->channels = avctx->channels;
     }
 
     avctx->sample_fmt = AV_SAMPLE_FMT_S16;
@@ -107,21 +106,21 @@ static int adx_decode_frame(AVCodecContext *avctx, void *data,
         return buf_size;
     }
 
-    if (!c->header_parsed && buf_size >= 2 && AV_RB16(buf) == 0x8000) {
+    if(AV_RB16(buf) == 0x8000){
         int header_size;
-        if ((ret = avpriv_adx_decode_header(avctx, buf, buf_size, &header_size,
+        if ((ret = avpriv_adx_decode_header(avctx, buf,
+                                            buf_size, &header_size,
                                             c->coeff)) < 0) {
             av_log(avctx, AV_LOG_ERROR, "error parsing ADX header\n");
             return AVERROR_INVALIDDATA;
         }
-        c->channels      = avctx->channels;
-        c->header_parsed = 1;
-        if (buf_size < header_size)
+        c->channels = avctx->channels;
+        if(buf_size < header_size)
             return AVERROR_INVALIDDATA;
-        buf      += header_size;
+        buf += header_size;
         buf_size -= header_size;
     }
-    if (!c->header_parsed)
+    if(c->channels <= 0)
         return AVERROR_INVALIDDATA;
 
     /* calculate number of blocks in the packet */
@@ -165,13 +164,6 @@ static int adx_decode_frame(AVCodecContext *avctx, void *data,
     return buf - avpkt->data;
 }
 
-static void adx_decode_flush(AVCodecContext *avctx)
-{
-    ADXContext *c = avctx->priv_data;
-    memset(c->prev, 0, sizeof(c->prev));
-    c->eof = 0;
-}
-
 AVCodec ff_adpcm_adx_decoder = {
     .name           = "adpcm_adx",
     .type           = AVMEDIA_TYPE_AUDIO,
@@ -179,7 +171,6 @@ AVCodec ff_adpcm_adx_decoder = {
     .priv_data_size = sizeof(ADXContext),
     .init           = adx_decode_init,
     .decode         = adx_decode_frame,
-    .flush          = adx_decode_flush,
     .capabilities   = CODEC_CAP_DR1,
     .long_name      = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
 };

libavcodec/adxenc.c

@@ -19,9 +19,9 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
+#include "libavutil/intreadwrite.h"
 #include "avcodec.h"
 #include "adx.h"
-#include "bytestream.h"
 #include "put_bits.h"
 
 /**
@@ -33,135 +33,167 @@
  * adx2wav & wav2adx http://www.geocities.co.jp/Playtown/2004/
  */
 
-static void adx_encode(ADXContext *c, uint8_t *adx, const int16_t *wav,
-                       ADXChannelState *prev, int channels)
+/* 18 bytes <-> 32 samples */
+
+static void adx_encode(ADXContext *c, unsigned char *adx, const short *wav,
+                       ADXChannelState *prev)
 {
     PutBitContext pb;
     int scale;
-    int i, j;
-    int s0, s1, s2, d;
-    int max = 0;
-    int min = 0;
-    int data[BLOCK_SAMPLES];
+    int i;
+    int s0,s1,s2,d;
+    int max=0;
+    int min=0;
+    int data[32];
 
     s1 = prev->s1;
     s2 = prev->s2;
-    for (i = 0, j = 0; j < 32; i += channels, j++) {
+    for(i=0;i<32;i++) {
         s0 = wav[i];
         d = ((s0 << COEFF_BITS) - c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS;
-        data[j] = d;
-        if (max < d)
-            max = d;
-        if (min > d)
-            min = d;
+        data[i]=d;
+        if (max<d) max=d;
+        if (min>d) min=d;
         s2 = s1;
         s1 = s0;
     }
     prev->s1 = s1;
     prev->s2 = s2;
 
-    if (max == 0 && min == 0) {
-        memset(adx, 0, BLOCK_SIZE);
+    /* -8..+7 */
+
+    if (max==0 && min==0) {
+        memset(adx,0,18);
         return;
     }
 
-    if (max / 7 > -min / 8)
-        scale = max / 7;
-    else
-        scale = -min / 8;
+    if (max/7>-min/8) scale = max/7;
+    else scale = -min/8;
 
-    if (scale == 0)
-        scale = 1;
+    if (scale==0) scale=1;
 
     AV_WB16(adx, scale);
 
     init_put_bits(&pb, adx + 2, 16);
-    for (i = 0; i < BLOCK_SAMPLES; i++)
-        put_sbits(&pb, 4, av_clip(data[i] / scale, -8, 7));
+    for (i = 0; i < 32; i++)
+        put_sbits(&pb, 4, av_clip(data[i]/scale, -8, 7));
     flush_put_bits(&pb);
 }
 
-#define HEADER_SIZE 36
-
-static int adx_encode_header(AVCodecContext *avctx, uint8_t *buf, int bufsize)
+static int adx_encode_header(AVCodecContext *avctx,unsigned char *buf,size_t bufsize)
 {
+#if 0
+    struct {
+        uint32_t offset; /* 0x80000000 + sample start - 4 */
+        unsigned char unknown1[3]; /* 03 12 04 */
+        unsigned char channel; /* 1 or 2 */
+        uint32_t freq;
+        uint32_t size;
+        uint32_t unknown2; /* 01 f4 03 00 */
+        uint32_t unknown3; /* 00 00 00 00 */
+        uint32_t unknown4; /* 00 00 00 00 */
+
+    /* if loop
+        unknown3 00 15 00 01
+        unknown4 00 00 00 01
+        long loop_start_sample;
+        long loop_start_byte;
+        long loop_end_sample;
+        long loop_end_byte;
+        long
+    */
+    } adxhdr; /* big endian */
+    /* offset-6 "(c)CRI" */
+#endif
     ADXContext *c = avctx->priv_data;
 
-    if (bufsize < HEADER_SIZE)
-        return AVERROR(EINVAL);
-
-    bytestream_put_be16(&buf, 0x8000);              /* header signature */
-    bytestream_put_be16(&buf, HEADER_SIZE - 4);     /* copyright offset */
-    bytestream_put_byte(&buf, 3);                   /* encoding */
-    bytestream_put_byte(&buf, BLOCK_SIZE);          /* block size */
-    bytestream_put_byte(&buf, 4);                   /* sample size */
-    bytestream_put_byte(&buf, avctx->channels);     /* channels */
-    bytestream_put_be32(&buf, avctx->sample_rate);  /* sample rate */
-    bytestream_put_be32(&buf, 0);                   /* total sample count */
-    bytestream_put_be16(&buf, c->cutoff);           /* cutoff frequency */
-    bytestream_put_byte(&buf, 3);                   /* version */
-    bytestream_put_byte(&buf, 0);                   /* flags */
-    bytestream_put_be32(&buf, 0);                   /* unknown */
-    bytestream_put_be32(&buf, 0);                   /* loop enabled */
-    bytestream_put_be16(&buf, 0);                   /* padding */
-    bytestream_put_buffer(&buf, "(c)CRI", 6);       /* copyright signature */
-
-    return HEADER_SIZE;
+    AV_WB32(buf+0x00,0x80000000|0x20);
+    AV_WB32(buf+0x04,0x03120400|avctx->channels);
+    AV_WB32(buf+0x08,avctx->sample_rate);
+    AV_WB32(buf+0x0c,0); /* FIXME: set after */
+    AV_WB16(buf + 0x10, c->cutoff);
+    AV_WB32(buf + 0x12, 0x03000000);
+    AV_WB32(buf + 0x16, 0x00000000);
+    AV_WB32(buf + 0x1a, 0x00000000);
+    memcpy (buf + 0x1e, "(c)CRI", 6);
+    return 0x20+4;
 }
 
 static av_cold int adx_encode_init(AVCodecContext *avctx)
 {
     ADXContext *c = avctx->priv_data;
 
-    if (avctx->channels > 2) {
-        av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
-        return AVERROR(EINVAL);
-    }
-    avctx->frame_size = BLOCK_SAMPLES;
+    if (avctx->channels > 2)
+        return -1; /* only stereo or mono =) */
+    avctx->frame_size = 32;
 
-    avctx->coded_frame = avcodec_alloc_frame();
+    avctx->coded_frame= avcodec_alloc_frame();
+    avctx->coded_frame->key_frame= 1;
+
+//    avctx->bit_rate = avctx->sample_rate*avctx->channels*18*8/32;
 
     /* the cutoff can be adjusted, but this seems to work pretty well */
     c->cutoff = 500;
     ff_adx_calculate_coeffs(c->cutoff, avctx->sample_rate, COEFF_BITS, c->coeff);
 
+    av_log(avctx, AV_LOG_DEBUG, "adx encode init\n");
+
     return 0;
 }
 
 static av_cold int adx_encode_close(AVCodecContext *avctx)
 {
     av_freep(&avctx->coded_frame);
+
     return 0;
 }
 
-static int adx_encode_frame(AVCodecContext *avctx, uint8_t *frame,
-                            int buf_size, void *data)
+static int adx_encode_frame(AVCodecContext *avctx,
+                uint8_t *frame, int buf_size, void *data)
 {
-    ADXContext *c          = avctx->priv_data;
-    const int16_t *samples = data;
-    uint8_t *dst           = frame;
-    int ch;
+    ADXContext *c = avctx->priv_data;
+    const short *samples = data;
+    unsigned char *dst = frame;
+    int rest = avctx->frame_size;
 
+/*
+    input data size =
+    ffmpeg.c: do_audio_out()
+    frame_bytes = enc->frame_size * 2 * enc->channels;
+*/
+
+//    printf("sz=%d ",buf_size); fflush(stdout);
     if (!c->header_parsed) {
-        int hdrsize;
-        if ((hdrsize = adx_encode_header(avctx, dst, buf_size)) < 0) {
-            av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
-            return AVERROR(EINVAL);
-        }
-        dst      += hdrsize;
-        buf_size -= hdrsize;
+        int hdrsize = adx_encode_header(avctx,dst,buf_size);
+        dst+=hdrsize;
         c->header_parsed = 1;
     }
-    if (buf_size < BLOCK_SIZE * avctx->channels) {
-        av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
-        return AVERROR(EINVAL);
-    }
 
-    for (ch = 0; ch < avctx->channels; ch++) {
-        adx_encode(c, dst, samples + ch, &c->prev[ch], avctx->channels);
-        dst += BLOCK_SIZE;
+    if (avctx->channels==1) {
+        while(rest>=32) {
+            adx_encode(c, dst, samples, c->prev);
+            dst+=18;
+            samples+=32;
+            rest-=32;
+        }
+    } else {
+        while(rest>=32*2) {
+            short tmpbuf[32*2];
+            int i;
+
+            for(i=0;i<32;i++) {
+                tmpbuf[i] = samples[i*2];
+                tmpbuf[i+32] = samples[i*2+1];
+            }
+
+            adx_encode(c, dst,      tmpbuf,      c->prev);
+            adx_encode(c, dst + 18, tmpbuf + 32, c->prev + 1);
+            dst+=18*2;
+            samples+=32*2;
+            rest-=32*2;
+        }
     }
-    return dst - frame;
+    return dst-frame;
 }
 
 AVCodec ff_adpcm_adx_encoder = {
@@ -172,7 +204,6 @@ AVCodec ff_adpcm_adx_encoder = {
     .init           = adx_encode_init,
     .encode         = adx_encode_frame,
     .close          = adx_encode_close,
-    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
-                                                      AV_SAMPLE_FMT_NONE },
-    .long_name      = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
+    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
+    .long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
 };

libavcodec/alac.c

@@ -25,23 +25,27 @@
  * @author 2005 David Hammerton
  * @see http://crazney.net/programs/itunes/alac.html
  *
- * Note: This decoder expects a 36-byte QuickTime atom to be
+ * Note: This decoder expects a 36- (0x24-)byte QuickTime atom to be
  * passed through the extradata[_size] fields. This atom is tacked onto
  * the end of an 'alac' stsd atom and has the following format:
+ *  bytes 0-3   atom size (0x24), big-endian
+ *  bytes 4-7   atom type ('alac', not the 'alac' tag from start of stsd)
+ *  bytes 8-35  data bytes needed by decoder
  *
- * 32bit  atom size
- * 32bit  tag                  ("alac")
- * 32bit  tag version          (0)
- * 32bit  samples per frame    (used when not set explicitly in the frames)
- *  8bit  compatible version   (0)
+ * Extradata:
+ * 32bit  size
+ * 32bit  tag (=alac)
+ * 32bit  zero?
+ * 32bit  max sample per frame
+ *  8bit  ?? (zero?)
  *  8bit  sample size
- *  8bit  history mult         (40)
- *  8bit  initial history      (14)
- *  8bit  kmodifier            (10)
- *  8bit  channels
- * 16bit  maxRun               (255)
- * 32bit  max coded frame size (0 means unknown)
- * 32bit  average bitrate      (0 means unknown)
+ *  8bit  history mult
+ *  8bit  initial history
+ *  8bit  kmodifier
+ *  8bit  channels?
+ * 16bit  ??
+ * 32bit  max coded frame size
+ * 32bit  bitrate?
  * 32bit  samplerate
  */
 
@@ -351,17 +355,6 @@ static void interleave_stereo_24(int32_t *buffer[MAX_CHANNELS],
     }
 }
 
-static void interleave_stereo_32(int32_t *buffer[MAX_CHANNELS],
-                                 int32_t *buffer_out, int numsamples)
-{
-    int i;
-
-    for (i = 0; i < numsamples; i++) {
-        *buffer_out++ = buffer[0][i];
-        *buffer_out++ = buffer[1][i];
-    }
-}
-
 static int alac_decode_frame(AVCodecContext *avctx, void *data,
                              int *got_frame_ptr, AVPacket *avpkt)
 {
@@ -471,29 +464,24 @@ static int alac_decode_frame(AVCodecContext *avctx, void *data,
             if(ret<0)
                 return ret;
 
-            /* adaptive FIR filter */
-            if (prediction_type[ch] == 15) {
-                /* Prediction type 15 runs the adaptive FIR twice.
-                 * The first pass uses the special-case coef_num = 31, while
-                 * the second pass uses the coefs from the bitstream.
-                 *
-                 * However, this prediction type is not currently used by the
-                 * reference encoder.
-                 */
+            if (prediction_type[ch] == 0) {
+                /* adaptive fir */
                 predictor_decompress_fir_adapt(alac->predicterror_buffer[ch],
-                                               alac->predicterror_buffer[ch],
-                                               outputsamples, readsamplesize,
-                                               NULL, 31, 0);
-            } else if (prediction_type[ch] > 0) {
-                av_log(avctx, AV_LOG_WARNING, "unknown prediction type: %i\n",
-                       prediction_type[ch]);
+                                               alac->outputsamples_buffer[ch],
+                                               outputsamples,
+                                               readsamplesize,
+                                               predictor_coef_table[ch],
+                                               predictor_coef_num[ch],
+                                               prediction_quantitization[ch]);
+            } else {
+                av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[ch]);
+                /* I think the only other prediction type (or perhaps this is
+                 * just a boolean?) runs adaptive fir twice.. like:
+                 * predictor_decompress_fir_adapt(predictor_error, tempout, ...)
+                 * predictor_decompress_fir_adapt(predictor_error, outputsamples ...)
+                 * little strange..
+                 */
             }
-            predictor_decompress_fir_adapt(alac->predicterror_buffer[ch],
-                                           alac->outputsamples_buffer[ch],
-                                           outputsamples, readsamplesize,
-                                           predictor_coef_table[ch],
-                                           predictor_coef_num[ch],
-                                           prediction_quantitization[ch]);
         }
     } else {
         /* not compressed, easy case */
@@ -544,16 +532,6 @@ static int alac_decode_frame(AVCodecContext *avctx, void *data,
                 outbuffer[i] = alac->outputsamples_buffer[0][i] << 8;
         }
         break;
-    case 32:
-        if (channels == 2) {
-            interleave_stereo_32(alac->outputsamples_buffer,
-                                 (int32_t *)alac->frame.data[0], outputsamples);
-        } else {
-            int32_t *outbuffer = (int32_t *)alac->frame.data[0];
-            for (i = 0; i < outputsamples; i++)
-                outbuffer[i] = alac->outputsamples_buffer[0][i];
-        }
-        break;
     }
 
     if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8)
@@ -606,7 +584,7 @@ static int alac_set_info(ALACContext *alac)
 
     ptr += 4; /* size */
     ptr += 4; /* alac */
-    ptr += 4; /* version */
+    ptr += 4; /* 0 ? */
 
     if(AV_RB32(ptr) >= UINT_MAX/4){
         av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n");
@@ -615,15 +593,15 @@ static int alac_set_info(ALACContext *alac)
 
