Update for 2.2.3

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

.gitignore

@@ -27,7 +27,6 @@
 /ffserver
 /config.*
 /coverage.info
-/version.h
 /doc/*.1
 /doc/*.3
 /doc/*.html
@@ -35,23 +34,28 @@
 /doc/config.texi
 /doc/avoptions_codec.texi
 /doc/avoptions_format.texi
-/doc/examples/decoding_encoding
-/doc/examples/demuxing
+/doc/doxy/html/
+/doc/examples/avio_reading
+/doc/examples/avcodec
+/doc/examples/demuxing_decoding
+/doc/examples/filter_audio
 /doc/examples/filtering_audio
 /doc/examples/filtering_video
 /doc/examples/metadata
 /doc/examples/muxing
 /doc/examples/pc-uninstalled
+/doc/examples/remuxing
 /doc/examples/resampling_audio
 /doc/examples/scaling_video
+/doc/examples/transcode_aac
 /doc/fate.txt
-/doc/doxy/html/
 /doc/print_options
 /lcov/
 /libavcodec/*_tablegen
 /libavcodec/*_tables.c
 /libavcodec/*_tables.h
 /libavutil/avconfig.h
+/libavutil/ffversion.h
 /tests/audiogen
 /tests/base64
 /tests/data/

Changelog

@@ -1,7 +1,34 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.
 
-version <next>
+version 2.2:
+
+- HNM version 4 demuxer and video decoder
+- Live HDS muxer
+- setsar/setdar filters now support variables in ratio expressions
+- elbg filter
+- string validation in ffprobe
+- support for decoding through VDPAU in ffmpeg (the -hwaccel option)
+- complete Voxware MetaSound decoder
+- remove mp3_header_compress bitstream filter
+- Windows resource files for shared libraries
+- aeval filter
+- stereoscopic 3d metadata handling
+- WebP encoding via libwebp
+- ATRAC3+ decoder
+- VP8 in Ogg demuxing
+- side & metadata support in NUT
+- framepack filter
+- XYZ12 rawvideo support in NUT
+- Exif metadata support in WebP decoder
+- OpenGL device
+- Use metadata_header_padding to control padding in ID3 tags (currently used in
+  MP3, AIFF, and OMA files), FLAC header, and the AVI "junk" block.
+- Mirillis FIC video decoder
+- Support DNx444
+- libx265 encoder
+- dejudder filter
+- Autodetect VDA like all other hardware accelerations
 
 
 version 2.1:
@@ -46,7 +73,8 @@ version 2.1:
 - ReplayGain scanner
 - Enhanced Low Delay AAC (ER AAC ELD) decoding (no LD SBR support)
 - Linux framebuffer output device
-- HEVC decoder, raw HEVC demuxer, HEVC demuxing in TS, Matroska and MP4
+- HEVC decoder
+- raw HEVC, HEVC in MOV/MP4, HEVC in Matroska, HEVC in MPEG-TS demuxing
 - mergeplanes filter
 
 
@@ -62,7 +90,7 @@ version 2.0:
 - 10% faster aac encoding on x86 and MIPS
 - sine audio filter source
 - WebP demuxing and decoding support
-- new ffmpeg options -filter_script and -filter_complex_script, which allow a
+- ffmpeg options -filter_script and -filter_complex_script, which allow a
   filtergraph description to be read from a file
 - OpenCL support
 - audio phaser filter
@@ -70,7 +98,7 @@ version 2.0:
 - libquvi demuxer
 - uniform options syntax across all filters
 - telecine filter
-- new interlace filter
+- interlace filter
 - smptehdbars source
 - inverse telecine filters (fieldmatch and decimate)
 - colorbalance filter

LICENSE

@@ -47,7 +47,6 @@ Specifically, the GPL parts of FFmpeg are
     - vf_stereo3d.c
     - vf_super2xsai.c
     - vf_tinterlace.c
-    - vf_yadif.c
     - vsrc_mptestsrc.c
 
 Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
@@ -80,6 +79,7 @@ The following libraries are under GPL:
     - libutvideo
     - libvidstab
     - libx264
+    - libx265
     - libxavs
     - libxvid
 When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by

MAINTAINERS

@@ -31,7 +31,7 @@ ffprobe:
   ffprobe.c                             Stefano Sabatini
 
 ffserver:
-  ffserver.c, ffserver.h                Baptiste Coudurier
+  ffserver.c                            Reynaldo H. Verdejo Pinochet
 
 Commandline utility code:
   cmdutils.c, cmdutils.h                Michael Niedermayer
@@ -142,6 +142,7 @@ Codecs:
   ass*                                  Aurelien Jacobs
   asv*                                  Michael Niedermayer
   atrac3*                               Benjamin Larsson
+  atrac3plus*                           Maxim Poliakovski
   bgmc.c, bgmc.h                        Thilo Borgmann
   bink.c                                Kostya Shishkov
   binkaudio.c                           Peter Ross
@@ -150,6 +151,7 @@ Codecs:
   cdxl.c                                Paul B Mahol
   celp_filters.*                        Vitor Sessak
   cinepak.c                             Roberto Togni
+  cinepakenc.c                          Rl / Aetey G.T. AB
   cljr                                  Alex Beregszaszi
   cllc.c                                Derek Buitenhuis
   cook.c, cookdata.h                    Benjamin Larsson
@@ -197,8 +199,11 @@ Codecs:
   libtheoraenc.c                        David Conrad
   libutvideo*                           Derek Buitenhuis
   libvorbis.c                           David Conrad
+  libvpx*                               James Zern
   libx264.c                             Mans Rullgard, Jason Garrett-Glaser
+  libx265.c                             Derek Buitenhuis
   libxavs.c                             Stefan Gehrer
+  libzvbi-teletextdec.c                 Marton Balint
   loco.c                                Kostya Shishkov
   lzo.h, lzo.c                          Reimar Doeffinger
   mdec.c                                Michael Niedermayer
@@ -266,6 +271,7 @@ Codecs:
   vp5                                   Aurelien Jacobs
   vp6                                   Aurelien Jacobs
   vp8                                   David Conrad, Jason Garrett-Glaser, Ronald Bultje
+  vp9                                   Ronald Bultje, Clément Bœsch
   vqavideo.c                            Mike Melanson
   wavpack.c                             Kostya Shishkov
   wmaprodec.c                           Sascha Sommer
@@ -301,6 +307,7 @@ libavdevice
   iec61883.c                            Georg Lippitsch
   lavfi                                 Stefano Sabatini
   libdc1394.c                           Roman Shaposhnik
+  opengl_enc.c                          Lukasz Marek
   pulse_audio_enc.c                     Lukasz Marek
   sdl                                   Stefano Sabatini
   v4l2.c                                Luca Abeni
@@ -329,6 +336,7 @@ Filters:
   avf_avectorscope.c                    Paul B Mahol
   vf_blend.c                            Paul B Mahol
   vf_colorbalance.c                     Paul B Mahol
+  vf_dejudder.c                         Nicholas Robbins
   vf_delogo.c                           Jean Delvare (CC <khali@linux-fr.org>)
   vf_drawbox.c/drawgrid                 Andrey Utkin
   vf_extractplanes.c                    Paul B Mahol
@@ -385,6 +393,7 @@ Muxers/Demuxers:
   flvdec.c, flvenc.c                    Michael Niedermayer
   gxf.c                                 Reimar Doeffinger
   gxfenc.c                              Baptiste Coudurier
+  hls.c                                 Anssi Hannula
   idcin.c                               Mike Melanson
   idroqdec.c                            Mike Melanson
   iff.c                                 Jaikrishnan Menon
@@ -444,6 +453,7 @@ Muxers/Demuxers:
   siff.c                                Kostya Shishkov
   smacker.c                             Kostya Shishkov
   smjpeg*                               Paul B Mahol
+  spdif*                                Anssi Hannula
   srtdec.c                              Aurelien Jacobs
   swf.c                                 Baptiste Coudurier
   takdec.c                              Paul B Mahol
@@ -501,8 +511,9 @@ x86                                     Michael Niedermayer
 Releases
 ========
 
+2.2                                     Michael Niedermayer
 2.1                                     Michael Niedermayer
-2.0                                     Michael Niedermayer
+1.2                                     Michael Niedermayer
 
 If you want to maintain an older release, please contact us
 

Makefile

@@ -6,29 +6,34 @@ vpath %.cpp  $(SRC_PATH)
 vpath %.h    $(SRC_PATH)
 vpath %.S    $(SRC_PATH)
 vpath %.asm  $(SRC_PATH)
+vpath %.rc   $(SRC_PATH)
 vpath %.v    $(SRC_PATH)
 vpath %.texi $(SRC_PATH)
 vpath %/fate_config.sh.template $(SRC_PATH)
 
-PROGS-$(CONFIG_FFMPEG)   += ffmpeg
-PROGS-$(CONFIG_FFPLAY)   += ffplay
-PROGS-$(CONFIG_FFPROBE)  += ffprobe
-PROGS-$(CONFIG_FFSERVER) += ffserver
+AVPROGS-$(CONFIG_FFMPEG)   += ffmpeg
+AVPROGS-$(CONFIG_FFPLAY)   += ffplay
+AVPROGS-$(CONFIG_FFPROBE)  += ffprobe
+AVPROGS-$(CONFIG_FFSERVER) += ffserver
 
-PROGS      := $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
-INSTPROGS   = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+AVPROGS    := $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+INSTPROGS   = $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+PROGS      += $(AVPROGS)
 
-OBJS        = cmdutils.o $(EXEOBJS)
-OBJS-ffmpeg = ffmpeg_opt.o ffmpeg_filter.o
+AVBASENAMES  = ffmpeg ffplay ffprobe ffserver
+ALLAVPROGS   = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF))
+ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
+
+$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog) += cmdutils.o))
+$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog)-$(CONFIG_OPENCL) += cmdutils_opencl.o))
+
+OBJS-ffmpeg                   += ffmpeg_opt.o ffmpeg_filter.o
+OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o
 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
 HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
-TOOLS       = qt-faststart trasher
+TOOLS       = qt-faststart trasher uncoded_frame
 TOOLS-$(CONFIG_ZLIB) += cws2fws
 