     /* buffer size / 2 ? */
     alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr);
-    ptr++;                          /* compatible version */
+    ptr++;                          /* ??? */
     alac->setinfo_sample_size           = *ptr++;
     alac->setinfo_rice_historymult      = *ptr++;
     alac->setinfo_rice_initialhistory   = *ptr++;
     alac->setinfo_rice_kmodifier        = *ptr++;
     alac->numchannels                   = *ptr++;
-    bytestream_get_be16(&ptr);      /* maxRun */
+    bytestream_get_be16(&ptr);      /* ??? */
     bytestream_get_be32(&ptr);      /* max coded frame size */
-    bytestream_get_be32(&ptr);      /* average bitrate */
+    bytestream_get_be32(&ptr);      /* bitrate ? */
     bytestream_get_be32(&ptr);      /* samplerate */
 
     return 0;
@@ -649,7 +627,6 @@ static av_cold int alac_decode_init(AVCodecContext * avctx)
     switch (alac->setinfo_sample_size) {
     case 16: avctx->sample_fmt    = AV_SAMPLE_FMT_S16;
              break;
-    case 32:
     case 24: avctx->sample_fmt    = AV_SAMPLE_FMT_S32;
              break;
     default: av_log_ask_for_sample(avctx, "Sample depth %d is not supported.\n",

libavcodec/alacenc.c

@@ -259,7 +259,7 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch)
         // generate warm-up samples
         residual[0] = samples[0];
         for (i = 1; i <= lpc.lpc_order; i++)
-            residual[i] = samples[i] - samples[i-1];
+            residual[i] = sign_extend(samples[i] - samples[i-1], s->write_sample_size);
 
         // perform lpc on remaining samples
         for (i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) {
@@ -348,7 +348,6 @@ static void alac_entropy_coder(AlacEncodeContext *s)
 static void write_compressed_frame(AlacEncodeContext *s)
 {
     int i, j;
-    int prediction_type = 0;
 
     if (s->avctx->channels == 2)
         alac_stereo_decorrelation(s);
@@ -359,7 +358,7 @@ static void write_compressed_frame(AlacEncodeContext *s)
 
         calc_predictor_params(s, i);
 
-        put_bits(&s->pbctx, 4, prediction_type);
+        put_bits(&s->pbctx, 4, 0);  // prediction type : currently only type 0 has been RE'd
         put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant);
 
         put_bits(&s->pbctx, 3, s->rc.rice_modifier);
@@ -374,14 +373,6 @@ static void write_compressed_frame(AlacEncodeContext *s)
 
     for (i = 0; i < s->avctx->channels; i++) {
         alac_linear_predictor(s, i);
-
-        // TODO: determine when this will actually help. for now it's not used.
-        if (prediction_type == 15) {
-            // 2nd pass 1st order filter
-            for (j = s->avctx->frame_size - 1; j > 0; j--)
-                s->predictor_buf[j] -= s->predictor_buf[j - 1];
-        }
-
         alac_entropy_coder(s);
     }
 }
@@ -400,11 +391,8 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
         return -1;
     }
 
-    /* TODO: Correctly implement multi-channel ALAC.
-             It is similar to multi-channel AAC, in that it has a series of
-             single-channel (SCE), channel-pair (CPE), and LFE elements. */
-    if (avctx->channels > 2) {
-        av_log(avctx, AV_LOG_ERROR, "only mono or stereo input is currently supported\n");
+    if(avctx->channels > 2) {
+        av_log(avctx, AV_LOG_ERROR, "channels > 2 not supported\n");
         return AVERROR_PATCHWELCOME;
     }
 

libavcodec/allcodecs.c

@@ -79,7 +79,6 @@ void avcodec_register_all(void)
     REGISTER_ENCDEC  (ASV2, asv2);
     REGISTER_DECODER (AURA, aura);
     REGISTER_DECODER (AURA2, aura2);
-    REGISTER_ENCDEC  (AVRP, avrp);
     REGISTER_DECODER (AVS, avs);
     REGISTER_DECODER (BETHSOFTVID, bethsoftvid);
     REGISTER_DECODER (BFI, bfi);
@@ -184,8 +183,8 @@ void avcodec_register_all(void)
     REGISTER_DECODER (QDRAW, qdraw);
     REGISTER_DECODER (QPEG, qpeg);
     REGISTER_ENCDEC  (QTRLE, qtrle);
-    REGISTER_ENCDEC  (R10K,  r10k);
-    REGISTER_ENCDEC  (R210,  r210);
+    REGISTER_DECODER (R10K,  r10k);
+    REGISTER_DECODER (R210,  r210);
     REGISTER_ENCDEC  (RAWVIDEO, rawvideo);
     REGISTER_DECODER (RL2, rl2);
     REGISTER_ENCDEC  (ROQ, roq);
@@ -217,7 +216,6 @@ void avcodec_register_all(void)
     REGISTER_DECODER (UTVIDEO, utvideo);
     REGISTER_ENCDEC  (V210,  v210);
     REGISTER_DECODER (V210X, v210x);
-    REGISTER_ENCDEC  (V308, v308);
     REGISTER_ENCDEC  (V410, v410);
     REGISTER_DECODER (VB, vb);
     REGISTER_DECODER (VBLE, vble);
@@ -245,10 +243,8 @@ void avcodec_register_all(void)
     REGISTER_DECODER (XAN_WC3, xan_wc3);
     REGISTER_DECODER (XAN_WC4, xan_wc4);
     REGISTER_DECODER (XL, xl);
-    REGISTER_ENCDEC  (XWD, xwd);
     REGISTER_ENCDEC  (Y41P, y41p);
     REGISTER_DECODER (YOP, yop);
-    REGISTER_ENCDEC  (YUV4, yuv4);
     REGISTER_ENCDEC  (ZLIB, zlib);
     REGISTER_ENCDEC  (ZMBV, zmbv);
 
@@ -271,7 +267,6 @@ void avcodec_register_all(void)
     REGISTER_ENCDEC  (DCA, dca);
     REGISTER_DECODER (DSICINAUDIO, dsicinaudio);
     REGISTER_ENCDEC  (EAC3, eac3);
-    REGISTER_DECODER (FFWAVESYNTH, ffwavesynth);
     REGISTER_ENCDEC  (FLAC, flac);
     REGISTER_ENCDEC  (G723_1, g723_1);
     REGISTER_DECODER (G729, g729);
@@ -365,7 +360,6 @@ void avcodec_register_all(void)
     REGISTER_ENCDEC  (ADPCM_G722, adpcm_g722);
     REGISTER_ENCDEC  (ADPCM_G726, adpcm_g726);
     REGISTER_DECODER (ADPCM_IMA_AMV, adpcm_ima_amv);
-    REGISTER_DECODER (ADPCM_IMA_APC, adpcm_ima_apc);
     REGISTER_DECODER (ADPCM_IMA_DK3, adpcm_ima_dk3);
     REGISTER_DECODER (ADPCM_IMA_DK4, adpcm_ima_dk4);
     REGISTER_DECODER (ADPCM_IMA_EA_EACS, adpcm_ima_ea_eacs);
@@ -434,7 +428,6 @@ void avcodec_register_all(void)
     REGISTER_PARSER  (DVBSUB, dvbsub);
     REGISTER_PARSER  (DVDSUB, dvdsub);
     REGISTER_PARSER  (FLAC, flac);
-    REGISTER_PARSER  (GSM, gsm);
     REGISTER_PARSER  (H261, h261);
     REGISTER_PARSER  (H263, h263);
     REGISTER_PARSER  (H264, h264);

libavcodec/amrnbdec.c

@@ -978,10 +978,6 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
 
         pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse);
 
-        if (fixed_sparse.pitch_lag == 0) {
-            av_log(avctx, AV_LOG_ERROR, "The file is corrupted, pitch_lag = 0 is not allowed\n");
-            return AVERROR_INVALIDDATA;
-        }
         ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0,
                             AMR_SUBFRAME_SIZE);
 

libavcodec/amrwbdec.c

@@ -111,7 +111,7 @@ static av_cold int amrwb_decode_init(AVCodecContext *avctx)
 
 /**
  * Decode the frame header in the "MIME/storage" format. This format
- * is simpler and does not carry the auxiliary frame information.
+ * is simpler and does not carry the auxiliary information of the frame
  *
  * @param[in] ctx                  The Context
  * @param[in] buf                  Pointer to the input buffer
@@ -133,7 +133,7 @@ static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf)
 }
 
 /**
- * Decode quantized ISF vectors using 36-bit indexes (6K60 mode only).
+ * Decodes quantized ISF vectors using 36-bit indexes (6K60 mode only)
  *
  * @param[in]  ind                 Array of 5 indexes
  * @param[out] isf_q               Buffer for isf_q[LP_ORDER]
@@ -160,7 +160,7 @@ static void decode_isf_indices_36b(uint16_t *ind, float *isf_q)
 }
 
 /**
- * Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode).
+ * Decodes quantized ISF vectors using 46-bit indexes (except 6K60 mode)
  *
  * @param[in]  ind                 Array of 7 indexes
  * @param[out] isf_q               Buffer for isf_q[LP_ORDER]
@@ -193,8 +193,8 @@ static void decode_isf_indices_46b(uint16_t *ind, float *isf_q)
 }
 
 /**
- * Apply mean and past ISF values using the prediction factor.
- * Updates past ISF vector.
+ * Apply mean and past ISF values using the prediction factor
+ * Updates past ISF vector
  *
  * @param[in,out] isf_q            Current quantized ISF
  * @param[in,out] isf_past         Past quantized ISF
@@ -215,7 +215,7 @@ static void isf_add_mean_and_past(float *isf_q, float *isf_past)
 
 /**
  * Interpolate the fourth ISP vector from current and past frames
- * to obtain an ISP vector for each subframe.
+ * to obtain a ISP vector for each subframe
  *
  * @param[in,out] isp_q            ISPs for each subframe
  * @param[in]     isp4_past        Past ISP for subframe 4
@@ -232,9 +232,9 @@ static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past)
 }
 
 /**
- * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes).
- * Calculate integer lag and fractional lag always using 1/4 resolution.
- * In 1st and 3rd subframes the index is relative to last subframe integer lag.
+ * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes)
+ * Calculate integer lag and fractional lag always using 1/4 resolution
+ * In 1st and 3rd subframes the index is relative to last subframe integer lag
  *
  * @param[out]    lag_int          Decoded integer pitch lag
  * @param[out]    lag_frac         Decoded fractional pitch lag
@@ -271,9 +271,9 @@ static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index,
 }
 
 /**
- * Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes.
- * The description is analogous to decode_pitch_lag_high, but in 6k60 the
- * relative index is used for all subframes except the first.
+ * Decode a adaptive codebook index into pitch lag for 8k85 and 6k60 modes
+ * Description is analogous to decode_pitch_lag_high, but in 6k60 relative
+ * index is used for all subframes except the first
  */
 static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,
                                  uint8_t *base_lag_int, int subframe, enum Mode mode)
@@ -298,7 +298,7 @@ static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,
 
 /**
  * Find the pitch vector by interpolating the past excitation at the
- * pitch delay, which is obtained in this function.
+ * pitch delay, which is obtained in this function
  *
  * @param[in,out] ctx              The context
  * @param[in]     amr_subframe     Current subframe data
@@ -351,10 +351,10 @@ static void decode_pitch_vector(AMRWBContext *ctx,
 /**
  * The next six functions decode_[i]p_track decode exactly i pulses
  * positions and amplitudes (-1 or 1) in a subframe track using
- * an encoded pulse indexing (TS 26.190 section 5.8.2).
+ * an encoded pulse indexing (TS 26.190 section 5.8.2)
  *
  * The results are given in out[], in which a negative number means
- * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ).
+ * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) )
  *
  * @param[out] out                 Output buffer (writes i elements)
  * @param[in]  code                Pulse index (no. of bits varies, see below)
@@ -470,7 +470,7 @@ static void decode_6p_track(int *out, int code, int m, int off) ///code: 6m-2 bi
 
 /**
  * Decode the algebraic codebook index to pulse positions and signs,
- * then construct the algebraic codebook vector.
+ * then construct the algebraic codebook vector
  *
  * @param[out] fixed_vector        Buffer for the fixed codebook excitation
  * @param[in]  pulse_hi            MSBs part of the pulse index array (higher modes only)
@@ -541,7 +541,7 @@ static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi,
 }
 
 /**
- * Decode pitch gain and fixed gain correction factor.
+ * Decode pitch gain and fixed gain correction factor
  *
  * @param[in]  vq_gain             Vector-quantized index for gains
  * @param[in]  mode                Mode of the current frame
@@ -559,7 +559,7 @@ static void decode_gains(const uint8_t vq_gain, const enum Mode mode,
 }
 
 /**
- * Apply pitch sharpening filters to the fixed codebook vector.
+ * Apply pitch sharpening filters to the fixed codebook vector
  *
  * @param[in]     ctx              The context
  * @param[in,out] fixed_vector     Fixed codebook excitation
@@ -580,7 +580,7 @@ static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
 }
 
 /**
- * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced).
+ * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced)
  *
  * @param[in] p_vector, f_vector   Pitch and fixed excitation vectors
  * @param[in] p_gain, f_gain       Pitch and fixed gains
@@ -599,8 +599,8 @@ static float voice_factor(float *p_vector, float p_gain,
 }
 
 /**
- * Reduce fixed vector sparseness by smoothing with one of three IR filters,
- * also known as "adaptive phase dispersion".
+ * Reduce fixed vector sparseness by smoothing with one of three IR filters
+ * Also known as "adaptive phase dispersion"
  *
  * @param[in]     ctx              The context
  * @param[in,out] fixed_vector     Unfiltered fixed vector
@@ -670,7 +670,7 @@ static float *anti_sparseness(AMRWBContext *ctx,
 
 /**
  * Calculate a stability factor {teta} based on distance between
- * current and past isf. A value of 1 shows maximum signal stability.
+ * current and past isf. A value of 1 shows maximum signal stability
  */
 static float stability_factor(const float *isf, const float *isf_past)
 {
@@ -687,7 +687,7 @@ static float stability_factor(const float *isf, const float *isf_past)
 
 /**
  * Apply a non-linear fixed gain smoothing in order to reduce
- * fluctuation in the energy of excitation.
+ * fluctuation in the energy of excitation
  *
  * @param[in]     fixed_gain       Unsmoothed fixed gain
  * @param[in,out] prev_tr_gain     Previous threshold gain (updated)
@@ -718,7 +718,7 @@ static float noise_enhancer(float fixed_gain, float *prev_tr_gain,
 }
 
 /**
- * Filter the fixed_vector to emphasize the higher frequencies.
+ * Filter the fixed_vector to emphasize the higher frequencies
  *
  * @param[in,out] fixed_vector     Fixed codebook vector
  * @param[in]     voice_fac        Frame voicing factor
@@ -742,7 +742,7 @@ static void pitch_enhancer(float *fixed_vector, float voice_fac)
 }
 