-BASENAMES   = ffmpeg ffplay ffprobe ffserver
-ALLPROGS    = $(BASENAMES:%=%$(PROGSSUF)$(EXESUF))
-ALLPROGS_G  = $(BASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
-
 FFLIBS-$(CONFIG_AVDEVICE) += avdevice
 FFLIBS-$(CONFIG_AVFILTER) += avfilter
 FFLIBS-$(CONFIG_AVFORMAT) += avformat
@@ -50,16 +55,14 @@ include $(SRC_PATH)/common.mak
 FF_EXTRALIBS := $(FFEXTRALIBS)
 FF_DEP_LIBS  := $(DEP_LIBS)
 
-all: $(PROGS)
-
-$(PROGS): %$(EXESUF): %_g$(EXESUF)
-	$(CP) $< $@
-	$(STRIP) $@
+all: $(AVPROGS)
 
 $(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
 	$(LD) $(LDFLAGS) $(LD_O) $^ $(ELIBS)
 
 tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
+tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
+tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
 
 config.h: .config
 .config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
@@ -73,7 +76,7 @@ SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS      \
                ALTIVEC-OBJS VIS-OBJS                                     \
                MMX-OBJS YASM-OBJS                                        \
                MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MIPS32R2-OBJS  \
-               OBJS HOSTOBJS TESTOBJS
+               OBJS SLIBOBJS HOSTOBJS TESTOBJS
 
 define RESET
 $(1) :=
@@ -90,8 +93,10 @@ endef
 
 $(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))
 
+include $(SRC_PATH)/doc/Makefile
+
 define DOPROG
-OBJS-$(1) += $(1).o cmdutils.o $(EXEOBJS)
+OBJS-$(1) += $(1).o $(EXEOBJS) $(OBJS-$(1)-yes)
 $(1)$(PROGSSUF)_g$(EXESUF): $$(OBJS-$(1))
 $$(OBJS-$(1)): CFLAGS  += $(CFLAGS-$(1))
 $(1)$(PROGSSUF)_g$(EXESUF): LDFLAGS += $(LDFLAGS-$(1))
@@ -99,7 +104,13 @@ $(1)$(PROGSSUF)_g$(EXESUF): FF_EXTRALIBS += $(LIBS-$(1))
 -include $$(OBJS-$(1):.o=.d)
 endef
 
-$(foreach P,$(PROGS-yes),$(eval $(call DOPROG,$(P))))
+$(foreach P,$(PROGS),$(eval $(call DOPROG,$(P:$(PROGSSUF)$(EXESUF)=))))
+
+ffprobe.o cmdutils.o : libavutil/ffversion.h
+
+$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
+	$(CP) $< $@
+	$(STRIP) $@
 
 %$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
 	$(LD) $(LDFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
@@ -114,14 +125,14 @@ GIT_LOG     = $(SRC_PATH)/.git/logs/HEAD
 .version: $(wildcard $(GIT_LOG)) $(VERSION_SH) config.mak
 .version: M=@
 
-version.h .version:
-	$(M)$(VERSION_SH) $(SRC_PATH) version.h $(EXTRA_VERSION)
+libavutil/ffversion.h .version:
+	$(M)$(VERSION_SH) $(SRC_PATH) libavutil/ffversion.h $(EXTRA_VERSION)
 	$(Q)touch .version
 
 # force version.sh to run whenever version might have changed
 -include .version
 
-ifdef PROGS
+ifdef AVPROGS
 install: install-progs install-data
 endif
 
@@ -132,7 +143,7 @@ install-libs: install-libs-yes
 install-progs-yes:
 install-progs-$(CONFIG_SHARED): install-libs
 
-install-progs: install-progs-yes $(PROGS)
+install-progs: install-progs-yes $(AVPROGS)
 	$(Q)mkdir -p "$(BINDIR)"
 	$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"
 
@@ -144,13 +155,13 @@ install-data: $(DATA_FILES) $(EXAMPLES_FILES)
 uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data
 
 uninstall-progs:
-	$(RM) $(addprefix "$(BINDIR)/", $(ALLPROGS))
+	$(RM) $(addprefix "$(BINDIR)/", $(ALLAVPROGS))
 
 uninstall-data:
 	$(RM) -r "$(DATADIR)"
 
 clean::
-	$(RM) $(ALLPROGS) $(ALLPROGS_G)
+	$(RM) $(ALLAVPROGS) $(ALLAVPROGS_G)
 	$(RM) $(CLEANSUFFIXES)
 	$(RM) $(CLEANSUFFIXES:%=tools/%)
 	$(RM) -r coverage-html
@@ -158,14 +169,13 @@ clean::
 
 distclean::
 	$(RM) $(DISTCLEANSUFFIXES)
-	$(RM) config.* .config libavutil/avconfig.h .version version.h libavcodec/codec_names.h
+	$(RM) config.* .config libavutil/avconfig.h .version version.h libavutil/ffversion.h libavcodec/codec_names.h
 
 config:
 	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
 
 check: all alltools examples testprogs fate
 
-include $(SRC_PATH)/doc/Makefile
 include $(SRC_PATH)/tests/Makefile
 
 $(sort $(OBJDIRS)):

RELEASE

@@ -1 +1 @@
-2.0
+2.2.3

VERSION

@@ -0,0 +1 @@
+2.2.3

cmdutils.c

@@ -20,6 +20,7 @@
  */
 
 #include <string.h>
+#include <stdint.h>
 #include <stdlib.h>
 #include <errno.h>
 #include <math.h>
@@ -48,8 +49,8 @@
 #include "libavutil/dict.h"
 #include "libavutil/opt.h"
 #include "libavutil/cpu.h"
+#include "libavutil/ffversion.h"
 #include "cmdutils.h"
-#include "version.h"
 #if CONFIG_NETWORK
 #include "libavformat/network.h"
 #endif
@@ -57,10 +58,6 @@
 #include <sys/time.h>
 #include <sys/resource.h>
 #endif
-#if CONFIG_OPENCL
-#include "libavutil/opencl.h"
-#endif
-
 
 static int init_report(const char *env);
 
@@ -68,9 +65,8 @@ struct SwsContext *sws_opts;
 AVDictionary *swr_opts;
 AVDictionary *format_opts, *codec_opts, *resample_opts;
 
-const int this_year = 2013;
-
 static FILE *report_file;
+int hide_banner = 0;
 
 void init_opts(void)
 {
@@ -496,6 +492,9 @@ void parse_loglevel(int argc, char **argv, const OptionDef *options)
             fflush(report_file);
         }
     }
+    idx = locate_option(argc, argv, options, "hide_banner");
+    if (idx)
+        hide_banner = 1;
 }
 
 static const AVOption *opt_find(void *obj, const char *name, const char *unit,
@@ -986,26 +985,6 @@ int opt_timelimit(void *optctx, const char *opt, const char *arg)
     return 0;
 }
 
-#if CONFIG_OPENCL
-int opt_opencl(void *optctx, const char *opt, const char *arg)
-{
-    char *key, *value;
-    const char *opts = arg;
-    int ret = 0;
-    while (*opts) {
-        ret = av_opt_get_key_value(&opts, "=", ":", 0, &key, &value);
-        if (ret < 0)
-            return ret;
-        ret = av_opencl_set_option(key, value);
-        if (ret < 0)
-            return ret;
-        if (*opts)
-            opts++;
-    }
-    return ret;
-}
-#endif
-
 void print_error(const char *filename, int err)
 {
     char errbuf[128];
@@ -1071,7 +1050,7 @@ static void print_program_info(int flags, int level)
     av_log(NULL, level, "%s version " FFMPEG_VERSION, program_name);
     if (flags & SHOW_COPYRIGHT)
         av_log(NULL, level, " Copyright (c) %d-%d the FFmpeg developers",
-               program_birth_year, this_year);
+               program_birth_year, CONFIG_THIS_YEAR);
     av_log(NULL, level, "\n");
     av_log(NULL, level, "%sbuilt on %s %s with %s\n",
            indent, __DATE__, __TIME__, CC_IDENT);
@@ -1079,10 +1058,36 @@ static void print_program_info(int flags, int level)
     av_log(NULL, level, "%sconfiguration: " FFMPEG_CONFIGURATION "\n", indent);
 }
 
+static void print_buildconf(int flags, int level)
+{
+    const char *indent = flags & INDENT ? "  " : "";
+    char str[] = { FFMPEG_CONFIGURATION };
+    char *conflist, *remove_tilde, *splitconf;
+
+    // Change all the ' --' strings to '~--' so that
+    // they can be identified as tokens.
+    while ((conflist = strstr(str, " --")) != NULL) {
+        strncpy(conflist, "~--", 3);
+    }
+
+    // Compensate for the weirdness this would cause
+    // when passing 'pkg-config --static'.
+    while ((remove_tilde = strstr(str, "pkg-config~")) != NULL) {
+        strncpy(remove_tilde, "pkg-config ", 11);
+    }
+
+    splitconf = strtok(str, "~");
+    av_log(NULL, level, "\n%sconfiguration:\n", indent);
+    while (splitconf != NULL) {
+        av_log(NULL, level, "%s%s%s\n", indent, indent, splitconf);
+        splitconf = strtok(NULL, "~");
+    }
+}
+
 void show_banner(int argc, char **argv, const OptionDef *options)
 {
     int idx = locate_option(argc, argv, options, "version");
-    if (idx)
+    if (hide_banner || idx)
         return;
 
     print_program_info (INDENT|SHOW_COPYRIGHT, AV_LOG_INFO);
@@ -1099,6 +1104,14 @@ int show_version(void *optctx, const char *opt, const char *arg)
     return 0;
 }
 
+int show_buildconf(void *optctx, const char *opt, const char *arg)
+{
+    av_log_set_callback(log_callback_help);
+    print_buildconf      (INDENT|0, AV_LOG_INFO);
+
+    return 0;
+}
+
 int show_license(void *optctx, const char *opt, const char *arg)
 {
 #if CONFIG_NONFREE
@@ -1791,7 +1804,7 @@ int read_yesno(void)
 int cmdutils_read_file(const char *filename, char **bufptr, size_t *size)
 {
     int ret;
-    FILE *f = fopen(filename, "rb");
+    FILE *f = av_fopen_utf8(filename, "rb");
 
     if (!f) {
         av_log(NULL, AV_LOG_ERROR, "Cannot read file '%s': %s\n", filename,

cmdutils.h

@@ -43,16 +43,12 @@ extern const char program_name[];
  */
 extern const int program_birth_year;
 
-/**
- * this year, defined by the program for show_banner()
- */
-extern const int this_year;
-
 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
 extern struct SwsContext *sws_opts;
 extern AVDictionary *swr_opts;
 extern AVDictionary *format_opts, *codec_opts, *resample_opts;
+extern int hide_banner;
 
 /**
  * Register a program-specific cleanup routine.
@@ -103,8 +99,12 @@ int opt_max_alloc(void *optctx, const char *opt, const char *arg);
 
 int opt_codec_debug(void *optctx, const char *opt, const char *arg);
 
+#if CONFIG_OPENCL
 int opt_opencl(void *optctx, const char *opt, const char *arg);
 
+int opt_opencl_bench(void *optctx, const char *opt, const char *arg);
+#endif
+
 /**
  * Limit the execution time.
  */
@@ -414,6 +414,13 @@ void show_banner(int argc, char **argv, const OptionDef *options);
  */
 int show_version(void *optctx, const char *opt, const char *arg);
 
+/**
+ * Print the build configuration of the program to stdout. The contents
+ * depend on the definition of FFMPEG_CONFIGURATION.
+ * This option processing function does not utilize the arguments.
+ */
+int show_buildconf(void *optctx, const char *opt, const char *arg);
+
 /**
  * Print the license of the program to stdout. The license depends on
  * the license of the libraries compiled into the program.

cmdutils_common_opts.h

@@ -4,6 +4,7 @@
     { "help"       , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
     { "-help"      , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
     { "version"    , OPT_EXIT, {.func_arg = show_version},      "show version" },
+    { "buildconf"  , OPT_EXIT, {.func_arg = show_buildconf},    "show build configuration" },
     { "formats"    , OPT_EXIT, {.func_arg = show_formats  },    "show available formats" },
     { "codecs"     , OPT_EXIT, {.func_arg = show_codecs   },    "show available codecs" },
     { "decoders"   , OPT_EXIT, {.func_arg = show_decoders },    "show available decoders" },
@@ -20,6 +21,8 @@
     { "report"     , 0,        {(void*)opt_report}, "generate a report" },
     { "max_alloc"  , HAS_ARG,  {.func_arg = opt_max_alloc},     "set maximum size of a single allocated block", "bytes" },
     { "cpuflags"   , HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags }, "force specific cpu flags", "flags" },
+    { "hide_banner", OPT_BOOL | OPT_EXPERT, {&hide_banner},     "do not show program banner", "hide_banner" },
 #if CONFIG_OPENCL
+    { "opencl_bench", OPT_EXIT, {.func_arg = opt_opencl_bench}, "run benchmark on all OpenCL devices and show results" },
     { "opencl_options", HAS_ARG, {.func_arg = opt_opencl},      "set OpenCL environment options" },
 #endif

cmdutils_opencl.c

@@ -0,0 +1,274 @@
+/*
+ * Copyright (C) 2013 Lenny Wang
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/opt.h"
+#include "libavutil/time.h"
+#include "libavutil/log.h"
+#include "libavutil/opencl.h"
+#include "cmdutils.h"
+
+typedef struct {
+    int platform_idx;
+    int device_idx;
+    char device_name[64];
+    int64_t runtime;
+} OpenCLDeviceBenchmark;
+
+const char *ocl_bench_source = AV_OPENCL_KERNEL(
+inline unsigned char clip_uint8(int a)
+{
+    if (a & (~0xFF))
+        return (-a)>>31;
+    else
+        return a;
+}
+
+kernel void unsharp_bench(
+                    global unsigned char *src,
+                    global unsigned char *dst,
+                    global int *mask,
+                    int width,
+                    int height)
+{
+    int i, j, local_idx, lc_idx, sum = 0;
+    int2 thread_idx, block_idx, global_idx, lm_idx;
+    thread_idx.x = get_local_id(0);
+    thread_idx.y = get_local_id(1);
+    block_idx.x = get_group_id(0);
+    block_idx.y = get_group_id(1);
+    global_idx.x = get_global_id(0);
+    global_idx.y = get_global_id(1);
+    local uchar data[32][32];
+    local int lc[128];
+
+    for (i = 0; i <= 1; i++) {
+        lm_idx.y = -8 + (block_idx.y + i) * 16 + thread_idx.y;
+        lm_idx.y = lm_idx.y < 0 ? 0 : lm_idx.y;
+        lm_idx.y = lm_idx.y >= height ? height - 1: lm_idx.y;
+        for (j = 0; j <= 1; j++) {
+            lm_idx.x = -8 + (block_idx.x + j) * 16 + thread_idx.x;
+            lm_idx.x = lm_idx.x < 0 ? 0 : lm_idx.x;
+            lm_idx.x = lm_idx.x >= width ? width - 1: lm_idx.x;
+            data[i*16 + thread_idx.y][j*16 + thread_idx.x] = src[lm_idx.y*width + lm_idx.x];
+        }
+    }
+    local_idx = thread_idx.y*16 + thread_idx.x;
+    if (local_idx < 128)
+        lc[local_idx] = mask[local_idx];
+    barrier(CLK_LOCAL_MEM_FENCE);
+
+    \n#pragma unroll\n
+    for (i = -4; i <= 4; i++) {
+        lm_idx.y = 8 + i + thread_idx.y;
+        \n#pragma unroll\n
+        for (j = -4; j <= 4; j++) {
+            lm_idx.x = 8 + j + thread_idx.x;
+            lc_idx = (i + 4)*8 + j + 4;
+            sum += (int)data[lm_idx.y][lm_idx.x] * lc[lc_idx];
+        }
+    }
+    int temp = (int)data[thread_idx.y + 8][thread_idx.x + 8];
+    int res = temp + (((temp - (int)((sum + 1<<15) >> 16))) >> 16);
+    if (global_idx.x < width && global_idx.y < height)
+        dst[global_idx.x + global_idx.y*width] = clip_uint8(res);
+}
+);
+
+#define OCLCHECK(method, ... )                                                 \
+do {                                                                           \
+    status = method(__VA_ARGS__);                                              \
+    if (status != CL_SUCCESS) {                                                \
+        av_log(NULL, AV_LOG_ERROR, # method " error '%s'\n",                   \
+               av_opencl_errstr(status));                                      \
+        ret = AVERROR_EXTERNAL;                                                \
+        goto end;                                                              \
+    }                                                                          \
+} while (0)
+
+#define CREATEBUF(out, flags, size)                                            \
+do {                                                                           \
+    out = clCreateBuffer(ext_opencl_env->context, flags, size, NULL, &status); \
+    if (status != CL_SUCCESS) {                                                \
+        av_log(NULL, AV_LOG_ERROR, "Could not create OpenCL buffer\n");        \
+        ret = AVERROR_EXTERNAL;                                                \
+        goto end;                                                              \
+    }                                                                          \
+} while (0)
+
+static void fill_rand_int(int *data, int n)
+{
+    int i;
+    srand(av_gettime());
+    for (i = 0; i < n; i++)
+        data[i] = rand();
+}
+
+#define OPENCL_NB_ITER 5
+static int64_t run_opencl_bench(AVOpenCLExternalEnv *ext_opencl_env)
+{
+    int i, arg = 0, width = 1920, height = 1088;
+    int64_t start, ret = 0;
+    cl_int status;
+    size_t kernel_len;
+    char *inbuf;
+    int *mask;
+    int buf_size = width * height * sizeof(char);
+    int mask_size = sizeof(uint32_t) * 128;
+
+    cl_mem cl_mask, cl_inbuf, cl_outbuf;
+    cl_kernel kernel = NULL;
+    cl_program program = NULL;
+    size_t local_work_size_2d[2] = {16, 16};
+    size_t global_work_size_2d[2] = {(size_t)width, (size_t)height};
+
+    if (!(inbuf = av_malloc(buf_size)) || !(mask = av_malloc(mask_size))) {
+        av_log(NULL, AV_LOG_ERROR, "Out of memory\n");
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+    fill_rand_int((int*)inbuf, buf_size/4);
+    fill_rand_int(mask, mask_size/4);
+
+    CREATEBUF(cl_mask, CL_MEM_READ_ONLY, mask_size);
+    CREATEBUF(cl_inbuf, CL_MEM_READ_ONLY, buf_size);
+    CREATEBUF(cl_outbuf, CL_MEM_READ_WRITE, buf_size);
+
+    kernel_len = strlen(ocl_bench_source);
+    program = clCreateProgramWithSource(ext_opencl_env->context, 1, &ocl_bench_source,
+                                        &kernel_len, &status);
+    if (status != CL_SUCCESS || !program) {
+        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark program\n");
+        ret = AVERROR_EXTERNAL;
+        goto end;
+    }
+    status = clBuildProgram(program, 1, &(ext_opencl_env->device_id), NULL, NULL, NULL);
+    if (status != CL_SUCCESS) {
+        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to build benchmark program\n");
+        ret = AVERROR_EXTERNAL;
+        goto end;
+    }
+    kernel = clCreateKernel(program, "unsharp_bench", &status);
+    if (status != CL_SUCCESS) {
+        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark kernel\n");
+        ret = AVERROR_EXTERNAL;
+        goto end;
+    }
+
+    OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_inbuf, CL_TRUE, 0,
+             buf_size, inbuf, 0, NULL, NULL);
+    OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_mask, CL_TRUE, 0,
+             mask_size, mask, 0, NULL, NULL);
+    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_inbuf);
+    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_outbuf);
+    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_mask);
+    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &width);
+    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &height);
+
+    start = av_gettime();
+    for (i = 0; i < OPENCL_NB_ITER; i++)
+        OCLCHECK(clEnqueueNDRangeKernel, ext_opencl_env->command_queue, kernel, 2, NULL,
+                 global_work_size_2d, local_work_size_2d, 0, NULL, NULL);
+    clFinish(ext_opencl_env->command_queue);
+    ret = (av_gettime() - start)/OPENCL_NB_ITER;
+end:
+    if (kernel)
+        clReleaseKernel(kernel);
+    if (program)
+        clReleaseProgram(program);
+    if (cl_inbuf)
+        clReleaseMemObject(cl_inbuf);
+    if (cl_outbuf)
+        clReleaseMemObject(cl_outbuf);
+    if (cl_mask)
+        clReleaseMemObject(cl_mask);
+    av_free(inbuf);
+    av_free(mask);
+    return ret;
+}
+
+static int compare_ocl_device_desc(const void *a, const void *b)
+{
+    return ((OpenCLDeviceBenchmark*)a)->runtime - ((OpenCLDeviceBenchmark*)b)->runtime;
+}
+
+int opt_opencl_bench(void *optctx, const char *opt, const char *arg)
+{
+    int i, j, nb_devices = 0, count = 0;
+    int64_t score = 0;
+    AVOpenCLDeviceList *device_list;
+    AVOpenCLDeviceNode *device_node = NULL;
+    OpenCLDeviceBenchmark *devices = NULL;
+    cl_platform_id platform;
+
+    av_opencl_get_device_list(&device_list);
+    for (i = 0; i < device_list->platform_num; i++)
+        nb_devices += device_list->platform_node[i]->device_num;
+    if (!nb_devices) {
+        av_log(NULL, AV_LOG_ERROR, "No OpenCL device detected!\n");
+        return AVERROR(EINVAL);
+    }
+    if (!(devices = av_malloc(sizeof(OpenCLDeviceBenchmark) * nb_devices))) {
+        av_log(NULL, AV_LOG_ERROR, "Could not allocate buffer\n");
+        return AVERROR(ENOMEM);
+    }
+
+    for (i = 0; i < device_list->platform_num; i++) {
+        for (j = 0; j < device_list->platform_node[i]->device_num; j++) {
+            device_node = device_list->platform_node[i]->device_node[j];
+            platform = device_list->platform_node[i]->platform_id;
+            score = av_opencl_benchmark(device_node, platform, run_opencl_bench);
+            if (score > 0) {
+                devices[count].platform_idx = i;
+                devices[count].device_idx = j;
+                devices[count].runtime = score;
+                strcpy(devices[count].device_name, device_node->device_name);
+                count++;
+            }
+        }
+    }
+    qsort(devices, count, sizeof(OpenCLDeviceBenchmark), compare_ocl_device_desc);
+    fprintf(stderr, "platform_idx\tdevice_idx\tdevice_name\truntime\n");
+    for (i = 0; i < count; i++)
+        fprintf(stdout, "%d\t%d\t%s\t%"PRId64"\n",
+                devices[i].platform_idx, devices[i].device_idx,
+                devices[i].device_name, devices[i].runtime);
+
+    av_opencl_free_device_list(&device_list);
+    av_free(devices);
+    return 0;
+}
+
+int opt_opencl(void *optctx, const char *opt, const char *arg)
+{
+    char *key, *value;
+    const char *opts = arg;
+    int ret = 0;
+    while (*opts) {
+        ret = av_opt_get_key_value(&opts, "=", ":", 0, &key, &value);
+        if (ret < 0)
+            return ret;
+        ret = av_opencl_set_option(key, value);
+        if (ret < 0)
+            return ret;
+        if (*opts)
+            opts++;
+    }
+    return ret;
+}