 /**
- * Conduct 16th order linear predictive coding synthesis from excitation.
+ * Conduct 16th order linear predictive coding synthesis from excitation
  *
  * @param[in]     ctx              Pointer to the AMRWBContext
  * @param[in]     lpc              Pointer to the LPC coefficients
@@ -802,7 +802,7 @@ static void de_emphasis(float *out, float *in, float m, float mem[1])
 
 /**
  * Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using
- * a FIR interpolation filter. Uses past data from before *in address.
+ * a FIR interpolation filter. Uses past data from before *in address
  *
  * @param[out] out                 Buffer for interpolated signal
  * @param[in]  in                  Current signal data (length 0.8*o_size)
@@ -832,7 +832,7 @@ static void upsample_5_4(float *out, const float *in, int o_size)
 
 /**
  * Calculate the high-band gain based on encoded index (23k85 mode) or
- * on the low-band speech signal and the Voice Activity Detection flag.
+ * on the low-band speech signal and the Voice Activity Detection flag
  *
  * @param[in] ctx                  The context
  * @param[in] synth                LB speech synthesis at 12.8k
@@ -857,7 +857,7 @@ static float find_hb_gain(AMRWBContext *ctx, const float *synth,
 
 /**
  * Generate the high-band excitation with the same energy from the lower
- * one and scaled by the given gain.
+ * one and scaled by the given gain
  *
  * @param[in]  ctx                 The context
  * @param[out] hb_exc              Buffer for the excitation
@@ -880,7 +880,7 @@ static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
 }
 
 /**
- * Calculate the auto-correlation for the ISF difference vector.
+ * Calculate the auto-correlation for the ISF difference vector
  */
 static float auto_correlation(float *diff_isf, float mean, int lag)
 {
@@ -896,7 +896,7 @@ static float auto_correlation(float *diff_isf, float mean, int lag)
 
 /**
  * Extrapolate a ISF vector to the 16kHz range (20th order LP)
- * used at mode 6k60 LP filter for the high frequency band.
+ * used at mode 6k60 LP filter for the high frequency band
  *
  * @param[out] out                 Buffer for extrapolated isf
  * @param[in]  isf                 Input isf vector
@@ -981,7 +981,7 @@ static void lpc_weighting(float *out, const float *lpc, float gamma, int size)
 
 /**
  * Conduct 20th order linear predictive coding synthesis for the high
- * frequency band excitation at 16kHz.
+ * frequency band excitation at 16kHz
  *
  * @param[in]     ctx              The context
  * @param[in]     subframe         Current subframe index (0 to 3)
@@ -1019,8 +1019,8 @@ static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples,
 }
 
 /**
- * Apply a 15th order filter to high-band samples.
- * The filter characteristic depends on the given coefficients.
+ * Apply to high-band samples a 15th order filter
+ * The filter characteristic depends on the given coefficients
  *
  * @param[out]    out              Buffer for filtered output
  * @param[in]     fir_coef         Filter coefficients
@@ -1048,7 +1048,7 @@ static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1],
 }
 
 /**
- * Update context state before the next subframe.
+ * Update context state before the next subframe
  */
 static void update_sub_state(AMRWBContext *ctx)
 {

libavcodec/apedec.c

@@ -20,7 +20,7 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#define BITSTREAM_READER_LE
+#define ALT_BITSTREAM_READER_LE
 #include "avcodec.h"
 #include "dsputil.h"
 #include "get_bits.h"

libavcodec/arm/asm.S

@@ -68,7 +68,6 @@ ELF     .size   \name, . - \name
         .purgem endfunc
     .endm
         .text
-        .align          2
     .if \export
         .global EXTERN_ASM\name
 EXTERN_ASM\name:
@@ -114,12 +113,6 @@ T       add             \rn, \rn, \rm
 T       ldr             \rt, [\rn]
 .endm
 
-.macro  ldr_dpre        rt,  rn,  rm:vararg
-A       ldr             \rt, [\rn, -\rm]!
-T       sub             \rn, \rn, \rm
-T       ldr             \rt, [\rn]
-.endm
-
 .macro  ldr_dpren       rt,  rn,  rm:vararg
 A       ldr             \rt, [\rn, -\rm]
 T       sub             \rt, \rn, \rm

libavcodec/arm/dca.h

@@ -25,7 +25,7 @@
 #include "config.h"
 #include "libavutil/intmath.h"
 
-#if HAVE_ARMV6 && HAVE_INLINE_ASM && AV_GCC_VERSION_AT_LEAST(4,4)
+#if HAVE_ARMV6 && HAVE_INLINE_ASM
 
 #define decode_blockcodes decode_blockcodes
 static inline int decode_blockcodes(int code1, int code2, int levels,

libavcodec/arm/fft_init_arm.c

@@ -42,12 +42,10 @@ av_cold void ff_fft_init_arm(FFTContext *s)
     if (HAVE_NEON) {
         s->fft_permute  = ff_fft_permute_neon;
         s->fft_calc     = ff_fft_calc_neon;
-#if CONFIG_MDCT
         s->imdct_calc   = ff_imdct_calc_neon;
         s->imdct_half   = ff_imdct_half_neon;
         s->mdct_calc    = ff_mdct_calc_neon;
         s->mdct_permutation = FF_MDCT_PERM_INTERLEAVE;
-#endif
     }
 }
 

libavcodec/arm/int_neon.S

@@ -66,10 +66,10 @@ function ff_scalarproduct_int16_neon, export=1
 
 3:      vpadd.s32       d16, d0,   d1
         vpadd.s32       d17, d2,   d3
-        vpadd.s32       d10, d4,   d5
-        vpadd.s32       d11, d6,   d7
+        vpadd.s32       d18, d4,   d5
+        vpadd.s32       d19, d6,   d7
         vpadd.s32       d0,  d16,  d17
-        vpadd.s32       d1,  d10,  d11
+        vpadd.s32       d1,  d18,  d19
         vpadd.s32       d2,  d0,   d1
         vpaddl.s32      d3,  d2
         vmov.32         r0,  d3[0]
@@ -106,10 +106,10 @@ function ff_scalarproduct_and_madd_int16_neon, export=1
 
         vpadd.s32       d16, d0,   d1
         vpadd.s32       d17, d2,   d3
-        vpadd.s32       d10, d4,   d5
-        vpadd.s32       d11, d6,   d7
+        vpadd.s32       d18, d4,   d5
+        vpadd.s32       d19, d6,   d7
         vpadd.s32       d0,  d16,  d17
-        vpadd.s32       d1,  d10,  d11
+        vpadd.s32       d1,  d18,  d19
         vpadd.s32       d2,  d0,   d1
         vpaddl.s32      d3,  d2
         vmov.32         r0,  d3[0]

libavcodec/arm/rv34dsp_init_neon.c

@@ -23,18 +23,11 @@
 #include "libavcodec/avcodec.h"
 #include "libavcodec/rv34dsp.h"
 
+void ff_rv34_inv_transform_neon(DCTELEM *block);
 void ff_rv34_inv_transform_noround_neon(DCTELEM *block);
 
-void ff_rv34_inv_transform_noround_dc_neon(DCTELEM *block);
-
-void ff_rv34_idct_add_neon(uint8_t *dst, int stride, DCTELEM *block);
-void ff_rv34_idct_dc_add_neon(uint8_t *dst, int stride, int dc);
-
 void ff_rv34dsp_init_neon(RV34DSPContext *c, DSPContext* dsp)
 {
-    c->rv34_inv_transform    = ff_rv34_inv_transform_noround_neon;
-    c->rv34_inv_transform_dc = ff_rv34_inv_transform_noround_dc_neon;
-
-    c->rv34_idct_add    = ff_rv34_idct_add_neon;
-    c->rv34_idct_dc_add = ff_rv34_idct_dc_add_neon;
+    c->rv34_inv_transform_tab[0] = ff_rv34_inv_transform_neon;
+    c->rv34_inv_transform_tab[1] = ff_rv34_inv_transform_noround_neon;
 }

libavcodec/arm/rv34dsp_neon.S

@@ -19,10 +19,13 @@
  */
 
 #include "asm.S"
-#include "neon.S"
 
-.macro rv34_inv_transform    r0
-        vld1.16         {q14-q15}, [\r0,:128]
+.macro rv34_inv_transform
+        mov             r1,  #16
+        vld1.16         {d28}, [r0,:64], r1     @ block[i+8*0]
+        vld1.16         {d29}, [r0,:64], r1     @ block[i+8*1]
+        vld1.16         {d30}, [r0,:64], r1     @ block[i+8*2]
+        vld1.16         {d31}, [r0,:64], r1     @ block[i+8*3]
         vmov.s16        d0,  #13
         vshll.s16       q12, d29, #3
         vshll.s16       q13, d29, #4
@@ -32,12 +35,12 @@
         vmlal.s16       q10, d30, d0
         vmull.s16       q11, d28, d0
         vmlsl.s16       q11, d30, d0
-        vsubw.s16       q12, q12, d29   @ z2 = block[i+4*1]*7
-        vaddw.s16       q13, q13, d29   @ z3 = block[i+4*1]*17
+        vsubw.s16       q12, q12, d29   @ z2 = block[i+8*1]*7
+        vaddw.s16       q13, q13, d29   @ z3 = block[i+8*1]*17
         vsubw.s16       q9,  q9,  d31
         vaddw.s16       q1,  q1,  d31
-        vadd.s32        q13, q13, q9    @ z3 = 17*block[i+4*1] +  7*block[i+4*3]
-        vsub.s32        q12, q12, q1    @ z2 = 7*block[i+4*1]  - 17*block[i+4*3]
+        vadd.s32        q13, q13, q9    @ z3 = 17*block[i+8*1] +  7*block[i+8*3]
+        vsub.s32        q12, q12, q1    @ z2 = 7*block[i+8*1]  - 17*block[i+8*3]
         vadd.s32        q1,  q10, q13   @ z0 + z3
         vadd.s32        q2,  q11, q12   @ z1 + z2
         vsub.s32        q8,  q10, q13   @ z0 - z3
@@ -67,39 +70,25 @@
         vsub.s32        q15, q14, q9    @ z0 - z3
 .endm
 
-/* void rv34_idct_add_c(uint8_t *dst, int stride, DCTELEM *block) */
-function ff_rv34_idct_add_neon, export=1
-        mov             r3,  r0
-        rv34_inv_transform   r2
-        vmov.i16        q12, #0
-        vrshrn.s32      d16, q1,  #10   @ (z0 + z3) >> 10
-        vrshrn.s32      d17, q2,  #10   @ (z1 + z2) >> 10
-        vrshrn.s32      d18, q3,  #10   @ (z1 - z2) >> 10
-        vrshrn.s32      d19, q15, #10   @ (z0 - z3) >> 10
-        vld1.32         {d28[]},  [r0,:32], r1
-        vld1.32         {d29[]},  [r0,:32], r1
-        vtrn.32         q8,  q9
-        vld1.32         {d28[1]}, [r0,:32], r1
-        vld1.32         {d29[1]}, [r0,:32], r1
-        vst1.16         {q12}, [r2,:128]!       @ memset(block,    0, 16)
-        vst1.16         {q12}, [r2,:128]        @ memset(block+16, 0, 16)
-        vtrn.16         d16, d17
-        vtrn.32         d28, d29
-        vtrn.16         d18, d19
-        vaddw.u8        q0,   q8,  d28
-        vaddw.u8        q1,   q9,  d29
-        vqmovun.s16     d28,  q0
-        vqmovun.s16     d29,  q1
-        vst1.32         {d28[0]}, [r3,:32], r1
-        vst1.32         {d28[1]}, [r3,:32], r1
-        vst1.32         {d29[0]}, [r3,:32], r1
-        vst1.32         {d29[1]}, [r3,:32], r1
+/* void ff_rv34_inv_transform_neon(DCTELEM *block); */
+function ff_rv34_inv_transform_neon, export=1
+        mov             r2,  r0
+        rv34_inv_transform
+        vrshrn.s32      d1,  q2,  #10   @ (z1 + z2) >> 10
+        vrshrn.s32      d0,  q1,  #10   @ (z0 + z3) >> 10
+        vrshrn.s32      d2,  q3,  #10   @ (z1 - z2) >> 10
+        vrshrn.s32      d3,  q15, #10   @ (z0 - z3) >> 10
+        vst4.16         {d0[0], d1[0], d2[0], d3[0]}, [r2,:64], r1
+        vst4.16         {d0[1], d1[1], d2[1], d3[1]}, [r2,:64], r1
+        vst4.16         {d0[2], d1[2], d2[2], d3[2]}, [r2,:64], r1
+        vst4.16         {d0[3], d1[3], d2[3], d3[3]}, [r2,:64], r1
         bx              lr
 endfunc
 
 /* void rv34_inv_transform_noround_neon(DCTELEM *block); */
 function ff_rv34_inv_transform_noround_neon, export=1
-        rv34_inv_transform   r0
+        mov             r2,  r0
+        rv34_inv_transform
         vshl.s32        q11, q2,  #1
         vshl.s32        q10, q1,  #1
         vshl.s32        q12, q3,  #1
@@ -112,45 +101,9 @@ function ff_rv34_inv_transform_noround_neon, export=1
         vshrn.s32       d1,  q11, #11   @ (z1 + z2)*3 >> 11
         vshrn.s32       d2,  q12, #11   @ (z1 - z2)*3 >> 11
         vshrn.s32       d3,  q13, #11   @ (z0 - z3)*3 >> 11
-        vst4.16         {d0[0], d1[0], d2[0], d3[0]}, [r0,:64]!
-        vst4.16         {d0[1], d1[1], d2[1], d3[1]}, [r0,:64]!
-        vst4.16         {d0[2], d1[2], d2[2], d3[2]}, [r0,:64]!
-        vst4.16         {d0[3], d1[3], d2[3], d3[3]}, [r0,:64]!
-        bx              lr
-endfunc
-
-/* void ff_rv34_idct_dc_add_neon(uint8_t *dst, int stride, int dc) */
-function ff_rv34_idct_dc_add_neon, export=1
-        mov             r3,  r0
-        vld1.32         {d28[]},  [r0,:32], r1
-        vld1.32         {d29[]},  [r0,:32], r1
-        vdup.16         d0,  r2
-        vmov.s16        d1,  #169
-        vld1.32         {d28[1]}, [r0,:32], r1
-        vmull.s16       q1,  d0,  d1    @ dc * 13 * 13
-        vld1.32         {d29[1]}, [r0,:32], r1
-        vrshrn.s32      d0,  q1,  #10   @ (dc * 13 * 13 + 0x200) >> 10
-        vmov            d1,  d0
-        vaddw.u8        q2,  q0,  d28
-        vaddw.u8        q3,  q0,  d29
-        vqmovun.s16     d28, q2
-        vqmovun.s16     d29, q3
-        vst1.32         {d28[0]}, [r3,:32], r1
-        vst1.32         {d29[0]}, [r3,:32], r1
-        vst1.32         {d28[1]}, [r3,:32], r1
-        vst1.32         {d29[1]}, [r3,:32], r1
-        bx              lr
-endfunc
-
-/* void rv34_inv_transform_dc_noround_c(DCTELEM *block) */
-function ff_rv34_inv_transform_noround_dc_neon, export=1
-        vld1.16         {d28[]}, [r0,:16]       @ block[0]
-        vmov.i16        d4,  #251
-        vorr.s16        d4,  #256               @ 13^2 * 3
-        vmull.s16       q3,  d28, d4
-        vshrn.s32       d0,  q3,  #11
-        vmov.i16        d1,  d0
-        vst1.64         {q0}, [r0,:128]!
-        vst1.64         {q0}, [r0,:128]!
+        vst4.16         {d0[0], d1[0], d2[0], d3[0]}, [r2,:64], r1
+        vst4.16         {d0[1], d1[1], d2[1], d3[1]}, [r2,:64], r1
+        vst4.16         {d0[2], d1[2], d2[2], d3[2]}, [r2,:64], r1
+        vst4.16         {d0[3], d1[3], d2[3], d3[3]}, [r2,:64], r1
         bx              lr
 endfunc

libavcodec/arm/rv40dsp_init_neon.c

@@ -54,20 +54,6 @@ void ff_avg_rv40_chroma_mc4_neon(uint8_t *, uint8_t *, int, int, int, int);
 void ff_rv40_weight_func_16_neon(uint8_t *, uint8_t *, uint8_t *, int, int, int);
 void ff_rv40_weight_func_8_neon(uint8_t *, uint8_t *, uint8_t *, int, int, int);
 
-int ff_rv40_h_loop_filter_strength_neon(uint8_t *src, int stride,
-                                        int beta, int beta2, int edge,
-                                        int *p1, int *q1);
-int ff_rv40_v_loop_filter_strength_neon(uint8_t *src, int stride,
-                                        int beta, int beta2, int edge,
-                                        int *p1, int *q1);
-
-void ff_rv40_h_weak_loop_filter_neon(uint8_t *src, int stride, int filter_p1,
-                                     int filter_q1, int alpha, int beta,
-                                     int lim_p0q0, int lim_q1, int lim_p1);
-void ff_rv40_v_weak_loop_filter_neon(uint8_t *src, int stride, int filter_p1,
-                                     int filter_q1, int alpha, int beta,
-                                     int lim_p0q0, int lim_q1, int lim_p1);
-
 void ff_rv40dsp_init_neon(RV34DSPContext *c, DSPContext* dsp)
 {
     c->put_pixels_tab[0][ 1] = ff_put_rv40_qpel16_mc10_neon;
@@ -130,9 +116,4 @@ void ff_rv40dsp_init_neon(RV34DSPContext *c, DSPContext* dsp)
 
     c->rv40_weight_pixels_tab[0] = ff_rv40_weight_func_16_neon;
     c->rv40_weight_pixels_tab[1] = ff_rv40_weight_func_8_neon;
-
-    c->rv40_loop_filter_strength[0] = ff_rv40_h_loop_filter_strength_neon;
-    c->rv40_loop_filter_strength[1] = ff_rv40_v_loop_filter_strength_neon;
-    c->rv40_weak_loop_filter[0]     = ff_rv40_h_weak_loop_filter_neon;
-    c->rv40_weak_loop_filter[1]     = ff_rv40_v_weak_loop_filter_neon;
 }

libavcodec/arm/rv40dsp_neon.S

@@ -372,7 +372,7 @@ endfunc
 
 function ff_\type\()_rv40_qpel8_mc33_neon, export=1
         mov             r3,  #8
-        b               X(ff_\type\()_pixels8_xy2_neon)
+        b               ff_\type\()_pixels8_xy2_neon
 endfunc
 
 function ff_\type\()_rv40_qpel8_mc13_neon, export=1
@@ -652,7 +652,7 @@ endfunc
 
 function ff_\type\()_rv40_qpel16_mc33_neon, export=1
         mov             r3,  #16
-        b               X(ff_\type\()_pixels16_xy2_neon)
+        b               ff_\type\()_pixels16_xy2_neon
 endfunc
 .endm
 
@@ -722,199 +722,3 @@ function ff_rv40_weight_func_8_neon, export=1
         bne             1b
         bx              lr
 endfunc
-
-function ff_rv40_h_loop_filter_strength_neon, export=1
-        pkhbt           r2,  r3,  r2,  lsl #18
-
-        ldr             r3,  [r0]
-        ldr_dpre        r12, r0,  r1
-        teq             r3,  r12
-        beq             1f
-
-        sub             r0,  r0,  r1,  lsl #1
-
-        vld1.32         {d4[]},   [r0,:32], r1  @ -3
-        vld1.32         {d0[]},   [r0,:32], r1  @ -2
-        vld1.32         {d4[1]},  [r0,:32], r1  @ -1
-        vld1.32         {d5[]},   [r0,:32], r1  @  0
-        vld1.32         {d1[]},   [r0,:32], r1  @  1
-        vld1.32         {d5[0]},  [r0,:32], r1  @  2
-
-        vpaddl.u8       q8,  q0                 @ -2, -2, -2, -2,  1,  1,  1,  1
-        vpaddl.u8       q9,  q2                 @ -3, -3, -1, -1,  2,  2,  0,  0
-        vdup.32         d30, r2                 @ beta2, beta << 2
-        vpadd.u16       d16, d16, d17           @ -2, -2,  1,  1
-        vpadd.u16       d18, d18, d19           @ -3, -1,  2,  0
-        vabd.u16        d16, d18, d16
-        vclt.u16        d16, d16, d30
-
-        ldrd            r2,  r3,  [sp, #4]
-        vmovl.u16       q12, d16
-        vtrn.16         d16, d17
-        vshr.u32        q12, q12, #15
-        ldr             r0,  [sp]
-        vst1.32         {d24[1]}, [r2,:32]
-        vst1.32         {d25[1]}, [r3,:32]
-
-        cmp             r0,  #0
-        it              eq
-        bxeq            lr
-
-        vand            d18, d16, d17
-        vtrn.32         d18, d19
-        vand            d18, d18, d19
-        vmov.u16        r0,  d18[0]
-        bx              lr
-1:
-        ldrd            r2,  r3,  [sp, #4]
-        mov             r0,  #0
-        str             r0,  [r2]
-        str             r0,  [r3]
-        bx              lr
-endfunc
-
-function ff_rv40_v_loop_filter_strength_neon, export=1
-        sub             r0,  r0,  #3
-        pkhbt           r2,  r3,  r2,  lsl #18
-
-        vld1.8          {d0},     [r0], r1
-        vld1.8          {d1},     [r0], r1
-        vld1.8          {d2},     [r0], r1
-        vld1.8          {d3},     [r0], r1
-
-        vaddl.u8        q0,  d0,  d1
-        vaddl.u8        q1,  d2,  d3
-        vdup.32         q15, r2
-        vadd.u16        q0,  q0,  q1            @ -3, -2, -1,  0,  1,  2
-        vext.16         q1,  q0,  q0,  #1       @ -2, -1,  0,  1,  2
-        vabd.u16        q0,  q1,  q0
-        vclt.u16        q0,  q0,  q15
-
-        ldrd            r2,  r3,  [sp, #4]
-        vmovl.u16       q1,  d0
-        vext.16         d1,  d0,  d1,  #3
-        vshr.u32        q1,  q1,  #15
-        ldr             r0,  [sp]
-        vst1.32         {d2[1]},  [r2,:32]
-        vst1.32         {d3[1]},  [r3,:32]
-
-        cmp             r0,  #0
-        it              eq
-        bxeq            lr
-
-        vand            d0,  d0,  d1
-        vtrn.16         d0,  d1
-        vand            d0,  d0,  d1
-        vmov.u16        r0,  d0[0]
-        bx              lr
-endfunc
-
-.macro  rv40_weak_loop_filter
-        vdup.16         d30, r2                 @ filter_p1
-        vdup.16         d31, r3                 @ filter_q1
-        ldrd            r2,  r3,  [sp]
-        vdup.16         d28, r2                 @ alpha
-        vdup.16         d29, r3                 @ beta
-        ldr             r12, [sp, #8]
-        vdup.16         d25, r12                @ lim_p0q0
-        ldrd            r2,  r3,  [sp, #12]
-        vsubl.u8        q9,  d5,  d4            @ x, t
-        vabdl.u8        q8,  d5,  d4            @ x, abs(t)
-        vneg.s16        q15, q15
-        vceq.i16        d16, d19, #0            @ !t
-        vshl.s16        d19, d19, #2            @ t << 2
-        vmul.u16        d18, d17, d28           @ alpha * abs(t)
-        vand            d24, d30, d31           @ filter_p1 & filter_q1
-        vsubl.u8        q1,  d0,  d4            @ p1p2, p1p0
-        vsubl.u8        q3,  d1,  d5            @ q1q2, q1q0
-        vmov.i16        d22, #3
-        vshr.u16        d18, d18, #7
-        vadd.i16        d22, d22, d24           @ 3 - (filter_p1 & filter_q1)
-        vsubl.u8        q10, d0,  d1            @ src[-2] - src[1]
-        vcle.u16        d18, d18, d22
-        vand            d20, d20, d24
-        vneg.s16        d23, d25                @ -lim_p0q0
-        vadd.s16        d19, d19, d20
-        vbic            d16, d18, d16           @ t && u <= 3 - (fp1 & fq1)
-        vtrn.32         d4,  d5                 @ -3,  2, -1,  0
-        vrshr.s16       d19, d19, #3
-        vmov            d28, d29                @ beta
-        vswp            d3,  d6                 @ q1q2, p1p0
-        vmin.s16        d19, d19, d25
-        vand            d30, d30, d16
-        vand            d31, d31, d16
-        vadd.s16        q10, q1,  q3            @ p1p2 + p1p0, q1q2 + q1q0
-        vmax.s16        d19, d19, d23           @ diff
-        vabs.s16        q1,  q1                 @ abs(p1p2), abs(q1q2)
-        vand            d18, d19, d16           @ diff
-        vcle.u16        q1,  q1,  q14
-        vneg.s16        d19, d18                @ -diff
-        vdup.16         d26, r3                 @ lim_p1
-        vaddw.u8        q2,  q9,  d5            @ src[-1]+diff, src[0]-diff
-        vhsub.s16       q11, q10, q9
-        vand            q1,  q1,  q15
-        vqmovun.s16     d4,  q2                 @ -1,  0
-        vand            q9,  q11, q1
-        vdup.16         d27, r2                 @ lim_q1
-        vneg.s16        q9,  q9
-        vneg.s16        q14, q13
-        vmin.s16        q9,  q9,  q13
-        vtrn.32         d0,  d1                 @ -2,  1,  -2,  1
-        vmax.s16        q9,  q9,  q14
-        vaddw.u8        q3,  q9,  d0
-        vqmovun.s16     d5,  q3                 @ -2,  1
-.endm
-
-function ff_rv40_h_weak_loop_filter_neon, export=1
-        sub             r0,  r0,  r1,  lsl #1
-        sub             r0,  r0,  r1
-
-        vld1.32         {d4[]},   [r0,:32], r1
-        vld1.32         {d0[]},   [r0,:32], r1
-        vld1.32         {d4[1]},  [r0,:32], r1
-        vld1.32         {d5[]},   [r0,:32], r1
-        vld1.32         {d1[]},   [r0,:32], r1
-        vld1.32         {d5[0]},  [r0,:32]
-
-        sub             r0,  r0,  r1,  lsl #2
-
-        rv40_weak_loop_filter
-
-        vst1.32         {d5[0]},  [r0,:32], r1
-        vst1.32         {d4[0]},  [r0,:32], r1
-        vst1.32         {d4[1]},  [r0,:32], r1
-        vst1.32         {d5[1]},  [r0,:32], r1
-
-        bx              lr
-endfunc
-
-function ff_rv40_v_weak_loop_filter_neon, export=1
-        sub             r12, r0,  #3
-        sub             r0,  r0,  #2
-
-        vld1.8          {d4},     [r12], r1
-        vld1.8          {d5},     [r12], r1
-        vld1.8          {d2},     [r12], r1
-        vld1.8          {d3},     [r12], r1
-
-        vtrn.16         q2,  q1
-        vtrn.8          d4,  d5
-        vtrn.8          d2,  d3
-
-        vrev64.32       d5,  d5
-        vtrn.32         q2,  q1
-        vdup.32         d0,  d3[0]
-        vdup.32         d1,  d2[0]
-
-        rv40_weak_loop_filter
-
-        vtrn.32         q2,  q3
-        vswp            d4,  d5
-
-        vst4.8          {d4[0],d5[0],d6[0],d7[0]}, [r0], r1
-        vst4.8          {d4[1],d5[1],d6[1],d7[1]}, [r0], r1
-        vst4.8          {d4[2],d5[2],d6[2],d7[2]}, [r0], r1
-        vst4.8          {d4[3],d5[3],d6[3],d7[3]}, [r0], r1
-
-        bx              lr
-endfunc

libavcodec/arm/simple_idct_arm.S

@@ -491,8 +491,8 @@ __end_bef_a_evaluation:
         bal __end_a_evaluation
 
 
-        .align
 __constant_ptr__:  @@ see #defines at the beginning of the source code for values.
+        .align
         .word   W1
         .word   W2
         .word   W3

libavcodec/ass_split.c

@@ -366,7 +366,7 @@ int ff_ass_split_override_codes(const ASSCodesCallbacks *callbacks, void *priv,
     char new_line[2];
     int text_len = 0;
 
-    while (*buf) {
+    while (buf && *buf) {
         if (text && callbacks->text &&
             (sscanf(buf, "\\%1[nN]", new_line) == 1 ||
              !strncmp(buf, "{\\", 2))) {

libavcodec/asv1.c

@@ -408,7 +408,7 @@ static int decode_frame(AVCodecContext *avctx,
     p->pict_type= AV_PICTURE_TYPE_I;
     p->key_frame= 1;
 
-    av_fast_padded_malloc(&a->bitstream_buffer, &a->bitstream_buffer_size, buf_size);
+    av_fast_malloc(&a->bitstream_buffer, &a->bitstream_buffer_size, buf_size + FF_INPUT_BUFFER_PADDING_SIZE);
     if (!a->bitstream_buffer)
         return AVERROR(ENOMEM);
 

libavcodec/atrac3.c

@@ -744,7 +744,7 @@ static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf,
 
         result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, out_samples[0], 0, JOINT_STEREO);
         if (result != 0)
-            return result;
+            return (result);
 
         /* Framedata of the su2 in the joint-stereo mode is encoded in
          * reverse byte order so we need to swap it first. */
@@ -785,7 +785,7 @@ static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf,
         /* Decode Sound Unit 2. */
         result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], out_samples[1], 1, JOINT_STEREO);
         if (result != 0)
-            return result;
+            return (result);
 
         /* Reconstruct the channel coefficients. */
         reverseMatrixing(out_samples[0], out_samples[1], q->matrix_coeff_index_prev, q->matrix_coeff_index_now);
@@ -804,7 +804,7 @@ static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf,
 
             result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], out_samples[i], i, q->codingMode);
             if (result != 0)
-                return result;
+                return (result);
         }
     }
 

libavcodec/avcodec.h

@@ -255,15 +255,11 @@ enum CodecID {
     CODEC_ID_VBLE,
     CODEC_ID_DXTORY,
     CODEC_ID_V410,
-    CODEC_ID_XWD,
     CODEC_ID_Y41P       = MKBETAG('Y','4','1','P'),
     CODEC_ID_UTVIDEO = 0x800,
     CODEC_ID_ESCAPE130  = MKBETAG('E','1','3','0'),
-    CODEC_ID_AVRP       = MKBETAG('A','V','R','P'),
 
     CODEC_ID_G2M        = MKBETAG( 0 ,'G','2','M'),
-    CODEC_ID_V308       = MKBETAG('V','3','0','8'),
-    CODEC_ID_YUV4       = MKBETAG('Y','U','V','4'),
 
     /* various PCM "codecs" */
     CODEC_ID_FIRST_AUDIO = 0x10000,     ///< A dummy id pointing at the start of audio codecs
@@ -326,7 +322,6 @@ enum CodecID {
     CODEC_ID_ADPCM_EA_MAXIS_XA,
     CODEC_ID_ADPCM_IMA_ISS,
     CODEC_ID_ADPCM_G722,
-    CODEC_ID_ADPCM_IMA_APC,
 
     /* AMR */
     CODEC_ID_AMR_NB = 0x12000,
@@ -409,7 +404,6 @@ enum CodecID {
     CODEC_ID_BMV_AUDIO,
     CODEC_ID_G729 = 0x15800,
     CODEC_ID_G723_1= 0x15801,
-    CODEC_ID_FFWAVESYNTH = MKBETAG('F','F','W','S'),
     CODEC_ID_8SVX_RAW   = MKBETAG('8','S','V','X'),
 