common.mak

@@ -10,7 +10,7 @@ ifndef SUBDIR
 ifndef V
 Q      = @
 ECHO   = printf "$(1)\t%s\n" $(2)
-BRIEF  = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP
+BRIEF  = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES
 SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM
 
 MSG    = $@
@@ -43,6 +43,7 @@ endef
 COMPILE_C = $(call COMPILE,CC)
 COMPILE_CXX = $(call COMPILE,CXX)
 COMPILE_S = $(call COMPILE,AS)
+COMPILE_HOSTC = $(call COMPILE,HOSTCC)
 
 %.o: %.c
 	$(COMPILE_C)
@@ -56,6 +57,12 @@ COMPILE_S = $(call COMPILE,AS)
 %.o: %.S
 	$(COMPILE_S)
 
+%_host.o: %.c
+	$(COMPILE_HOSTC)
+
+%.o: %.rc
+	$(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $<
+
 %.i: %.c
 	$(CC) $(CCFLAGS) $(CC_E) $<
 
@@ -82,14 +89,15 @@ endif
 include $(SRC_PATH)/arch.mak
 
 OBJS      += $(OBJS-yes)
+SLIBOBJS  += $(SLIBOBJS-yes)
 FFLIBS    := $(FFLIBS-yes) $(FFLIBS)
 TESTPROGS += $(TESTPROGS-yes)
 
 LDLIBS       = $(FFLIBS:%=%$(BUILDSUF))
 FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)
 
-EXAMPLES  := $(EXAMPLES:%=$(SUBDIR)%-example$(EXESUF))
 OBJS      := $(sort $(OBJS:%=$(SUBDIR)%))
+SLIBOBJS  := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
 TESTOBJS  := $(TESTOBJS:%=$(SUBDIR)%) $(TESTPROGS:%=$(SUBDIR)%-test.o)
 TESTPROGS := $(TESTPROGS:%=$(SUBDIR)%-test$(EXESUF))
 HOSTOBJS  := $(HOSTPROGS:%=$(SUBDIR)%.o)
@@ -113,18 +121,19 @@ checkheaders: $(HOBJS)
 alltools: $(TOOLS)
 
 $(HOSTOBJS): %.o: %.c
-	$(call COMPILE,HOSTCC)
+	$(COMPILE_HOSTC)
 
 $(HOSTPROGS): %$(HOSTEXESUF): %.o
-	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $< $(HOSTLIBS)
+	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTLIBS)
 
 $(OBJS):     | $(sort $(dir $(OBJS)))
 $(HOBJS):    | $(sort $(dir $(HOBJS)))
 $(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
+$(SLIBOBJS): | $(sort $(dir $(SLIBOBJS)))
 $(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
 $(TOOLOBJS): | tools
 
-OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(TESTOBJS))
+OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
 
 CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
 DISTCLEANSUFFIXES = *.pc
@@ -139,4 +148,4 @@ endef
 
 $(eval $(RULES))
 
--include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d))
+-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d))

compat/avisynth/avisynth_c.h

@@ -13,7 +13,8 @@
 //
 // You should have received a copy of the GNU General Public License
 // along with this program; if not, write to the Free Software
-// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
+// Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
+// MA 02110-1301 USA, or visit
 // http://www.gnu.org/copyleft/gpl.html .
 //
 // As a special exception, I give you permission to link to the

compat/avisynth/avxsynth_c.h

@@ -13,7 +13,8 @@
 //
 // You should have received a copy of the GNU General Public License
 // along with this program; if not, write to the Free Software
-// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
+// Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
+// MA 02110-1301 USA, or visit
 // http://www.gnu.org/copyleft/gpl.html .
 //
 // As a special exception, I give you permission to link to the

compat/getopt.c

@@ -38,8 +38,6 @@ static int optind = 1;
 static int optopt;
 static char *optarg;
 
-#undef fprintf
-
 static int getopt(int argc, char *argv[], char *opts)
 {
     static int sp = 1;

compat/os2threads.h

@@ -32,6 +32,8 @@
 #undef __STRICT_ANSI__          /* for _beginthread() */
 #include <stdlib.h>
 
+#include "libavutil/mem.h"
+
 typedef TID  pthread_t;
 typedef void pthread_attr_t;
 

compat/va_copy.h

@@ -24,3 +24,6 @@
 #if !defined(va_copy) && defined(_MSC_VER)
 #define va_copy(dst, src) ((dst) = (src))
 #endif
+#if !defined(va_copy) && defined(__GNUC__) && __GNUC__ < 3
+#define va_copy(dst, src) __va_copy(dst, src)
+#endif

compat/windows/makedef

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

doc/APIchanges

@@ -4,7 +4,7 @@ since the last major version increase or the API was added.
 The last version increases were:
 libavcodec:    2013-03-xx
 libavdevice:   2013-03-xx
-libavfilter:   2012-06-22
+libavfilter:   2013-12-xx
 libavformat:   2013-03-xx
 libavresample: 2012-10-05
 libpostproc:   2011-04-18
@@ -15,6 +15,136 @@ libavutil:     2012-10-22
 
 API changes, most recent first:
 