     /* subtitle codecs */
@@ -747,27 +741,10 @@ typedef struct RcOverride{
 /* Codec can export data for HW decoding (XvMC). */
 #define CODEC_CAP_HWACCEL         0x0010
 /**
- * Encoder or decoder requires flushing with NULL input at the end in order to
- * give the complete and correct output.
- *
- * NOTE: If this flag is not set, the codec is guaranteed to never be fed with
- *       with NULL data. The user can still send NULL data to the public encode
- *       or decode function, but libavcodec will not pass it along to the codec
- *       unless this flag is set.
- *
- * Decoders:
- * The decoder has a non-zero delay and needs to be fed with avpkt->data=NULL,
+ * Codec has a nonzero delay and needs to be fed with avpkt->data=NULL,
  * avpkt->size=0 at the end to get the delayed data until the decoder no longer
- * returns frames.
- *
- * Encoders:
- * The encoder needs to be fed with NULL data at the end of encoding until the
- * encoder no longer returns data.
- *
- * NOTE: For encoders implementing the AVCodec.encode2() function, setting this
- *       flag also means that the encoder must set the pts and duration for
- *       each output packet. If this flag is not set, the pts and duration will
- *       be determined by libavcodec from the input frame.
+ * returns frames. If this is not set, the codec is guaranteed to never be fed
+ * with NULL data.
  */
 #define CODEC_CAP_DELAY           0x0020
 /**
@@ -818,14 +795,6 @@ typedef struct RcOverride{
  * Codec supports changed parameters at any point.
  */
 #define CODEC_CAP_PARAM_CHANGE     0x4000
-/**
- * Codec supports avctx->thread_count == 0 (auto).
- */
-#define CODEC_CAP_AUTO_THREADS     0x8000
-/**
- * Audio encoder supports receiving a different number of samples in each call.
- */
-#define CODEC_CAP_VARIABLE_FRAME_SIZE 0x10000
 /**
  * Codec is lossless.
  */
@@ -1087,12 +1056,13 @@ typedef struct AVFrame {
      */
     int quality;
 
-#if FF_API_AVFRAME_AGE
     /**
-     * @deprecated unused
+     * buffer age (1->was last buffer and dint change, 2->..., ...).
+     * Set to INT_MAX if the buffer has not been used yet.
+     * - encoding: unused
+     * - decoding: MUST be set by get_buffer() for video.
      */
-    attribute_deprecated int age;
-#endif
+    int age;
 
     /**
      * is this picture used as reference
@@ -1315,29 +1285,6 @@ typedef struct AVFrame {
      */
     uint8_t **extended_data;
 
-    /**
-     * sample aspect ratio for the video frame, 0/1 if unknown\unspecified
-     * - encoding: unused
-     * - decoding: Read by user.
-     */
-    AVRational sample_aspect_ratio;
-
-    /**
-     * width and height of the video frame
-     * - encoding: unused
-     * - decoding: Read by user.
-     */
-    int width, height;
-
-    /**
-     * format of the frame, -1 if unknown or unset
-     * Values correspond to enum PixelFormat for video frames,
-     * enum AVSampleFormat for audio)
-     * - encoding: unused
-     * - decoding: Read by user.
-     */
-    int format;
-
     /**
      * frame timestamp estimated using various heuristics, in stream time base
      * Code outside libavcodec should access this field using:
@@ -1356,19 +1303,39 @@ typedef struct AVFrame {
      */
     int64_t pkt_pos;
 
+    /**
+     * reordered sample aspect ratio for the video frame, 0/1 if unknown\unspecified
+     * Code outside libavcodec should access this field using:
+     *  av_opt_ptr(avcodec_get_frame_class(), frame, "sample_aspect_ratio");
+     * - encoding: unused
+     * - decoding: Read by user.
+     */
+    AVRational sample_aspect_ratio;
+
+    /**
+     * width and height of the video frame
+     * Code outside libavcodec should access this field using:
+     *  av_opt_ptr(avcodec_get_frame_class(), frame, "width");
+     * - encoding: unused
+     * - decoding: Read by user.
+     */
+    int width, height;
+
+    /**
+     * format of the frame, -1 if unknown or unset
+     * It should be cast to the corresponding enum (enum PixelFormat
+     * for video, enum AVSampleFormat for audio)
+     * Code outside libavcodec should access this field using:
+     *  av_opt_ptr(avcodec_get_frame_class(), frame, "format");
+     * - encoding: unused
+     * - decoding: Read by user.
+     */
+    int format;
+
 } AVFrame;
 
 struct AVCodecInternal;
 
-enum AVFieldOrder {
-    AV_FIELD_UNKNOWN,
-    AV_FIELD_PROGRESSIVE,
-    AV_FIELD_TT,          //< Top coded_first, top displayed first
-    AV_FIELD_BB,          //< Bottom coded first, bottom displayed first
-    AV_FIELD_TB,          //< Top coded first, bottom displayed first
-    AV_FIELD_BT,          //< Bottom coded first, top displayed first
-};
-
 /**
  * main external API structure.
  * New fields can be added to the end with minor version bumps.
@@ -1410,7 +1377,7 @@ typedef struct AVCodecContext {
      * Some codecs need additional format info. It is stored here.
      * If any muxer uses this then ALL demuxers/parsers AND encoders for the
      * specific codec MUST set it correctly otherwise stream copy breaks.
-     * In general use of this field by muxers is not recommended.
+     * In general use of this field by muxers is not recommanded.
      * - encoding: Set by libavcodec.
      * - decoding: Set by libavcodec. (FIXME: Is this OK?)
      */
@@ -2729,7 +2696,7 @@ typedef struct AVCodecContext {
 
 #if FF_API_X264_GLOBAL_OPTS
     /**
-     * Influence how often B-frames are used.
+     * Influences how often B-frames are used.
      * - encoding: Set by user.
      * - decoding: unused
      */
@@ -2810,7 +2777,7 @@ typedef struct AVCodecContext {
     int mv0_threshold;
 
     /**
-     * Adjust sensitivity of b_frame_strategy 1.
+     * Adjusts sensitivity of b_frame_strategy 1.
      * - encoding: Set by user.
      * - decoding: unused
      */
@@ -3094,7 +3061,7 @@ typedef struct AVCodecContext {
 
 #if FF_API_FLAC_GLOBAL_OPTS
     /**
-     * Determine which LPC analysis algorithm to use.
+     * Determines which LPC analysis algorithm to use.
      * - encoding: Set by user
      * - decoding: unused
      */
@@ -3225,12 +3192,6 @@ typedef struct AVCodecContext {
      */
     struct AVCodecInternal *internal;
 
-    /** Field order
-     * - encoding: set by libavcodec
-     * - decoding: Set by libavcodec
-     */
-    enum AVFieldOrder field_order;
-
     /**
      * Current statistics for PTS correction.
      * - decoding: maintained and used by libavcodec, not intended to be used by user apps
@@ -3325,19 +3286,6 @@ typedef struct AVCodec {
      * Initialize codec static data, called from avcodec_register().
      */
     void (*init_static_data)(struct AVCodec *codec);
-
-    /**
-     * Encode data to an AVPacket.
-     *
-     * @param      avctx          codec context
-     * @param      avpkt          output AVPacket (may contain a user-provided buffer)
-     * @param[in]  frame          AVFrame containing the raw data to be encoded
-     * @param[out] got_packet_ptr encoder sets to 0 or 1 to indicate that a
-     *                            non-empty packet was returned in avpkt.
-     * @return 0 on success, negative error code on failure
-     */
-    int (*encode2)(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame,
-                   int *got_packet_ptr);
 } AVCodec;
 
 /**
@@ -3622,7 +3570,7 @@ typedef struct ReSampleContext ReSampleContext;
  * @param linear           if 1 then the used FIR filter will be linearly interpolated
                            between the 2 closest, if 0 the closest will be used
  * @param cutoff           cutoff frequency, 1.0 corresponds to half the output sampling rate
- * @return allocated ReSampleContext, NULL if error occurred
+ * @return allocated ReSampleContext, NULL if error occured
  */
 ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
                                         int output_rate, int input_rate,
@@ -4212,11 +4160,6 @@ int avcodec_open2(AVCodecContext *avctx, AVCodec *codec, AVDictionary **options)
  * @warning The end of the input buffer avpkt->data should be set to 0 to ensure that
  * no overreading happens for damaged MPEG streams.
  *
- * @warning You must not provide a custom get_buffer() when using
- * avcodec_decode_audio3().  Doing so will override it with
- * avcodec_default_get_buffer.  Use avcodec_decode_audio4() instead,
- * which does allow the application to provide a custom get_buffer().
- *
  * @note You might have to align the input buffer avpkt->data and output buffer
  * samples. The alignment requirements depend on the CPU: On some CPUs it isn't
  * necessary at all, on others it won't work at all if not aligned and on others
@@ -4326,7 +4269,7 @@ int avcodec_decode_audio4(AVCodecContext *avctx, AVFrame *frame,
  */
 int avcodec_decode_video2(AVCodecContext *avctx, AVFrame *picture,
                          int *got_picture_ptr,
-                         const AVPacket *avpkt);
+                         AVPacket *avpkt);
 
 /**
  * Decode a subtitle message.
@@ -4349,22 +4292,19 @@ int avcodec_decode_subtitle2(AVCodecContext *avctx, AVSubtitle *sub,
                             AVPacket *avpkt);
 
 /**
- * Free all allocated data in the given subtitle struct.
+ * Frees all allocated data in the given subtitle struct.
  *
  * @param sub AVSubtitle to free.
  */
 void avsubtitle_free(AVSubtitle *sub);
 
-#if FF_API_OLD_ENCODE_AUDIO
 /**
  * Encode an audio frame from samples into buf.
  *
- * @deprecated Use avcodec_encode_audio2 instead.
- *
  * @note The output buffer should be at least FF_MIN_BUFFER_SIZE bytes large.
- * However, for codecs with avctx->frame_size equal to 0 (e.g. PCM) the user
- * will know how much space is needed because it depends on the value passed
- * in buf_size as described below. In that case a lower value can be used.
+ * However, for PCM audio the user will know how much space is needed
+ * because it depends on the value passed in buf_size as described
+ * below. In that case a lower value can be used.
  *
  * @param avctx the codec context
  * @param[out] buf the output buffer
@@ -4372,79 +4312,13 @@ void avsubtitle_free(AVSubtitle *sub);
  * @param[in] samples the input buffer containing the samples
  * The number of samples read from this buffer is frame_size*channels,
  * both of which are defined in avctx.
- * For codecs which have avctx->frame_size equal to 0 (e.g. PCM) the number of
- * samples read from samples is equal to:
- * buf_size * 8 / (avctx->channels * av_get_bits_per_sample(avctx->codec_id))
- * This also implies that av_get_bits_per_sample() must not return 0 for these
- * codecs.
+ * For PCM audio the number of samples read from samples is equal to
+ * buf_size * input_sample_size / output_sample_size.
  * @return On error a negative value is returned, on success zero or the number
  * of bytes used to encode the data read from the input buffer.
  */
-int attribute_deprecated avcodec_encode_audio(AVCodecContext *avctx,
-                                              uint8_t *buf, int buf_size,
-                                              const short *samples);
-#endif
-
-/**
- * Encode a frame of audio.
- *
- * Takes input samples from frame and writes the next output packet, if
- * available, to avpkt. The output packet does not necessarily contain data for
- * the most recent frame, as encoders can delay, split, and combine input frames
- * internally as needed.
- *
- * @param avctx     codec context
- * @param avpkt     output AVPacket.
- *                  The user can supply an output buffer by setting
- *                  avpkt->data and avpkt->size prior to calling the
- *                  function, but if the size of the user-provided data is not
- *                  large enough, encoding will fail. All other AVPacket fields
- *                  will be reset by the encoder using av_init_packet(). If
- *                  avpkt->data is NULL, the encoder will allocate it.
- *                  The encoder will set avpkt->size to the size of the
- *                  output packet.
- * @param[in] frame AVFrame containing the raw audio data to be encoded.
- *                  May be NULL when flushing an encoder that has the
- *                  CODEC_CAP_DELAY capability set.
- *                  There are 2 codec capabilities that affect the allowed
- *                  values of frame->nb_samples.
- *                  If CODEC_CAP_SMALL_LAST_FRAME is set, then only the final
- *                  frame may be smaller than avctx->frame_size, and all other
- *                  frames must be equal to avctx->frame_size.
- *                  If CODEC_CAP_VARIABLE_FRAME_SIZE is set, then each frame
- *                  can have any number of samples.
- *                  If neither is set, frame->nb_samples must be equal to
- *                  avctx->frame_size for all frames.
- * @param[out] got_packet_ptr This field is set to 1 by libavcodec if the
- *                            output packet is non-empty, and to 0 if it is
- *                            empty. If the function returns an error, the
- *                            packet can be assumed to be invalid, and the
- *                            value of got_packet_ptr is undefined and should
- *                            not be used.
- * @return          0 on success, negative error code on failure
- */
-int avcodec_encode_audio2(AVCodecContext *avctx, AVPacket *avpkt,
-                          const AVFrame *frame, int *got_packet_ptr);
-
-/**
- * Fill audio frame data and linesize.
- * AVFrame extended_data channel pointers are allocated if necessary for
- * planar audio.
- *
- * @param frame       the AVFrame
- *                    frame->nb_samples must be set prior to calling the
- *                    function. This function fills in frame->data,
- *                    frame->extended_data, frame->linesize[0].
- * @param nb_channels channel count
- * @param sample_fmt  sample format
- * @param buf         buffer to use for frame data
- * @param buf_size    size of buffer
- * @param align       plane size sample alignment
- * @return            0 on success, negative error code on failure
- */
-int avcodec_fill_audio_frame(AVFrame *frame, int nb_channels,
-                             enum AVSampleFormat sample_fmt, const uint8_t *buf,
-                             int buf_size, int align);
+int avcodec_encode_audio(AVCodecContext *avctx, uint8_t *buf, int buf_size,
+                         const short *samples);
 
 /**
  * Encode a video frame from pict into buf.
@@ -4755,15 +4629,6 @@ void *av_fast_realloc(void *ptr, unsigned int *size, size_t min_size);
  */
 void av_fast_malloc(void *ptr, unsigned int *size, size_t min_size);
 
-/**
- * Same behaviour av_fast_malloc but the buffer has additional
- * FF_INPUT_PADDING_SIZE at the end which will will always be 0.
- *
- * In addition the whole buffer will initially and after resizes
- * be 0-initialized so that no uninitialized data will ever appear.
- */
-void av_fast_padded_malloc(void *ptr, unsigned int *size, size_t min_size);
-
 /**
  * Copy image src to dst. Wraps av_picture_data_copy() above.
  */
@@ -4792,7 +4657,7 @@ int av_picture_pad(AVPicture *dst, const AVPicture *src, int height, int width,
 unsigned int av_xiphlacing(unsigned char *s, unsigned int v);
 
 /**
- * Log a generic warning message about a missing feature. This function is
+ * Logs a generic warning message about a missing feature. This function is
  * intended to be used internally by FFmpeg (libavcodec, libavformat, etc.)
  * only, and would normally not be used by applications.
  * @param[in] avc a pointer to an arbitrary struct of which the first field is

libavcodec/avpacket.c

@@ -20,7 +20,6 @@
  */
 
 #include "avcodec.h"
-#include "internal.h"
 #include "libavutil/avassert.h"
 #include "bytestream.h"
 
@@ -31,23 +30,19 @@ void av_destruct_packet_nofree(AVPacket *pkt)
     pkt->side_data_elems = 0;
 }
 
-void ff_packet_free_side_data(AVPacket *pkt)
+void av_destruct_packet(AVPacket *pkt)
 {
     int i;
+
+    av_free(pkt->data);
+    pkt->data = NULL; pkt->size = 0;
+
     for (i = 0; i < pkt->side_data_elems; i++)
         av_free(pkt->side_data[i].data);
     av_freep(&pkt->side_data);
     pkt->side_data_elems = 0;
 }
 
-void av_destruct_packet(AVPacket *pkt)
-{
-    av_free(pkt->data);
-    pkt->data = NULL; pkt->size = 0;
-
-    ff_packet_free_side_data(pkt);
-}
-
 void av_init_packet(AVPacket *pkt)
 {
     pkt->pts   = AV_NOPTS_VALUE;
@@ -244,6 +239,8 @@ int av_packet_split_side_data(AVPacket *pkt){
         unsigned int size;
         uint8_t *p;
 
+        av_dup_packet(pkt);
+
         p = pkt->data + pkt->size - 8 - 5;
         for (i=1; ; i++){
             size = AV_RB32(p);

libavcodec/avs.c

@@ -165,15 +165,6 @@ static av_cold int avs_decode_init(AVCodecContext * avctx)
     return 0;
 }
 
-static av_cold int avs_decode_end(AVCodecContext *avctx)
-{
-    AvsContext *s = avctx->priv_data;
-    if (s->picture.data[0])
-        avctx->release_buffer(avctx, &s->picture);
-    return 0;
-}
-
-
 AVCodec ff_avs_decoder = {
     .name           = "avs",
     .type           = AVMEDIA_TYPE_VIDEO,
@@ -181,7 +172,6 @@ AVCodec ff_avs_decoder = {
     .priv_data_size = sizeof(AvsContext),
     .init           = avs_decode_init,
     .decode         = avs_decode_frame,
-    .close          = avs_decode_end,
     .capabilities   = CODEC_CAP_DR1,
     .long_name = NULL_IF_CONFIG_SMALL("AVS (Audio Video Standard) video"),
 };

libavcodec/bethsoftvideo.c

@@ -34,7 +34,6 @@
 
 typedef struct BethsoftvidContext {
     AVFrame frame;
-    GetByteContext g;
 } BethsoftvidContext;
 
 static av_cold int bethsoftvid_decode_init(AVCodecContext *avctx)
@@ -48,19 +47,19 @@ static av_cold int bethsoftvid_decode_init(AVCodecContext *avctx)
     return 0;
 }
 
-static int set_palette(BethsoftvidContext *ctx)
+static int set_palette(AVFrame * frame, const uint8_t * palette_buffer, int buf_size)
 {
-    uint32_t *palette = (uint32_t *)ctx->frame.data[1];
+    uint32_t * palette = (uint32_t *)frame->data[1];
     int a;
 
-    if (bytestream2_get_bytes_left(&ctx->g) < 256*3)
+    if (buf_size < 256*3)
         return AVERROR_INVALIDDATA;
 
     for(a = 0; a < 256; a++){
-        palette[a] = 0xFFU << 24 | bytestream2_get_be24u(&ctx->g) * 4;
+        palette[a] = 0xFF << 24 | AV_RB24(&palette_buffer[a * 3]) * 4;
         palette[a] |= palette[a] >> 6 & 0x30303;
     }
-    ctx->frame.palette_has_changed = 1;
+    frame->palette_has_changed = 1;
     return 256*3;
 }
 
@@ -68,6 +67,8 @@ static int bethsoftvid_decode_frame(AVCodecContext *avctx,
                               void *data, int *data_size,
                               AVPacket *avpkt)
 {
+    const uint8_t *buf = avpkt->data;
+    int buf_size = avpkt->size;
     BethsoftvidContext * vid = avctx->priv_data;
     char block_type;
     uint8_t * dst;
@@ -81,32 +82,29 @@ static int bethsoftvid_decode_frame(AVCodecContext *avctx,
         av_log(avctx, AV_LOG_ERROR, "reget_buffer() failed\n");
         return -1;
     }
-
-    bytestream2_init(&vid->g, avpkt->data, avpkt->size);
     dst = vid->frame.data[0];
     frame_end = vid->frame.data[0] + vid->frame.linesize[0] * avctx->height;
 
-    switch(block_type = bytestream2_get_byte(&vid->g)){
-        case PALETTE_BLOCK: {
-            return set_palette(vid);
-        }
+    switch(block_type = *buf++){
+        case PALETTE_BLOCK:
+            return set_palette(&vid->frame, buf, buf_size);
         case VIDEO_YOFF_P_FRAME:
-            yoffset = bytestream2_get_le16(&vid->g);
+            yoffset = bytestream_get_le16(&buf);
             if(yoffset >= avctx->height)
                 return -1;
             dst += vid->frame.linesize[0] * yoffset;
     }
 
     // main code
-    while((code = bytestream2_get_byte(&vid->g))){
+    while((code = *buf++)){
         int length = code & 0x7f;
 
         // copy any bytes starting at the current position, and ending at the frame width
         while(length > remaining){
             if(code < 0x80)
-                bytestream2_get_buffer(&vid->g, dst, remaining);
+                bytestream_get_buffer(&buf, dst, remaining);
             else if(block_type == VIDEO_I_FRAME)
-                memset(dst, bytestream2_peek_byte(&vid->g), remaining);
+                memset(dst, buf[0], remaining);
             length -= remaining;      // decrement the number of bytes to be copied
             dst += remaining + wrap_to_next_line;    // skip over extra bytes at end of frame
             remaining = avctx->width;
@@ -116,9 +114,9 @@ static int bethsoftvid_decode_frame(AVCodecContext *avctx,
 
         // copy any remaining bytes after / if line overflows
         if(code < 0x80)
-            bytestream2_get_buffer(&vid->g, dst, length);
+            bytestream_get_buffer(&buf, dst, length);
         else if(block_type == VIDEO_I_FRAME)
-            memset(dst, bytestream2_get_byte(&vid->g), length);
+            memset(dst, *buf++, length);
         remaining -= length;
         dst += length;
     }
@@ -127,7 +125,7 @@ static int bethsoftvid_decode_frame(AVCodecContext *avctx,
     *data_size = sizeof(AVFrame);
     *(AVFrame*)data = vid->frame;
 
-    return avpkt->size;
+    return buf_size;
 }
 
 static av_cold int bethsoftvid_decode_end(AVCodecContext *avctx)

libavcodec/bfi.c

@@ -37,7 +37,7 @@ typedef struct BFIContext {
     uint32_t pal[256];
 } BFIContext;
 
-static av_cold int bfi_decode_init(AVCodecContext *avctx)
+static av_cold int bfi_decode_init(AVCodecContext * avctx)
 {
     BFIContext *bfi = avctx->priv_data;
     avctx->pix_fmt = PIX_FMT_PAL8;
@@ -46,10 +46,10 @@ static av_cold int bfi_decode_init(AVCodecContext *avctx)
     return 0;
 }
 
-static int bfi_decode_frame(AVCodecContext *avctx, void *data,
+static int bfi_decode_frame(AVCodecContext * avctx, void *data,
                             int *data_size, AVPacket *avpkt)
 {
-    GetByteContext g;
+    const uint8_t *buf = avpkt->data, *buf_end = avpkt->data + avpkt->size;
     int buf_size = avpkt->size;
     BFIContext *bfi = avctx->priv_data;
     uint8_t *dst = bfi->dst;
@@ -68,18 +68,16 @@ static int bfi_decode_frame(AVCodecContext *avctx, void *data,
         return -1;
     }
 
-    bytestream2_init(&g, avpkt->data, buf_size);
-
     /* Set frame parameters and palette, if necessary */
     if (!avctx->frame_number) {
         bfi->frame.pict_type = AV_PICTURE_TYPE_I;
         bfi->frame.key_frame = 1;
         /* Setting the palette */
-        if (avctx->extradata_size > 768) {
+        if(avctx->extradata_size>768) {
             av_log(NULL, AV_LOG_ERROR, "Palette is too large.\n");
             return -1;
         }
-        pal = (uint32_t *)bfi->frame.data[1];
+        pal = (uint32_t *) bfi->frame.data[1];
         for (i = 0; i < avctx->extradata_size / 3; i++) {
             int shift = 16;
             *pal = 0xFF << 24;
@@ -98,47 +96,46 @@ static int bfi_decode_frame(AVCodecContext *avctx, void *data,
         memcpy(bfi->frame.data[1], bfi->pal, sizeof(bfi->pal));
     }
 
-    bytestream2_skip(&g, 4); // Unpacked size, not required.
+    buf += 4; //Unpacked size, not required.
 
     while (dst != frame_end) {
-        static const uint8_t lentab[4] = { 0, 2, 0, 1 };
-        unsigned int byte   = bytestream2_get_byte(&g), av_uninit(offset);
-        unsigned int code   = byte >> 6;
+        static const uint8_t lentab[4]={0,2,0,1};
+        unsigned int byte = *buf++, av_uninit(offset);
+        unsigned int code = byte >> 6;
         unsigned int length = byte & ~0xC0;
 
-        if (!bytestream2_get_bytes_left(&g)) {
-            av_log(avctx, AV_LOG_ERROR,
-                   "Input resolution larger than actual frame.\n");
+        if (buf >= buf_end) {
+            av_log(avctx, AV_LOG_ERROR, "Input resolution larger than actual frame.\n");
             return -1;
         }
 
         /* Get length and offset(if required) */
         if (length == 0) {
             if (code == 1) {
-                length = bytestream2_get_byte(&g);
-                offset = bytestream2_get_le16(&g);
+                length = bytestream_get_byte(&buf);
+                offset = bytestream_get_le16(&buf);
             } else {
-                length = bytestream2_get_le16(&g);
+                length = bytestream_get_le16(&buf);
                 if (code == 2 && length == 0)
                     break;
             }
         } else {
             if (code == 1)
-                offset = bytestream2_get_byte(&g);
+                offset = bytestream_get_byte(&buf);
         }
 
         /* Do boundary check */
-        if (dst + (length << lentab[code]) > frame_end)
+        if (dst + (length<<lentab[code]) > frame_end)
             break;
 
         switch (code) {
 
         case 0:                //Normal Chain
-            if (length >= bytestream2_get_bytes_left(&g)) {
+            if (length >= buf_end - buf) {
                 av_log(avctx, AV_LOG_ERROR, "Frame larger than buffer.\n");
                 return -1;
             }
-            bytestream2_get_buffer(&g, dst, length);
+            bytestream_get_buffer(&buf, dst, length);
             dst += length;
             break;
 
@@ -156,8 +153,8 @@ static int bfi_decode_frame(AVCodecContext *avctx, void *data,
             break;
 
         case 3:                //Fill Chain
-            colour1 = bytestream2_get_byte(&g);
-            colour2 = bytestream2_get_byte(&g);
+            colour1 = bytestream_get_byte(&buf);
+            colour2 = bytestream_get_byte(&buf);
             while (length--) {
                 *dst++ = colour1;
                 *dst++ = colour2;
@@ -175,7 +172,7 @@ static int bfi_decode_frame(AVCodecContext *avctx, void *data,
         dst += bfi->frame.linesize[0];
     }
     *data_size = sizeof(AVFrame);
-    *(AVFrame *)data = bfi->frame;
+    *(AVFrame *) data = bfi->frame;
     return buf_size;
 }
 

libavcodec/bgmc.c

@@ -474,8 +474,7 @@ int ff_bgmc_init(AVCodecContext *avctx, uint8_t **cf_lut, int **cf_lut_status)
         av_log(avctx, AV_LOG_ERROR, "Allocating buffer memory failed.\n");
         return AVERROR(ENOMEM);
     } else {
-        // initialize lut_status buffer to a value never used to compare
-        // against
+        // initialize lut_status buffer to a value never used to compare against
         memset(*cf_lut_status, -1, sizeof(*cf_lut_status) * LUT_BUFF);
     }
 
@@ -495,7 +494,7 @@ void ff_bgmc_end(uint8_t **cf_lut, int **cf_lut_status)
 /** Initialize decoding and reads the first value
  */
 void ff_bgmc_decode_init(GetBitContext *gb,
-                         unsigned int *h, unsigned int *l, unsigned int *v)
+                      unsigned int *h, unsigned int *l, unsigned int *v)
 {
     *h = TOP_VALUE;
     *l = 0;
@@ -514,9 +513,9 @@ void ff_bgmc_decode_end(GetBitContext *gb)
 /** Read and decode a block Gilbert-Moore coded symbol
  */
 void ff_bgmc_decode(GetBitContext *gb, unsigned int num, int32_t *dst,
-                    int delta, unsigned int sx,
-                    unsigned int *h, unsigned int *l, unsigned int *v,
-                    uint8_t *cf_lut, int *cf_lut_status)
+                 int delta, unsigned int sx,
+                 unsigned int *h, unsigned int *l, unsigned int *v,
+                 uint8_t *cf_lut, int *cf_lut_status)
 {
     unsigned int i;
     uint8_t *lut = bgmc_lut_getp(cf_lut, cf_lut_status, delta);
@@ -568,3 +567,4 @@ void ff_bgmc_decode(GetBitContext *gb, unsigned int num, int32_t *dst,
     *l = low;
     *v = value;
 }
+

libavcodec/bink.c

@@ -27,7 +27,7 @@
 #include "binkdsp.h"
 #include "mathops.h"
 
-#define BITSTREAM_READER_LE
+#define ALT_BITSTREAM_READER_LE
 #include "get_bits.h"
 
 #define BINK_FLAG_ALPHA 0x00100000

libavcodec/binkaudio.c

@@ -29,13 +29,13 @@
  */
 
 #include "avcodec.h"
-#define BITSTREAM_READER_LE
+#define ALT_BITSTREAM_READER_LE
 #include "get_bits.h"
 #include "dsputil.h"
 #include "dct.h"
 #include "rdft.h"
 #include "fmtconvert.h"
-#include "libavutil/intfloat.h"
+#include "libavutil/intfloat_readwrite.h"
 
 extern const uint16_t ff_wma_critical_freqs[25];
 
@@ -193,8 +193,8 @@ static int decode_block(BinkAudioContext *s, int16_t *out, int use_dct)
         if (s->version_b) {
             if (get_bits_left(gb) < 64)
                 return AVERROR_INVALIDDATA;
-            coeffs[0] = av_int2float(get_bits_long(gb, 32)) * s->root;
-            coeffs[1] = av_int2float(get_bits_long(gb, 32)) * s->root;
+            coeffs[0] = av_int2flt(get_bits(gb, 32)) * s->root;
+            coeffs[1] = av_int2flt(get_bits(gb, 32)) * s->root;
         } else {
             if (get_bits_left(gb) < 58)
                 return AVERROR_INVALIDDATA;

libavcodec/bintext.c

@@ -139,7 +139,7 @@ static int decode_frame(AVCodecContext *avctx,
         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
         return -1;
     }
-    s->frame.pict_type           = AV_PICTURE_TYPE_I;
+    s->frame.pict_type           = FF_I_TYPE;
     s->frame.palette_has_changed = 1;
     memcpy(s->frame.data[1], s->palette, 16 * 4);
 
@@ -211,37 +211,40 @@ static av_cold int decode_end(AVCodecContext *avctx)
 }
 
 AVCodec ff_bintext_decoder = {
-    .name           = "bintext",
-    .type           = AVMEDIA_TYPE_VIDEO,
-    .id             = CODEC_ID_BINTEXT,
-    .priv_data_size = sizeof(XbinContext),
-    .init           = decode_init,
-    .close          = decode_end,
-    .decode         = decode_frame,
-    .capabilities   = CODEC_CAP_DR1,
+    "bintext",
+    AVMEDIA_TYPE_VIDEO,
+    CODEC_ID_BINTEXT,
+    sizeof(XbinContext),
+    decode_init,
+    NULL,
+    decode_end,
+    decode_frame,
+    CODEC_CAP_DR1,
     .long_name = NULL_IF_CONFIG_SMALL("Binary text"),
 };
 