+2014-03-18 - e9c8a9a - lsws 2.5.102
+  Make gray16 full-scale.
+
+2014-xx-xx - xxxxxxx - lavu 53.05.0 - frame.h
+  Add av_frame_copy() for copying the frame data.
+
+2014-02-xx - xxxxxxx - lswr 0.18.100 - swresample.h
+  Add swr_is_initialized() for checking whether a resample context is initialized.
+
+2014-02-xx - xxxxxxx - lavr 1.2.0 - avresample.h
+  Add avresample_is_open() for checking whether a resample context is open.
+
+2014-xx-xx - xxxxxxx - lavu 53.04.0  - opt.h
+  Add AV_OPT_FLAG_EXPORT and AV_OPT_FLAG_READONLY to mark options meant (only)
+  for reading.
+
+2014-xx-xx - xxxxxxx - lavu 53.03.01 - opt.h
+  Deprecate unused AV_OPT_FLAG_METADATA.
+
+2014-02-xx - xxxxxxx - lavd 55.10.100 - avdevice.h
+  Add avdevice_list_devices() and avdevice_free_list_devices()
+
+2014-02-16 - db3c970 - lavf 55.33.100 - avio.h
+  Add avio_find_protocol_name() to find out the name of the protocol that would
+  be selected for a given URL.
+
+2014-02-xx - xxxxxxx - lavu 53.3.0 - frame.h
+  Add AV_FRAME_DATA_DOWNMIX_INFO value to the AVFrameSideDataType enum and
+  downmix_info.h API, which identify downmix-related metadata.
+
+2014-02-11 - 1b05ac2 - lavf 55.32.100 - avformat.h
+  Add av_write_uncoded_frame() and av_interleaved_write_uncoded_frame().
+
+2014-02-04 - 3adb5f8 / d9ae103 - lavf 55.30.100 / 55.11.0 - avformat.h
+  Add AVFormatContext.max_interleave_delta for controlling amount of buffering
+  when interleaving.
+
+2014-02-02 - xxxxxxx - lavf 55.29.100 - avformat.h
+  Add output_ts_offset muxing option to AVFormatContext.
+
+2014-01-27 - 102bd64 - lavd 55.7.100 - avdevice.h
+                       lavf 55.28.100 - avformat.h
+  Add avdevice_dev_to_app_control_message() function.
+
+2014-01-27 - 7151411 - lavd 55.6.100 - avdevice.h
+                       lavf 55.27.100 - avformat.h
+  Add avdevice_app_to_dev_control_message() function.
+
+2014-01-24 - 86bee79 - lavf 55.26.100 - avformat.h
+  Add AVFormatContext option metadata_header_padding to allow control over the
+  amount of padding added.
+
+2014-01-20 - eef74b2 / 93c553c - lavc 55.48.102 / 55.32.1 - avcodec.h
+  Edges are not required anymore on video buffers allocated by get_buffer2()
+  (i.e. as if the CODEC_FLAG_EMU_EDGE flag was always on). Deprecate
+  CODEC_FLAG_EMU_EDGE and avcodec_get_edge_width().
+
+2014-01-19 - xxxxxxx - lavf 55.25.100 - avformat.h
+  Add avformat_get_mov_video_tags() and avformat_get_mov_audio_tags().
+
+2014-01-19 - xxxxxxx - lavu 52.63.100 - rational.h
+  Add av_make_q() function.
+
+2014-01-05 - 4cf4da9 / 5b4797a - lavu 52.62.100 / 53.2.0 - frame.h
+  Add AV_FRAME_DATA_MATRIXENCODING value to the AVFrameSideDataType enum, which
+  identifies AVMatrixEncoding data.
+
+2014-01-05 - 751385f / 5c437fb - lavu 52.61.100 / 53.1.0 - channel_layout.h
+  Add values for various Dolby flags to the AVMatrixEncoding enum.
+
+2014-01-04 - xxxxxxx - lavu 52.60.100 - mathematics.h
+  Add av_add_stable() function.
+
+2013-12-22 - xxxxxxx - lavu 52.59.100 - avstring.h
+  Add av_strnlen() function.
+
+2013-12-xx - xxxxxxx - lavu 52.57.100 - opencl.h
+  Add av_opencl_benchmark() function.
+
+2013-11-xx - xxxxxxx - lavu 52.56.100 - ffversion.h
+  Moves version.h to libavutil/ffversion.h.
+  Install ffversion.h and make it public.
+
+2013-12-11 - 29c83d2 / b9fb59d,409a143 / 9431356,44967ab / d7b3ee9 - lavc 55.45.101 / 55.28.1 - avcodec.h
+  av_frame_alloc(), av_frame_unref() and av_frame_free() now can and should be
+  used instead of avcodec_alloc_frame(), avcodec_get_frame_defaults() and
+  avcodec_free_frame() respectively. The latter three functions are deprecated.
+
+2013-12-09 - 7a60348 / 7e244c6- - lavu 52.58.100 / 52.20.0 - frame.h
+  Add AV_FRAME_DATA_STEREO3D value to the AVFrameSideDataType enum and
+  stereo3d.h API, that identify codec-independent stereo3d information.
+
+2013-11-26 - 625b290 / 1eaac1d- - lavu 52.55.100 / 52.19.0 - frame.h
+  Add AV_FRAME_DATA_A53_CC value to the AVFrameSideDataType enum, which
+  identifies ATSC A53 Part 4 Closed Captions data.
+
+2013-11-XX - xxxxxxx - lavu 52.54.100 - avstring.h
+  Add av_utf8_decode() function.
+
+2013-11-22 - fb7d70c - lavc 55.44.100 - avcodec.h
+  Add HEVC profiles
+
+2013-11-xx - xxxxxxx - lavc 55.44.100 - avcodec.h
+  Add av_packet_{un,}pack_dictionary()
+  Add AV_PKT_METADATA_UPDATE side data type, used to transmit key/value
+  strings between a stream and the application.
+
+2013-11-14 - 7c888ae / cce3e0a - lavu 52.53.100 / 52.18.0 - mem.h
+  Move av_fast_malloc() and av_fast_realloc() for libavcodec to libavutil.
+
+2013-11-14 - b71e4d8 / 8941971 - lavc 55.43.100 / 55.27.0 - avcodec.h
+  Deprecate AVCodecContext.error_rate, it is replaced by the 'error_rate'
+  private option of the mpegvideo encoder family.
+
+2013-11-14 - 31c09b7 / 728c465 - lavc 55.26.0 - vdpau.h
+  Add av_vdpau_get_profile().
+  Add av_vdpau_alloc_context(). This function must from now on be
+  used for allocating AVVDPAUContext.
+
+2013-11-04 - be41f21 / cd8f772 - lavc 55.41.100 / 55.25.0 - avcodec.h
+                       lavu 52.51.100 - frame.h
+  Add ITU-R BT.2020 and other not yet included values to color primaries,
+  transfer characteristics and colorspaces.
+
+2013-11-04 - xxxxxxx - lavu 52.50.100 - avutil.h
+  Add av_fopen_utf8()
+
+2013-10-31 - 78265fc / 28096e0 - lavu 52.49.100 / 52.17.0 - frame.h
+  Add AVFrame.flags and AV_FRAME_FLAG_CORRUPT.
+
 2013-10-27 - xxxxxxx - lavc 55.39.100 - avcodec.h
   Add CODEC_CAP_DELAY support to avcodec_decode_subtitle2.
 
@@ -35,11 +165,11 @@ API changes, most recent first:
   Add audio/video/subtitle AVCodec fields to AVFormatContext to force specific
   decoders
 
-2013-08-xx - xxxxxxx - lavfi 3.11.0 - avfilter.h
+2013-09-28 - 7381d31 / 0767bfd - lavfi 3.88.100 / 3.11.0 - avfilter.h
   Add AVFilterGraph.execute and AVFilterGraph.opaque for custom slice threading
   implementations.
 
-2013-09-21 - xxxxxxx - lavu 52.16.0 - pixfmt.h
+2013-09-21 - 85f8a3c / e208e6d - lavu 52.46.100 / 52.16.0 - pixfmt.h
   Add interleaved 4:2:2 8/10-bit formats AV_PIX_FMT_NV16 and
   AV_PIX_FMT_NV20.
 

doc/Doxyfile

@@ -31,7 +31,7 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.
 
-PROJECT_NUMBER         =
+PROJECT_NUMBER         = 2.2.3
 
 # With the PROJECT_LOGO tag one can specify a logo or icon that is included
 # in the documentation. The maximum height of the logo should not exceed 55
@@ -793,13 +793,13 @@ HTML_FILE_EXTENSION    = .html
 # each generated HTML page. If it is left blank doxygen will generate a
 # standard header.
 
-#HTML_HEADER            = doc/doxy/header.html
+HTML_HEADER            =
 
 # The HTML_FOOTER tag can be used to specify a personal HTML footer for
 # each generated HTML page. If it is left blank doxygen will generate a
 # standard footer.
 
-#HTML_FOOTER            = doc/doxy/footer.html
+HTML_FOOTER            =
 
 # The HTML_STYLESHEET tag can be used to specify a user-defined cascading
 # style sheet that is used by each HTML page. It can be used to
@@ -808,7 +808,7 @@ HTML_FILE_EXTENSION    = .html
 # the style sheet file to the HTML output directory, so don't put your own
 # stylesheet in the HTML output directory as well, or it will be erased!
 
-#HTML_STYLESHEET        = doc/doxy/doxy_stylesheet.css
+HTML_STYLESHEET        =
 
 # The HTML_COLORSTYLE_HUE tag controls the color of the HTML output.
 # Doxygen will adjust the colors in the stylesheet and background images

doc/Makefile

@@ -14,11 +14,11 @@ COMPONENTS-$(CONFIG_AVFORMAT)   += ffmpeg-formats ffmpeg-protocols
 COMPONENTS-$(CONFIG_AVDEVICE)   += ffmpeg-devices
 COMPONENTS-$(CONFIG_AVFILTER)   += ffmpeg-filters
 
-MANPAGES1   = $(PROGS-yes:%=doc/%.1)    $(PROGS-yes:%=doc/%-all.1)    $(COMPONENTS-yes:%=doc/%.1)
+MANPAGES1   = $(AVPROGS-yes:%=doc/%.1)    $(AVPROGS-yes:%=doc/%-all.1)    $(COMPONENTS-yes:%=doc/%.1)
 MANPAGES3   = $(LIBRARIES-yes:%=doc/%.3)
 MANPAGES    = $(MANPAGES1) $(MANPAGES3)
-PODPAGES    = $(PROGS-yes:%=doc/%.pod)  $(PROGS-yes:%=doc/%-all.pod)  $(COMPONENTS-yes:%=doc/%.pod)  $(LIBRARIES-yes:%=doc/%.pod)
-HTMLPAGES   = $(PROGS-yes:%=doc/%.html) $(PROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
+PODPAGES    = $(AVPROGS-yes:%=doc/%.pod)  $(AVPROGS-yes:%=doc/%-all.pod)  $(COMPONENTS-yes:%=doc/%.pod)  $(LIBRARIES-yes:%=doc/%.pod)
+HTMLPAGES   = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
               doc/developer.html                                        \
               doc/faq.html                                              \
               doc/fate.html                                             \
@@ -36,6 +36,25 @@ DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
 DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
 DOCS = $(DOCS-yes)
 
+DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
+DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE)           += avcodec
+DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
+DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE)      += filter_audio
+DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE)   += filtering_audio
+DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE)   += filtering_video
+DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE)          += metadata
+DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE)            += muxing
+DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE)          += remuxing
+DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)  += resampling_audio
+DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)     += scaling_video
+DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE)     += transcode_aac
+ALL_DOC_EXAMPLES_LIST = $(DOC_EXAMPLES-) $(DOC_EXAMPLES-yes)
+
+DOC_EXAMPLES       := $(DOC_EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
+ALL_DOC_EXAMPLES   := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)$(EXESUF))
+ALL_DOC_EXAMPLES_G := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
+PROGS              += $(DOC_EXAMPLES)
+
 all-$(CONFIG_DOC): doc
 
 doc: documentation
@@ -43,7 +62,9 @@ doc: documentation
 apidoc: doc/doxy/html
 documentation: $(DOCS)
 
-TEXIDEP = awk '/^@(verbatim)?include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)
+examples: $(DOC_EXAMPLES)
+
+TEXIDEP = perl $(SRC_PATH)/doc/texidep.pl $(SRC_PATH) $< $@ >$(@:%=%.d)
 
 doc/%.txt: TAG = TXT
 doc/%.txt: doc/%.texi
@@ -84,9 +105,13 @@ doc/%.3: doc/%.pod $(GENTEXI)
 	$(M)pod2man --section=3 --center=" " --release=" " $< > $@
 
 $(DOCS) doc/doxy/html: | doc/
+$(DOC_EXAMPLES:%$(EXESUF)=%.o): | doc/examples
+OBJDIRS += doc/examples
 
-doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(INSTHEADERS)
-	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $^
+DOXY_INPUT      = $(addprefix $(SRC_PATH)/, $(INSTHEADERS) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c))
+
+doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(DOXY_INPUT)
+	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $< $(DOXY_INPUT)
 
 install-doc: install-html install-man
 
@@ -120,7 +145,7 @@ uninstall-html:
 	$(RM) -r "$(DOCDIR)"
 
 uninstall-man:
-	$(RM) $(addprefix "$(MANDIR)/man1/",$(PROGS-yes:%=%.1) $(PROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
+	$(RM) $(addprefix "$(MANDIR)/man1/",$(AVPROGS-yes:%=%.1) $(AVPROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
 	$(RM) $(addprefix "$(MANDIR)/man3/",$(LIBRARIES-yes:%=%.3))
 
 clean:: docclean
@@ -128,8 +153,13 @@ clean:: docclean
 distclean:: docclean
 	$(RM) doc/config.texi
 
-docclean:
-	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 $(CLEANSUFFIXES:%=doc/%) doc/avoptions_*.texi
+examplesclean:
+	$(RM) $(ALL_DOC_EXAMPLES) $(ALL_DOC_EXAMPLES_G)
+	$(RM) $(CLEANSUFFIXES:%=doc/examples/%)
+
+docclean: examplesclean
+	$(RM) $(CLEANSUFFIXES:%=doc/%)
+	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 doc/avoptions_*.texi
 	$(RM) -r doc/doxy/html
 
 -include $(wildcard $(DOCS:%=%.d))

doc/RELEASE_NOTES

@@ -1,7 +1,7 @@
 Release Notes
 =============
 
-* 2.1 "Fourier"  October, 2013
+* 2.2 "Muybridge"  March, 2014
 
 
 General notes

doc/avutil.txt

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

doc/bitstream_filters.texi

@@ -117,8 +117,6 @@ ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
 
 @section movsub
 
-@section mp3_header_compress
-
 @section mp3_header_decompress
 
 @section noise

doc/codecs.texi

@@ -172,7 +172,13 @@ Set max video quantizer scale (VBR). Must be included between -1 and
 Set max difference between the quantizer scale (VBR).
 
 @item bf @var{integer} (@emph{encoding,video})
-Set max number of B frames.
+Set max number of B frames between non-B-frames.
+
+Must be an integer between -1 and 16. 0 means that B-frames are
+disabled. If a value of -1 is used, it will choose an automatic value
+depending on the encoder.
+
+Default value is 0.
 
 @item b_qfactor @var{float} (@emph{encoding,video})
 Set qp factor between P and B frames.
@@ -871,6 +877,9 @@ Set frame skip factor.
 
 @item skip_exp @var{integer} (@emph{encoding,video})
 Set frame skip exponent.
+Negative values behave identical to the corresponding positive ones, except
+that the score is normalized.
+Positive values exist primarly for compatibility reasons and are not so useful.
 
 @item skipcmp @var{integer} (@emph{encoding,video})
 Set frame skip compare function.
@@ -1083,5 +1092,9 @@ instead of alpha. Default is 0.
 
 @c man end CODEC OPTIONS
 
+@ifclear config-writeonly
 @include decoders.texi
+@end ifclear
+@ifclear config-readonly
 @include encoders.texi
+@end ifclear

doc/decoders.texi

@@ -14,7 +14,7 @@ You can disable all the decoders with the configure option
 with the options @code{--enable-decoder=@var{DECODER}} /
 @code{--disable-decoder=@var{DECODER}}.
 
-The option @code{-codecs} of the ff* tools will display the list of
+The option @code{-decoders} of the ff* tools will display the list of
 enabled decoders.
 
 @c man end DECODERS
@@ -52,6 +52,37 @@ top-field-first is assumed
 @chapter Audio Decoders
 @c man begin AUDIO DECODERS
 
+A description of some of the currently available audio decoders
+follows.
+
+@section ac3
+
+AC-3 audio decoder.
+
+This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as well as
+the undocumented RealAudio 3 (a.k.a. dnet).
+
+@subsection AC-3 Decoder Options
+
+@table @option
+
+@item -drc_scale @var{value}
+Dynamic Range Scale Factor. The factor to apply to dynamic range values
+from the AC-3 stream. This factor is applied exponentially.
+There are 3 notable scale factor ranges:
+@table @option
+@item drc_scale == 0
+DRC disabled. Produces full range audio.
+@item 0 < drc_scale <= 1
+DRC enabled.  Applies a fraction of the stream DRC value.
+Audio reproduction is between full range and full compression.
+@item drc_scale > 1
+DRC enabled. Applies drc_scale asymmetrically.
+Loud sounds are fully compressed.  Soft sounds are enhanced.
+@end table
+
+@end table
+
 @section ffwavesynth
 
 Internal wave synthetizer.

doc/default.css

@@ -17,8 +17,8 @@ a:visited {
 }
 
 #banner img {
-    padding-bottom: 1px;
-    padding-top: 5px;
+    margin-bottom: 1px;
+    margin-top: 5px;
 }
 
 #body {

doc/demuxers.texi

@@ -296,7 +296,7 @@ teletext packet PTS and DTS values untouched.
 
 Raw video demuxer.
 
-This demuxer allows to read raw video data. Since there is no header
+This demuxer allows one to read raw video data. Since there is no header
 specifying the assumed video parameters, the user must specify them
 in order to be able to decode the data correctly.
 

doc/developer.texi

@@ -92,7 +92,7 @@ for markup commands, i.e. use @code{@@param} and not @code{\param}.
  * more text ...
  * ...
  */
-typedef struct Foobar@{
+typedef struct Foobar @{
     int var1; /**< var1 description */
     int var2; ///< var2 description
     /** var3 description */
@@ -248,7 +248,7 @@ Contributions should be licensed under the
 @uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
 including an "or any later version" clause, or, if you prefer
 a gift-style license, the
-@uref{http://www.isc.org/software/license/, ISC} or
+@uref{http://opensource.org/licenses/isc-license.txt, ISC} or
 @uref{http://mit-license.org/, MIT} license.
 @uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
 an "or any later version" clause is also acceptable, but LGPL is

doc/devices.texi

@@ -17,5 +17,9 @@ for programmatic use.
 
 @c man end DEVICE OPTIONS
 
+@ifclear config-writeonly
 @include indevs.texi
+@end ifclear
+@ifclear config-readonly
 @include outdevs.texi
+@end ifclear

doc/doxy-wrapper.sh

@@ -8,7 +8,5 @@ shift 2
 doxygen - <<EOF
 @INCLUDE        = ${DOXYFILE}
 INPUT           = $@
-HTML_HEADER     = ${SRC_PATH}/doc/doxy/header.html
-HTML_FOOTER     = ${SRC_PATH}/doc/doxy/footer.html
-HTML_STYLESHEET = ${SRC_PATH}/doc/doxy/doxy_stylesheet.css
+EXAMPLE_PATH    = ${SRC_PATH}/doc/examples
 EOF

doc/doxy/footer.html

@@ -1,9 +0,0 @@
-
- <footer class="footer pagination-right">
-   <span class="label label-info">
-    Generated on $datetime for $projectname by&#160;<a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion
-   </span>
- </footer>
-</div>
-</body>
-</html>

doc/doxy/header.html

@@ -1,16 +0,0 @@
-<!DOCTYPE html>
-<html>
-<head>
-<meta http-equiv="Content-Type" content="text/html; charset=UTF-8"/>
-<meta http-equiv="X-UA-Compatible" content="IE=9"/>
-<!--BEGIN PROJECT_NAME--><title>$projectname: $title</title><!--END PROJECT_NAME-->
-<!--BEGIN !PROJECT_NAME--><title>$title</title><!--END !PROJECT_NAME-->
-<link href="$relpath$doxy_stylesheet.css" rel="stylesheet" type="text/css" />
-<!--Header replace -->
-
-</head>
-
-<div class="container">
-
-<!--Header replace -->
-<div class="menu">

doc/encoders.texi

@@ -14,7 +14,7 @@ You can disable all the encoders with the configure option
 with the options @code{--enable-encoder=@var{ENCODER}} /
 @code{--disable-encoder=@var{ENCODER}}.
 
-The option @code{-codecs} of the ff* tools will display the list of
+The option @code{-encoders} of the ff* tools will display the list of
 enabled encoders.
 
 @c man end ENCODERS
@@ -38,8 +38,8 @@ As this encoder is experimental, unexpected behavior may exist from time to
 time. For a more stable AAC encoder, see @ref{libvo-aacenc}. However, be warned
 that it has a worse quality reported by some users.
 
-@c Comment this out until somebody writes the respective documentation.
-@c See also @ref{libfaac}, @ref{libaacplus}, and @ref{libfdk-aac-enc}.
+@c todo @ref{libaacplus}
+See also @ref{libfdk-aac-enc,,libfdk_aac} and @ref{libfaac}.
 
 @subsection Options
 
@@ -494,6 +494,285 @@ Selected by Encoder (default)
 
 @end table
 
+@anchor{libfaac}
+@section libfaac
+
+libfaac AAC (Advanced Audio Coding) encoder wrapper.
+
+Requires the presence of the libfaac headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libfaac --enable-nonfree}.
+
+This encoder is considered to be of higher quality with respect to the
+@ref{aacenc,,the native experimental FFmpeg AAC encoder}.
+
+For more information see the libfaac project at
+@url{http://www.audiocoding.com/faac.html/}.
+
+@subsection Options
+
+The following shared FFmpeg codec options are recognized.
+
+The following options are supported by the libfaac wrapper. The
+@command{faac}-equivalent of the options are listed in parentheses.
+
+@table @option
+@item b (@emph{-b})
+Set bit rate in bits/s for ABR (Average Bit Rate) mode. If the bit rate
+is not explicitly specified, it is automatically set to a suitable
+value depending on the selected profile. @command{faac} bitrate is
+expressed in kilobits/s.
+
+Note that libfaac does not support CBR (Constant Bit Rate) but only
+ABR (Average Bit Rate).
+
+If VBR mode is enabled this option is ignored.
+
+@item ar (@emph{-R})
+Set audio sampling rate (in Hz).
+
+@item ac (@emph{-c})
+Set the number of audio channels.
+
+@item cutoff (@emph{-C})
+Set cutoff frequency. If not specified (or explicitly set to 0) it
+will use a value automatically computed by the library. Default value
+is 0.
+
+@item profile
+Set audio profile.
+
+The following profiles are recognized:
+@table @samp
+@item aac_main
+Main AAC (Main)
+
+@item aac_low
+Low Complexity AAC (LC)
+
+@item aac_ssr
+Scalable Sample Rate (SSR)
+
+@item aac_ltp
+Long Term Prediction (LTP)
+@end table
+
+If not specified it is set to @samp{aac_low}.
+
+@item flags +qscale
+Set constant quality VBR (Variable Bit Rate) mode.
+
+@item global_quality
+Set quality in VBR mode as an integer number of lambda units.
+
+Only relevant when VBR mode is enabled with @code{flags +qscale}.  The
+value is converted to QP units by dividing it by @code{FF_QP2LAMBDA},
+and used to set the quality value used by libfaac. A reasonable range
+for the option value in QP units is [10-500], the higher the value the
+higher the quality.
+
+@item q (@emph{-q})
+Enable VBR mode when set to a non-negative value, and set constant
+quality value as a double floating point value in QP units.
+
+The value sets the quality value used by libfaac. A reasonable range
+for the option value is [10-500], the higher the value the higher the
+quality.
+
+This option is valid only using the @command{ffmpeg} command-line
+tool. For library interface users, use @option{global_quality}.
+@end table
+
+@subsection Examples
+
+@itemize
+@item
+Use @command{ffmpeg} to convert an audio file to ABR 128 kbps AAC in an M4A (MP4)
+container:
+@example
+ffmpeg -i input.wav -codec:a libfaac -b:a 128k -output.m4a
+@end example
+
+@item
+Use @command{ffmpeg} to convert an audio file to VBR AAC, using the
+LTP AAC profile:
+@example
+ffmpeg -i input.wav -c:a libfaac -profile:a aac_ltp -q:a 100 output.m4a
+@end example
+@end itemize
+
+@anchor{libfdk-aac-enc}
+@section libfdk_aac
+
+libfdk-aac AAC (Advanced Audio Coding) encoder wrapper.
+
+The libfdk-aac library is based on the Fraunhofer FDK AAC code from
+the Android project.
+
+Requires the presence of the libfdk-aac headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libfdk-aac}. The library is also incompatible with GPL,
+so if you allow the use of GPL, you should configure with
+@code{--enable-gpl --enable-nonfree --enable-libfdk-aac}.
+
+This encoder is considered to be of higher quality with respect to
+both @ref{aacenc,,the native experimental FFmpeg AAC encoder} and
+@ref{libfaac}.
+
+VBR encoding, enabled through the @option{vbr} or @option{flags
++qscale} options, is experimental and only works with some
+combinations of parameters.
+
+Support for encoding 7.1 audio is only available with libfdk-aac 0.1.3 or
+higher.
+
+For more information see the fdk-aac project at
+@url{http://sourceforge.net/p/opencore-amr/fdk-aac/}.
+
+@subsection Options
+
+The following options are mapped on the shared FFmpeg codec options.
+
+@table @option
+@item b
+Set bit rate in bits/s. If the bitrate is not explicitly specified, it
+is automatically set to a suitable value depending on the selected
+profile.
+
+In case VBR mode is enabled the option is ignored.
+
+@item ar
+Set audio sampling rate (in Hz).
+
+@item channels
+Set the number of audio channels.
+
+@item flags +qscale
+Enable fixed quality, VBR (Variable Bit Rate) mode.
+Note that VBR is implicitly enabled when the @option{vbr} value is
+positive.
+
+@item cutoff
+Set cutoff frequency. If not specified (or explicitly set to 0) it
+will use a value automatically computed by the library. Default value
+is 0.
+
+@item profile
+Set audio profile.
+
+The following profiles are recognized:
+@table @samp
+@item aac_low
+Low Complexity AAC (LC)
+
+@item aac_he
+High Efficiency AAC (HE-AAC)
+
+@item aac_he_v2
+High Efficiency AAC version 2 (HE-AACv2)
+
+@item aac_ld
+Low Delay AAC (LD)
+
+@item aac_eld
+Enhanced Low Delay AAC (ELD)
+@end table
+
+If not specified it is set to @samp{aac_low}.
+@end table
+
+The following are private options of the libfdk_aac encoder.
+
+@table @option
+@item afterburner
+Enable afterburner feature if set to 1, disabled if set to 0. This
+improves the quality but also the required processing power.
+
+Default value is 1.
+
+@item eld_sbr
+Enable SBR (Spectral Band Replication) for ELD if set to 1, disabled
+if set to 0.
+
+Default value is 0.
+
+@item signaling
+Set SBR/PS signaling style.
+
+It can assume one of the following values:
+@table @samp
+@item default
+choose signaling implicitly (explicit hierarchical by default,
+implicit if global header is disabled)
+
+@item implicit
+implicit backwards compatible signaling
+
+@item explicit_sbr
+explicit SBR, implicit PS signaling
+
+@item explicit_hierarchical
+explicit hierarchical signaling
+@end table
+
+Default value is @samp{default}.
+
+@item latm
+Output LATM/LOAS encapsulated data if set to 1, disabled if set to 0.
+
+Default value is 0.
+
+@item header_period
+Set StreamMuxConfig and PCE repetition period (in frames) for sending
+in-band configuration buffers within LATM/LOAS transport layer.
+
+Must be a 16-bits non-negative integer.
+
+Default value is 0.
+
+@item vbr
+Set VBR mode, from 1 to 5. 1 is lowest quality (though still pretty
+good) and 5 is highest quality. A value of 0 will disable VBR, and CBR
+(Constant Bit Rate) is enabled.
+
+Currently only the @samp{aac_low} profile supports VBR encoding.
+
+VBR modes 1-5 correspond to roughly the following average bit rates:
+
+@table @samp
+@item 1
+32 kbps/channel
+@item 2
+40 kbps/channel
+@item 3
+48-56 kbps/channel
+@item 4
+64 kbps/channel
+@item 5
+about 80-96 kbps/channel
+@end table
+
+Default value is 0.
+@end table
+
+@subsection Examples
+
+@itemize
+@item
+Use @command{ffmpeg} to convert an audio file to VBR AAC in an M4A (MP4)
+container:
+@example
+ffmpeg -i input.wav -codec:a libfdk_aac -vbr 3 output.m4a
+@end example
+
+@item
+Use @command{ffmpeg} to convert an audio file to CBR 64k kbps AAC, using the
+High-Efficiency AAC profile:
+@example
+ffmpeg -i input.wav -c:a libfdk_aac -profile:a aac_he -b:a 64k output.m4a
+@end example
+@end itemize
+
 @anchor{libmp3lame}
 @section libmp3lame
 
@@ -513,7 +792,7 @@ The following options are supported by the libmp3lame wrapper. The
 
 @table @option
 @item b (@emph{-b})
-Set bitrate expressed in bits/s for CBR. LAME @code{bitrate} is
+Set bitrate expressed in bits/s for CBR or ABR. LAME @code{bitrate} is
 expressed in kilobits/s.
 
 @item q (@emph{-V})
@@ -528,13 +807,18 @@ while producing the worst quality.
 
 @item reservoir
 Enable use of bit reservoir when set to 1. Default value is 1. LAME
-has this enabled by default, but can be overriden by use
+has this enabled by default, but can be overridden by use
 @option{--nores} option.
 
 @item joint_stereo (@emph{-m j})
 Enable the encoder to use (on a frame by frame basis) either L/R
 stereo or mid/side stereo. Default value is 1.
 
+@item abr (@emph{--abr})
+Enable the encoder to use ABR when set to 1. The @command{lame}
+@option{--abr} sets the target bitrate, while this options only
+tells FFmpeg to use ABR still relies on @option{b} to set bitrate.
+
 @end table
 
 @section libopencore-amrnb
@@ -786,7 +1070,7 @@ Set maximum frame size, or duration of a frame in milliseconds. The
 argument must be exactly the following: 2.5, 5, 10, 20, 40, 60. Smaller
 frame sizes achieve lower latency but less quality at a given bitrate.
 Sizes greater than 20ms are only interesting at fairly low bitrates.
-The default of FFmpeg is 10ms, but is 20ms in @command{opusenc}.
+The default is 20ms.
 
 @item packet_loss (@emph{expect-loss})
 Set expected packet loss percentage. The default is 0.
@@ -863,32 +1147,111 @@ transient response is a higher bitrate.
 
 @end table
 
+@anchor{libwavpack}
 @section libwavpack
 
 A wrapper providing WavPack encoding through libwavpack.
 
 Only lossless mode using 32-bit integer samples is supported currently.
-The @option{compression_level} option can be used to control speed vs.
-compression tradeoff, with the values mapped to libwavpack as follows:
+
+Requires the presence of the libwavpack headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libwavpack}.
+
+Note that a libavcodec-native encoder for the WavPack codec exists so users can
+encode audios with this codec without using this encoder. See @ref{wavpackenc}.
+
+@subsection Options
+
+@command{wavpack} command line utility's corresponding options are listed in
+parentheses, if any.
 
 @table @option
+@item frame_size (@emph{--blocksize})
+Default is 32768.
 
-@item 0
-Fast mode - corresponding to the wavpack @option{-f} option.
+@item compression_level
+Set speed vs. compression tradeoff. Acceptable arguments are listed below:
+
+@table @samp
+@item 0 (@emph{-f})
+Fast mode.
 
 @item 1
 Normal (default) settings.
 
-@item 2
-High quality - corresponding to the wavpack @option{-h} option.
+@item 2 (@emph{-h})
+High quality.
 
-@item 3
-Very high quality - corresponding to the wavpack @option{-hh} option.
+@item 3 (@emph{-hh})
+Very high quality.
 