 AVCodec ff_xbin_decoder = {
-    .name           = "xbin",
-    .type           = AVMEDIA_TYPE_VIDEO,
-    .id             = CODEC_ID_XBIN,
-    .priv_data_size = sizeof(XbinContext),
-    .init           = decode_init,
-    .close          = decode_end,
-    .decode         = decode_frame,
-    .capabilities   = CODEC_CAP_DR1,
+    "xbin",
+    AVMEDIA_TYPE_VIDEO,
+    CODEC_ID_XBIN,
+    sizeof(XbinContext),
+    decode_init,
+    NULL,
+    decode_end,
+    decode_frame,
+    CODEC_CAP_DR1,
     .long_name = NULL_IF_CONFIG_SMALL("eXtended BINary text"),
 };
 
 AVCodec ff_idf_decoder = {
-    .name           = "idf",
-    .type           = AVMEDIA_TYPE_VIDEO,
-    .id             = CODEC_ID_IDF,
-    .priv_data_size = sizeof(XbinContext),
-    .init           = decode_init,
-    .close          = decode_end,
-    .decode         = decode_frame,
-    .capabilities   = CODEC_CAP_DR1,
+    "idf",
+    AVMEDIA_TYPE_VIDEO,
+    CODEC_ID_IDF,
+    sizeof(XbinContext),
+    decode_init,
+    NULL,
+    decode_end,
+    decode_frame,
+    CODEC_CAP_DR1,
     .long_name = NULL_IF_CONFIG_SMALL("iCEDraw text"),
 };

libavcodec/bitstream.c

@@ -103,7 +103,7 @@ static int alloc_table(VLC *vlc, int size, int use_static)
     vlc->table_size += size;
     if (vlc->table_size > vlc->table_allocated) {
         if(use_static)
-            abort(); // cannot do anything, init_vlc() is used with too little memory
+            abort(); //cant do anything, init_vlc() is used with too little memory
         vlc->table_allocated += (1 << vlc->bits);
         vlc->table = av_realloc_f(vlc->table,
                                   vlc->table_allocated, sizeof(VLC_TYPE) * 2);

libavcodec/bmp.c

@@ -49,7 +49,6 @@ static int bmp_decode_frame(AVCodecContext *avctx,
     unsigned int ihsize;
     int i, j, n, linesize;
     uint32_t rgb[3];
-    uint32_t alpha = 0;
     uint8_t *ptr;
     int dsize;
     const uint8_t *buf0 = buf;
@@ -132,8 +131,6 @@ static int bmp_decode_frame(AVCodecContext *avctx,
         rgb[0] = bytestream_get_le32(&buf);
         rgb[1] = bytestream_get_le32(&buf);
         rgb[2] = bytestream_get_le32(&buf);
-        if (ihsize >= 108)
-            alpha = bytestream_get_le32(&buf);
     }
 
     avctx->width = width;
@@ -144,21 +141,21 @@ static int bmp_decode_frame(AVCodecContext *avctx,
     switch(depth){
     case 32:
         if(comp == BMP_BITFIELDS){
-            if (rgb[0] == 0xFF000000 && rgb[1] == 0x00FF0000 && rgb[2] == 0x0000FF00)
-                avctx->pix_fmt = alpha ? PIX_FMT_ABGR : PIX_FMT_0BGR;
-            else if (rgb[0] == 0x00FF0000 && rgb[1] == 0x0000FF00 && rgb[2] == 0x000000FF)
-                avctx->pix_fmt = alpha ? PIX_FMT_BGRA : PIX_FMT_BGR0;
-            else if (rgb[0] == 0x0000FF00 && rgb[1] == 0x00FF0000 && rgb[2] == 0xFF000000)
-                avctx->pix_fmt = alpha ? PIX_FMT_ARGB : PIX_FMT_0RGB;
-            else if (rgb[0] == 0x000000FF && rgb[1] == 0x0000FF00 && rgb[2] == 0x00FF0000)
-                avctx->pix_fmt = alpha ? PIX_FMT_RGBA : PIX_FMT_RGB0;
-            else {
-                av_log(avctx, AV_LOG_ERROR, "Unknown bitfields %0X %0X %0X\n", rgb[0], rgb[1], rgb[2]);
-                return AVERROR(EINVAL);
+            rgb[0] = (rgb[0] >> 15) & 3;
+            rgb[1] = (rgb[1] >> 15) & 3;
+            rgb[2] = (rgb[2] >> 15) & 3;
+
+            if(rgb[0] + rgb[1] + rgb[2] != 3 ||
+               rgb[0] == rgb[1] || rgb[0] == rgb[2] || rgb[1] == rgb[2]){
+                break;
             }
         } else {
-            avctx->pix_fmt = PIX_FMT_BGRA;
+            rgb[0] = 2;
+            rgb[1] = 1;
+            rgb[2] = 0;
         }
+
+        avctx->pix_fmt = PIX_FMT_BGRA;
         break;
     case 24:
         avctx->pix_fmt = PIX_FMT_BGR24;
@@ -166,18 +163,8 @@ static int bmp_decode_frame(AVCodecContext *avctx,
     case 16:
         if(comp == BMP_RGB)
             avctx->pix_fmt = PIX_FMT_RGB555;
-        else if (comp == BMP_BITFIELDS) {
-            if (rgb[0] == 0xF800 && rgb[1] == 0x07E0 && rgb[2] == 0x001F)
-               avctx->pix_fmt = PIX_FMT_RGB565;
-            else if (rgb[0] == 0x7C00 && rgb[1] == 0x03E0 && rgb[2] == 0x001F)
-               avctx->pix_fmt = PIX_FMT_RGB555;
-            else if (rgb[0] == 0x0F00 && rgb[1] == 0x00F0 && rgb[2] == 0x000F)
-               avctx->pix_fmt = PIX_FMT_RGB444;
-            else {
-               av_log(avctx, AV_LOG_ERROR, "Unknown bitfields %0X %0X %0X\n", rgb[0], rgb[1], rgb[2]);
-               return AVERROR(EINVAL);
-            }
-        }
+        if(comp == BMP_BITFIELDS)
+            avctx->pix_fmt = rgb[1] == 0x07E0 ? PIX_FMT_RGB565 : PIX_FMT_RGB555;
         break;
     case 8:
         if(hsize - ihsize - 14 > 0)
@@ -260,7 +247,7 @@ static int bmp_decode_frame(AVCodecContext *avctx,
                 ((uint32_t*)p->data[1])[i] = (0xff<<24) | bytestream_get_le24(&buf);
         }else{
             for(i = 0; i < colors; i++)
-                ((uint32_t*)p->data[1])[i] = 0xFFU << 24 | bytestream_get_le32(&buf);
+                ((uint32_t*)p->data[1])[i] = bytestream_get_le32(&buf);
         }
         buf = buf0 + hsize;
     }
@@ -295,7 +282,6 @@ static int bmp_decode_frame(AVCodecContext *avctx,
             break;
         case 8:
         case 24:
-        case 32:
             for(i = 0; i < avctx->height; i++){
                 memcpy(ptr, buf, n);
                 buf += n;
@@ -325,6 +311,29 @@ static int bmp_decode_frame(AVCodecContext *avctx,
                 ptr += linesize;
             }
             break;
+        case 32:
+            for(i = 0; i < avctx->height; i++){
+                const uint8_t *src = buf;
+                uint8_t *dst = ptr;
+
+                for(j = 0; j < avctx->width; j++){
+                    dst[0] = src[rgb[2]];
+                    dst[1] = src[rgb[1]];
+                    dst[2] = src[rgb[0]];
+/* The Microsoft documentation states:
+ * "The high byte in each DWORD is not used."
+ * Both GIMP and ImageMagick store the alpha transparency value
+ * in the high byte for 32bit bmp files.
+ */
+                    dst[3] = src[3];
+                    dst += 4;
+                    src += 4;
+                }
+
+                buf += n;
+                ptr += linesize;
+            }
+            break;
         default:
             av_log(avctx, AV_LOG_ERROR, "BMP decoder is broken\n");
             return -1;

libavcodec/bmpenc.c

@@ -24,11 +24,9 @@
 #include "avcodec.h"
 #include "bytestream.h"
 #include "bmp.h"
-#include <assert.h>
 
 static const uint32_t monoblack_pal[] = { 0x000000, 0xFFFFFF };
 static const uint32_t rgb565_masks[]  = { 0xF800, 0x07E0, 0x001F };
-static const uint32_t rgb444_masks[]  = { 0x0F00, 0x00F0, 0x000F };
 
 static av_cold int bmp_encode_init(AVCodecContext *avctx){
     BMPContext *s = avctx->priv_data;
@@ -37,15 +35,13 @@ static av_cold int bmp_encode_init(AVCodecContext *avctx){
     avctx->coded_frame = (AVFrame*)&s->picture;
 
     switch (avctx->pix_fmt) {
-    case PIX_FMT_BGRA:
-        avctx->bits_per_coded_sample = 32;
-        break;
     case PIX_FMT_BGR24:
         avctx->bits_per_coded_sample = 24;
         break;
     case PIX_FMT_RGB555:
+        avctx->bits_per_coded_sample = 16;
+        break;
     case PIX_FMT_RGB565:
-    case PIX_FMT_RGB444:
         avctx->bits_per_coded_sample = 16;
         break;
     case PIX_FMT_RGB8:
@@ -73,7 +69,6 @@ static int bmp_encode_frame(AVCodecContext *avctx, unsigned char *buf, int buf_s
     AVFrame * const p= (AVFrame*)&s->picture;
     int n_bytes_image, n_bytes_per_row, n_bytes, i, n, hsize;
     const uint32_t *pal = NULL;
-    uint32_t palette256[256];
     int pad_bytes_per_row, pal_entries = 0, compression = BMP_RGB;
     int bit_count = avctx->bits_per_coded_sample;
     uint8_t *ptr;
@@ -82,11 +77,6 @@ static int bmp_encode_frame(AVCodecContext *avctx, unsigned char *buf, int buf_s
     p->pict_type= AV_PICTURE_TYPE_I;
     p->key_frame= 1;
     switch (avctx->pix_fmt) {
-    case PIX_FMT_RGB444:
-        compression = BMP_BITFIELDS;
-        pal = rgb444_masks; // abuse pal to hold color masks
-        pal_entries = 3;
-        break;
     case PIX_FMT_RGB565:
         compression = BMP_BITFIELDS;
         pal = rgb565_masks; // abuse pal to hold color masks
@@ -97,10 +87,7 @@ static int bmp_encode_frame(AVCodecContext *avctx, unsigned char *buf, int buf_s
     case PIX_FMT_RGB4_BYTE:
     case PIX_FMT_BGR4_BYTE:
     case PIX_FMT_GRAY8:
-        assert(bit_count == 8);
-        ff_set_systematic_pal2(palette256, avctx->pix_fmt);
-        pal = palette256;
-        break;
+        ff_set_systematic_pal2((uint32_t*)p->data[1], avctx->pix_fmt);
     case PIX_FMT_PAL8:
         pal = (uint32_t *)p->data[1];
         break;
@@ -170,8 +157,8 @@ AVCodec ff_bmp_encoder = {
     .init           = bmp_encode_init,
     .encode         = bmp_encode_frame,
     .pix_fmts = (const enum PixelFormat[]){
-        PIX_FMT_BGRA, PIX_FMT_BGR24,
-        PIX_FMT_RGB565, PIX_FMT_RGB555, PIX_FMT_RGB444,
+        PIX_FMT_BGR24,
+        PIX_FMT_RGB555, PIX_FMT_RGB565,
         PIX_FMT_RGB8, PIX_FMT_BGR8, PIX_FMT_RGB4_BYTE, PIX_FMT_BGR4_BYTE, PIX_FMT_GRAY8, PIX_FMT_PAL8,
         PIX_FMT_MONOBLACK,
         PIX_FMT_NONE},

libavcodec/bytestream.h

@@ -27,7 +27,7 @@
 #include "libavutil/intreadwrite.h"
 
 typedef struct {
-    const uint8_t *buffer, *buffer_end, *buffer_start;
+    const uint8_t *buffer, *buffer_end;
 } GetByteContext;
 
 #define DEF_T(type, name, bytes, read, write)                             \
@@ -39,15 +39,11 @@ static av_always_inline void bytestream_put_ ##name(uint8_t **b, const type valu
     write(*b, value);\
     (*b) += bytes;\
 }\
-static av_always_inline type bytestream2_get_ ## name ## u(GetByteContext *g)\
-{\
-    return bytestream_get_ ## name(&g->buffer);\
-}\
 static av_always_inline type bytestream2_get_ ## name(GetByteContext *g)\
 {\
     if (g->buffer_end - g->buffer < bytes)\
         return 0;\
-    return bytestream2_get_ ## name ## u(g);\
+    return bytestream_get_ ## name(&g->buffer);\
 }\
 static av_always_inline type bytestream2_peek_ ## name(GetByteContext *g)\
 {\
@@ -75,47 +71,10 @@ DEF  (byte, 1, AV_RB8 , AV_WB8 )
 #undef DEF64
 #undef DEF_T
 
-#if HAVE_BIGENDIAN
-#   define bytestream2_get_ne16  bytestream2_get_be16
-#   define bytestream2_get_ne24  bytestream2_get_be24
-#   define bytestream2_get_ne32  bytestream2_get_be32
-#   define bytestream2_get_ne64  bytestream2_get_be64
-#   define bytestream2_get_ne16u bytestream2_get_be16u
-#   define bytestream2_get_ne24u bytestream2_get_be24u
-#   define bytestream2_get_ne32u bytestream2_get_be32u
-#   define bytestream2_get_ne64u bytestream2_get_be64u
-#   define bytestream2_put_ne16  bytestream2_put_be16
-#   define bytestream2_put_ne24  bytestream2_put_be24
-#   define bytestream2_put_ne32  bytestream2_put_be32
-#   define bytestream2_put_ne64  bytestream2_put_be64
-#   define bytestream2_peek_ne16 bytestream2_peek_be16
-#   define bytestream2_peek_ne24 bytestream2_peek_be24
-#   define bytestream2_peek_ne32 bytestream2_peek_be32
-#   define bytestream2_peek_ne64 bytestream2_peek_be64
-#else
-#   define bytestream2_get_ne16  bytestream2_get_le16
-#   define bytestream2_get_ne24  bytestream2_get_le24
-#   define bytestream2_get_ne32  bytestream2_get_le32
-#   define bytestream2_get_ne64  bytestream2_get_le64
-#   define bytestream2_get_ne16u bytestream2_get_le16u
-#   define bytestream2_get_ne24u bytestream2_get_le24u
-#   define bytestream2_get_ne32u bytestream2_get_le32u
-#   define bytestream2_get_ne64u bytestream2_get_le64u
-#   define bytestream2_put_ne16  bytestream2_put_le16
-#   define bytestream2_put_ne24  bytestream2_put_le24
-#   define bytestream2_put_ne32  bytestream2_put_le32
-#   define bytestream2_put_ne64  bytestream2_put_le64
-#   define bytestream2_peek_ne16 bytestream2_peek_le16
-#   define bytestream2_peek_ne24 bytestream2_peek_le24
-#   define bytestream2_peek_ne32 bytestream2_peek_le32
-#   define bytestream2_peek_ne64 bytestream2_peek_le64
-#endif
-
 static av_always_inline void bytestream2_init(GetByteContext *g,
                                               const uint8_t *buf, int buf_size)
 {
     g->buffer =  buf;
-    g->buffer_start = buf;
     g->buffer_end = buf + buf_size;
 }
 
@@ -130,34 +89,6 @@ static av_always_inline void bytestream2_skip(GetByteContext *g,
     g->buffer += FFMIN(g->buffer_end - g->buffer, size);
 }
 
-static av_always_inline int bytestream2_tell(GetByteContext *g)
-{
-    return (int)(g->buffer - g->buffer_start);
-}
-
-static av_always_inline int bytestream2_seek(GetByteContext *g, int offset,
-                                             int whence)
-{
-    switch (whence) {
-    case SEEK_CUR:
-        offset = av_clip(offset, -(g->buffer - g->buffer_start),
-                         g->buffer_end - g->buffer);
-        g->buffer += offset;
-        break;
-    case SEEK_END:
-        offset = av_clip(offset, -(g->buffer_end - g->buffer_start), 0);
-        g->buffer = g->buffer_end + offset;
-        break;
-    case SEEK_SET:
-        offset = av_clip(offset, 0, g->buffer_end - g->buffer_start);
-        g->buffer = g->buffer_start + offset;
-        break;
-    default:
-        return AVERROR(EINVAL);
-    }
-    return bytestream2_tell(g);
-}
-
 static av_always_inline unsigned int bytestream2_get_buffer(GetByteContext *g,
                                                             uint8_t *dst,
                                                             unsigned int size)

libavcodec/cabac.c

@@ -29,7 +29,6 @@
 #include "libavutil/common.h"
 #include "get_bits.h"
 #include "cabac.h"
-#include "cabac_functions.h"
 
 static const uint8_t lps_range[64][4]= {
 {128,176,208,240}, {128,167,197,227}, {128,158,187,216}, {123,150,178,205},
@@ -53,7 +52,7 @@ static const uint8_t lps_range[64][4]= {
 uint8_t ff_h264_mlps_state[4*64];
 uint8_t ff_h264_lps_range[4*2*64];
 uint8_t ff_h264_lps_state[2*64];
-static uint8_t h264_mps_state[2 * 64];
+uint8_t ff_h264_mps_state[2*64];
 
 static const uint8_t mps_state[64]= {
   1, 2, 3, 4, 5, 6, 7, 8,
@@ -110,6 +109,10 @@ void ff_init_cabac_encoder(CABACContext *c, uint8_t *buf, int buf_size){
     c->low= 0;
     c->range= 0x1FE;
     c->outstanding_count= 0;
+#ifdef STRICT_LIMITS
+    c->sym_count =0;
+#endif
+
     c->pb.bit_left++; //avoids firstBitFlag
 }
 
@@ -142,9 +145,9 @@ void ff_init_cabac_states(CABACContext *c){
         }
 
         ff_h264_mlps_state[128+2*i+0]=
-        h264_mps_state[2 * i + 0] = 2 * mps_state[i] + 0;
+        ff_h264_mps_state[2*i+0]= 2*mps_state[i]+0;
         ff_h264_mlps_state[128+2*i+1]=
-        h264_mps_state[2 * i + 1] = 2 * mps_state[i] + 1;
+        ff_h264_mps_state[2*i+1]= 2*mps_state[i]+1;
 
         if( i ){
             ff_h264_lps_state[2*i+0]=
@@ -167,37 +170,12 @@ void ff_init_cabac_states(CABACContext *c){
 #include "avcodec.h"
 #include "cabac.h"
 
-static inline void put_cabac_bit(CABACContext *c, int b){
-    put_bits(&c->pb, 1, b);
-    for(;c->outstanding_count; c->outstanding_count--){
-        put_bits(&c->pb, 1, 1-b);
-    }
-}
-
-static inline void renorm_cabac_encoder(CABACContext *c){
-    while(c->range < 0x100){
-        //FIXME optimize
-        if(c->low<0x100){
-            put_cabac_bit(c, 0);
-        }else if(c->low<0x200){
-            c->outstanding_count++;
-            c->low -= 0x100;
-        }else{
-            put_cabac_bit(c, 1);
-            c->low -= 0x200;
-        }
-
-        c->range+= c->range;
-        c->low += c->low;
-    }
-}
-
 static void put_cabac(CABACContext *c, uint8_t * const state, int bit){
     int RangeLPS= ff_h264_lps_range[2*(c->range&0xC0) + *state];
 
     if(bit == ((*state)&1)){
         c->range -= RangeLPS;
-        *state    = h264_mps_state[*state];
+        *state= ff_h264_mps_state[*state];
     }else{
         c->low += c->range - RangeLPS;
         c->range = RangeLPS;
@@ -205,6 +183,10 @@ static void put_cabac(CABACContext *c, uint8_t * const state, int bit){
     }
 
     renorm_cabac_encoder(c);
+
+#ifdef STRICT_LIMITS
+    c->symCount++;
+#endif
 }
 
 /**
@@ -226,6 +208,10 @@ static void put_cabac_bypass(CABACContext *c, int bit){
         put_cabac_bit(c, 1);
         c->low -= 0x400;
     }
+
+#ifdef STRICT_LIMITS
+    c->symCount++;
+#endif
 }
 
 /**
@@ -250,9 +236,74 @@ static int put_cabac_terminate(CABACContext *c, int bit){
         flush_put_bits(&c->pb); //FIXME FIXME FIXME XXX wrong
     }
 
+#ifdef STRICT_LIMITS
+    c->symCount++;
+#endif
+
     return (put_bits_count(&c->pb)+7)>>3;
 }
 
+/**
+ * put (truncated) unary binarization.
+ */
+static void put_cabac_u(CABACContext *c, uint8_t * state, int v, int max, int max_index, int truncated){
+    int i;
+
+    assert(v <= max);
+
+    for(i=0; i<v; i++){
+        put_cabac(c, state, 1);
+        if(i < max_index) state++;
+    }
+    if(truncated==0 || v<max)
+        put_cabac(c, state, 0);
+}
+
+/**
+ * put unary exp golomb k-th order binarization.
+ */
+static void put_cabac_ueg(CABACContext *c, uint8_t * state, int v, int max, int is_signed, int k, int max_index){
+    int i;
+
+    if(v==0)
+        put_cabac(c, state, 0);
+    else{
+        const int sign= v < 0;
+
+        if(is_signed) v= FFABS(v);
+
+        if(v<max){
+            for(i=0; i<v; i++){
+                put_cabac(c, state, 1);
+                if(i < max_index) state++;
+            }
+
+            put_cabac(c, state, 0);
+        }else{
+            int m= 1<<k;
+
+            for(i=0; i<max; i++){
+                put_cabac(c, state, 1);
+                if(i < max_index) state++;
+            }
+
+            v -= max;
+            while(v >= m){ //FIXME optimize
+                put_cabac_bypass(c, 1);
+                v-= m;
+                m+= m;
+            }
+            put_cabac_bypass(c, 0);
+            while(m>>=1){
+                put_cabac_bypass(c, v&m);
+            }
+        }
+
+        if(is_signed)
+            put_cabac_bypass(c, sign);
+    }
+}
+
 int main(void){
     CABACContext c;
     uint8_t b[9*SIZE];
@@ -282,6 +333,19 @@ START_TIMER
 STOP_TIMER("put_cabac")
     }
 
+#if 0
+    for(i=0; i<SIZE; i++){
+START_TIMER
+        put_cabac_u(&c, state, r[i], 6, 3, i&1);
+STOP_TIMER("put_cabac_u")
+    }
+
+    for(i=0; i<SIZE; i++){
+START_TIMER
+        put_cabac_ueg(&c, state, r[i], 3, 0, 1, 2);
+STOP_TIMER("put_cabac_ueg")
+    }
+#endif
     put_cabac_terminate(&c, 1);
 
     ff_init_cabac_decoder(&c, b, SIZE);
@@ -301,6 +365,21 @@ START_TIMER
             av_log(NULL, AV_LOG_ERROR, "CABAC failure at %d\n", i);
 STOP_TIMER("get_cabac")
     }
+#if 0
+    for(i=0; i<SIZE; i++){
+START_TIMER
+        if( r[i] != get_cabac_u(&c, state, (i&1) ? 6 : 7, 3, i&1) )
+            av_log(NULL, AV_LOG_ERROR, "CABAC unary (truncated) binarization failure at %d\n", i);
+STOP_TIMER("get_cabac_u")
+    }
+
+    for(i=0; i<SIZE; i++){
+START_TIMER
+        if( r[i] != get_cabac_ueg(&c, state, 3, 0, 1, 2))
+            av_log(NULL, AV_LOG_ERROR, "CABAC unary (truncated) binarization failure at %d\n", i);
+STOP_TIMER("get_cabac_ueg")
+    }
+#endif
     if(!get_cabac_terminate(&c))
         av_log(NULL, AV_LOG_ERROR, "where's the Terminator?\n");
 

libavcodec/cabac.h

@@ -27,10 +27,13 @@
 #ifndef AVCODEC_CABAC_H
 #define AVCODEC_CABAC_H
 
-#include <stdint.h>
+#include <stddef.h>
 
 #include "put_bits.h"
 
+//#undef NDEBUG
+#include <assert.h>
+
 #define CABAC_BITS 16
 #define CABAC_MASK ((1<<CABAC_BITS)-1)
 
@@ -38,14 +41,237 @@ typedef struct CABACContext{
     int low;
     int range;
     int outstanding_count;
+#ifdef STRICT_LIMITS
+    int symCount;
+#endif
     const uint8_t *bytestream_start;
     const uint8_t *bytestream;
     const uint8_t *bytestream_end;
     PutBitContext pb;
 }CABACContext;
 
+extern uint8_t ff_h264_mlps_state[4*64];
+extern uint8_t ff_h264_lps_range[4*2*64];  ///< rangeTabLPS
+extern uint8_t ff_h264_mps_state[2*64];     ///< transIdxMPS
+extern uint8_t ff_h264_lps_state[2*64];     ///< transIdxLPS
+extern const uint8_t ff_h264_norm_shift[512];
+
+#if ARCH_X86
+#   include "x86/cabac.h"
+#endif
+
 void ff_init_cabac_encoder(CABACContext *c, uint8_t *buf, int buf_size);
 void ff_init_cabac_decoder(CABACContext *c, const uint8_t *buf, int buf_size);
 void ff_init_cabac_states(CABACContext *c);
 
+
+static inline void put_cabac_bit(CABACContext *c, int b){
+    put_bits(&c->pb, 1, b);
+    for(;c->outstanding_count; c->outstanding_count--){
+        put_bits(&c->pb, 1, 1-b);
+    }
+}
+
+static inline void renorm_cabac_encoder(CABACContext *c){
+    while(c->range < 0x100){
+        //FIXME optimize
+        if(c->low<0x100){
+            put_cabac_bit(c, 0);
+        }else if(c->low<0x200){
+            c->outstanding_count++;
+            c->low -= 0x100;
+        }else{
+            put_cabac_bit(c, 1);
+            c->low -= 0x200;
+        }
+
+        c->range+= c->range;
+        c->low += c->low;
+    }
+}
+
+static void refill(CABACContext *c){
+#if CABAC_BITS == 16
+        c->low+= (c->bytestream[0]<<9) + (c->bytestream[1]<<1);
+#else
+        c->low+= c->bytestream[0]<<1;
+#endif
+    c->low -= CABAC_MASK;
+    c->bytestream+= CABAC_BITS/8;
+}
+
+static inline void renorm_cabac_decoder(CABACContext *c){
+    while(c->range < 0x100){
+        c->range+= c->range;
+        c->low+= c->low;
+        if(!(c->low & CABAC_MASK))
+            refill(c);
+    }
+}
+
+static inline void renorm_cabac_decoder_once(CABACContext *c){
+    int shift= (uint32_t)(c->range - 0x100)>>31;
+    c->range<<= shift;
+    c->low  <<= shift;
+    if(!(c->low & CABAC_MASK))
+        refill(c);
+}
+
+#ifndef get_cabac_inline
+static void refill2(CABACContext *c){
+    int i, x;
+
+    x= c->low ^ (c->low-1);
+    i= 7 - ff_h264_norm_shift[x>>(CABAC_BITS-1)];
+
+    x= -CABAC_MASK;
+
+#if CABAC_BITS == 16
+        x+= (c->bytestream[0]<<9) + (c->bytestream[1]<<1);
+#else
+        x+= c->bytestream[0]<<1;
+#endif
+
+    c->low += x<<i;
+    c->bytestream+= CABAC_BITS/8;
+}
+
+static av_always_inline int get_cabac_inline(CABACContext *c, uint8_t * const state){
+    int s = *state;
+    int RangeLPS= ff_h264_lps_range[2*(c->range&0xC0) + s];
+    int bit, lps_mask;
+
+    c->range -= RangeLPS;
+    lps_mask= ((c->range<<(CABAC_BITS+1)) - c->low)>>31;
+
+    c->low -= (c->range<<(CABAC_BITS+1)) & lps_mask;
+    c->range += (RangeLPS - c->range) & lps_mask;
+
+    s^=lps_mask;
+    *state= (ff_h264_mlps_state+128)[s];
+    bit= s&1;
+
+    lps_mask= ff_h264_norm_shift[c->range];
+    c->range<<= lps_mask;
+    c->low  <<= lps_mask;
+    if(!(c->low & CABAC_MASK))
+        refill2(c);
+    return bit;
+}
+#endif
+
+static int av_noinline av_unused get_cabac_noinline(CABACContext *c, uint8_t * const state){
+    return get_cabac_inline(c,state);
+}
+
+static int av_unused get_cabac(CABACContext *c, uint8_t * const state){
+    return get_cabac_inline(c,state);
+}
+
+static int av_unused get_cabac_bypass(CABACContext *c){
+    int range;
+    c->low += c->low;
+
+    if(!(c->low & CABAC_MASK))
+        refill(c);
+
+    range= c->range<<(CABAC_BITS+1);
+    if(c->low < range){
+        return 0;
+    }else{
+        c->low -= range;
+        return 1;
+    }
+}
+
+
+#ifndef get_cabac_bypass_sign
+static av_always_inline int get_cabac_bypass_sign(CABACContext *c, int val){
+    int range, mask;
+    c->low += c->low;
+
+    if(!(c->low & CABAC_MASK))
+        refill(c);
+
+    range= c->range<<(CABAC_BITS+1);
+    c->low -= range;
+    mask= c->low >> 31;
+    range &= mask;
+    c->low += range;
+    return (val^mask)-mask;
+}
+#endif
+
+/**
+ *
+ * @return the number of bytes read or 0 if no end
+ */
+static int av_unused get_cabac_terminate(CABACContext *c){
+    c->range -= 2;
+    if(c->low < c->range<<(CABAC_BITS+1)){
+        renorm_cabac_decoder_once(c);
+        return 0;
+    }else{
+        return c->bytestream - c->bytestream_start;
+    }
+}
+
+#if 0
+/**
+ * Get (truncated) unary binarization.
+ */
+static int get_cabac_u(CABACContext *c, uint8_t * state, int max, int max_index, int truncated){
+    int i;
+
+    for(i=0; i<max; i++){
+        if(get_cabac(c, state)==0)
+            return i;
+
+        if(i< max_index) state++;
+    }
+
+    return truncated ? max : -1;
+}
+
+/**
+ * get unary exp golomb k-th order binarization.
+ */
+static int get_cabac_ueg(CABACContext *c, uint8_t * state, int max, int is_signed, int k, int max_index){
+    int i, v;
+    int m= 1<<k;
+
+    if(get_cabac(c, state)==0)
+        return 0;
+
+    if(0 < max_index) state++;
+
+    for(i=1; i<max; i++){
+        if(get_cabac(c, state)==0){
+            if(is_signed && get_cabac_bypass(c)){
+                return -i;
+            }else
+                return i;
+        }
+
+        if(i < max_index) state++;
+    }
+
+    while(get_cabac_bypass(c)){
+        i+= m;
+        m+= m;
+    }
+
+    v=0;
+    while(m>>=1){
+        v+= v + get_cabac_bypass(c);
+    }
+    i += v;
+
+    if(is_signed && get_cabac_bypass(c)){
+        return -i;
+    }else
+        return i;
+}
+#endif /* 0 */
+
 #endif /* AVCODEC_CABAC_H */