-@item 4-8
-Same as 3, but with extra processing enabled - corresponding to the wavpack
-@option{-x} option. I.e. 4 is the same as @option{-x2} and 8 is the same as
-@option{-x6}.
+@item 4-8 (@emph{-hh -x}@var{EXTRAPROC})
+Same as @samp{3}, but with extra processing enabled.
+
+@samp{4} is the same as @option{-x2} and @samp{8} is the same as @option{-x6}.
+
+@end table
+@end table
+
+@anchor{wavpackenc}
+@section wavpack
+
+WavPack lossless audio encoder.
+
+This is a libavcodec-native WavPack encoder. There is also an encoder based on
+libwavpack, but there is virtually no reason to use that encoder.
+
+See also @ref{libwavpack}.
+
+@subsection Options
+
+The equivalent options for @command{wavpack} command line utility are listed in
+parentheses.
+
+@subsubsection Shared options
+
+The following shared options are effective for this encoder. Only special notes
+about this particular encoder will be documented here. For the general meaning
+of the options, see @ref{codec-options,,the Codec Options chapter}.
+
+@table @option
+@item frame_size (@emph{--blocksize})
+For this encoder, the range for this option is between 128 and 131072. Default
+is automatically decided based on sample rate and number of channel.
+
+For the complete formula of calculating default, see
+@file{libavcodec/wavpackenc.c}.
+
+@item compression_level (@emph{-f}, @emph{-h}, @emph{-hh}, and @emph{-x})
+This option's syntax is consistent with @ref{libwavpack}'s.
+@end table
+
+@subsubsection Private options
+
+@table @option
+@item joint_stereo (@emph{-j})
+Set whether to enable joint stereo. Valid values are:
+
+@table @samp
+@item on (@emph{1})
+Force mid/side audio encoding.
+@item off (@emph{0})
+Force left/right audio encoding.
+@item auto
+Let the encoder decide automatically.
+@end table
+
+@item optimize_mono
+Set whether to enable optimization for mono. This option is only effective for
+non-mono streams. Available values:
+
+@table @samp
+@item on
+enabled
+@item off
+disabled
+@end table
 
 @end table
 
@@ -902,12 +1265,15 @@ follows.
 
 @section libtheora
 
-Theora format supported through libtheora.
+libtheora Theora encoder wrapper.
 
 Requires the presence of the libtheora headers and library during
 configuration. You need to explicitly configure the build with
 @code{--enable-libtheora}.
 
+For more information about the libtheora project see
+@url{http://www.theora.org/}.
+
 @subsection Options
 
 The following global options are mapped to internal libtheora options
@@ -915,11 +1281,11 @@ which affect the quality and the bitrate of the encoded stream.
 
 @table @option
 @item b
-Set the video bitrate, only works if the @code{qscale} flag in
-@option{flags} is not enabled.
+Set the video bitrate in bit/s for CBR (Constant Bit Rate) mode.  In
+case VBR (Variable Bit Rate) mode is enabled this option is ignored.
 
 @item flags
-Used to enable constant quality mode encoding through the
+Used to enable constant quality mode (VBR) encoding through the
 @option{qscale} flag, and to enable the @code{pass1} and @code{pass2}
 modes.
 
@@ -927,19 +1293,41 @@ modes.
 Set the GOP size.
 
 @item global_quality
-Set the global quality in lambda units, only works if the
-@code{qscale} flag in @option{flags} is enabled. The value is clipped
-in the [0 - 10*@code{FF_QP2LAMBDA}] range, and then multiplied for 6.3
-to get a value in the native libtheora range [0-63]. A higher value
-corresponds to a higher quality.
+Set the global quality as an integer in lambda units.
 
-For example, to set maximum constant quality encoding with
-@command{ffmpeg}:
-@example
-ffmpeg -i INPUT -flags:v qscale -global_quality:v "10*QP2LAMBDA" -codec:v libtheora OUTPUT.ogg
-@end example
+Only relevant when VBR mode is enabled with @code{flags +qscale}. The
+value is converted to QP units by dividing it by @code{FF_QP2LAMBDA},
+clipped in the [0 - 10] range, and then multiplied by 6.3 to get a
+value in the native libtheora range [0-63]. A higher value corresponds
+to a higher quality.
+
+@item q
+Enable VBR mode when set to a non-negative value, and set constant
+quality value as a double floating point value in QP units.
+
+The value is clipped in the [0-10] range, and then multiplied by 6.3
+to get a value in the native libtheora range [0-63].
+
+This option is valid only using the @command{ffmpeg} command-line
+tool. For library interface users, use @option{global_quality}.
 @end table
 
+@subsection Examples
+
+@itemize
+@item
+Set maximum constant quality (VBR) encoding with @command{ffmpeg}:
+@example
+ffmpeg -i INPUT -codec:v libtheora -q:v 10 OUTPUT.ogg
+@end example
+
+@item
+Use @command{ffmpeg} to convert a CBR 1000 kbps Theora video stream:
+@example
+ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg
+@end example
+@end itemize
+
 @section libvpx
 
 VP8 format supported through libvpx.
@@ -1059,7 +1447,69 @@ g_error_resilient
 For more information about libvpx see:
 @url{http://www.webmproject.org/}
 
-@section libx264
+
+@section libwebp
+
+libwebp WebP Image encoder wrapper
+
+libwebp is Google's official encoder for WebP images. It can encode in either
+lossy or lossless mode. Lossy images are essentially a wrapper around a VP8
+frame. Lossless images are a separate codec developed by Google.
+
+@subsection Pixel Format
+
+Currently, libwebp only supports YUV420 for lossy and RGB for lossless due
+to limitations of the format and libwebp. Alpha is supported for either mode.
+Because of API limitations, if RGB is passed in when encoding lossy or YUV is
+passed in for encoding lossless, the pixel format will automatically be
+converted using functions from libwebp. This is not ideal and is done only for
+convenience.
+
+@subsection Options
+
+@table @option
+
+@item -lossless @var{boolean}
+Enables/Disables use of lossless mode. Default is 0.
+
+@item -compression_level @var{integer}
+For lossy, this is a quality/speed tradeoff. Higher values give better quality
+for a given size at the cost of increased encoding time. For lossless, this is
+a size/speed tradeoff. Higher values give smaller size at the cost of increased
+encoding time. More specifically, it controls the number of extra algorithms
+and compression tools used, and varies the combination of these tools. This
+maps to the @var{method} option in libwebp. The valid range is 0 to 6.
+Default is 4.
+
+@item -qscale @var{float}
+For lossy encoding, this controls image quality, 0 to 100. For lossless
+encoding, this controls the effort and time spent at compressing more. The
+default value is 75. Note that for usage via libavcodec, this option is called
+@var{global_quality} and must be multiplied by @var{FF_QP2LAMBDA}.
+
+@item -preset @var{type}
+Configuration preset. This does some automatic settings based on the general
+type of the image.
+@table @option
+@item none
+Do not use a preset.
+@item default
+Use the encoder default.
+@item picture
+Digital picture, like portrait, inner shot
+@item photo
+Outdoor photograph, with natural lighting
+@item drawing
+Hand or line drawing, with high-contrast details
+@item icon
+Small-sized colorful images
+@item text
+Text-like
+@end table
+
+@end table
+
+@section libx264, libx264rgb
 
 x264 H.264/MPEG-4 AVC encoder wrapper.
 
@@ -1075,12 +1525,22 @@ for detail retention (adaptive quantization, psy-RD, psy-trellis).
 Many libx264 encoder options are mapped to FFmpeg global codec
 options, while unique encoder options are provided through private
 options. Additionally the @option{x264opts} and @option{x264-params}
-private options allows to pass a list of key=value tuples as accepted
+private options allows one to pass a list of key=value tuples as accepted
 by the libx264 @code{x264_param_parse} function.
 
 The x264 project website is at
 @url{http://www.videolan.org/developers/x264.html}.
 
+The libx264rgb encoder is the same as libx264, except it accepts packed RGB
+pixel formats as input instead of YUV.
+
+@subsection Supported Pixel Formats
+
+x264 supports 8- to 10-bit color spaces. The exact bit depth is controlled at
+x264's configure time. FFmpeg only supports one bit depth in one particular
+build. In other words, it is not possible to build one FFmpeg with multiple
+versions of x264 with different bit depths.
+
 @subsection Options
 
 The following options are supported by the libx264 wrapper. The
@@ -1393,7 +1853,7 @@ Override the x264 configuration using a :-separated list of key=value
 parameters.
 
 This option is functionally the same as the @option{x264opts}, but is
-duplicated for compability with the Libav fork.
+duplicated for compatibility with the Libav fork.
 
 For example to specify libx264 encoding options with @command{ffmpeg}:
 @example
@@ -1587,7 +2047,7 @@ Set physical density of pixels, in dots per meter, unset by default
 Apple ProRes encoder.
 
 FFmpeg contains 2 ProRes encoders, the prores-aw and prores-ks encoder.
-The used encoder can be choosen with the @code{-vcodec} option.
+The used encoder can be chosen with the @code{-vcodec} option.
 
 @subsection Private Options for prores-ks
 

doc/examples/Makefile

@@ -11,19 +11,22 @@ CFLAGS += -Wall -g
 CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
 LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
 
-EXAMPLES=       decoding_encoding                  \
-                demuxing                           \
+EXAMPLES=       avio_reading                       \
+                avcodec                            \
+                demuxing_decoding                  \
                 filtering_video                    \
                 filtering_audio                    \
                 metadata                           \
                 muxing                             \
+                remuxing                           \
                 resampling_audio                   \
                 scaling_video                      \
+                transcode_aac                      \
 
 OBJS=$(addsuffix .o,$(EXAMPLES))
 
 # the following examples make explicit use of the math library
-decoding_encoding: LDLIBS += -lm
+avcodec:           LDLIBS += -lm
 muxing:            LDLIBS += -lm
 resampling_audio:  LDLIBS += -lm
 

doc/examples/README

@@ -5,14 +5,19 @@ Both following use cases rely on pkg-config and make, thus make sure
 that you have them installed and working on your system.
 
 
-1) Build the installed examples in a generic read/write user directory
+Method 1: build the installed examples in a generic read/write user directory
 
 Copy to a read/write user directory and just use "make", it will link
 to the libraries on your system, assuming the PKG_CONFIG_PATH is
 correctly configured.
 
-2) Build the examples in-tree
+Method 2: build the examples in-tree
 
 Assuming you are in the source FFmpeg checkout directory, you need to build
-FFmpeg (no need to make install in any prefix). Then you can go into
-doc/examples and run a command such as PKG_CONFIG_PATH=pc-uninstalled make.
+FFmpeg (no need to make install in any prefix). Then just run "make examples".
+This will build the examples using the FFmpeg build system. You can clean those
+examples using "make examplesclean"
+
+If you want to try the dedicated Makefile examples (to emulate the first
+method), go into doc/examples and run a command such as
+PKG_CONFIG_PATH=pc-uninstalled make.

doc/examples/avcodec.c

@@ -24,10 +24,10 @@
  * @file
  * libavcodec API use example.
  *
+ * @example avcodec.c
  * Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),
  * not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the
  * format handling
- * @example doc/examples/decoding_encoding.c
  */
 
 #include <math.h>
@@ -156,7 +156,7 @@ static void audio_encode_example(const char *filename)
     }
 
     /* frame containing input raw audio */
-    frame = avcodec_alloc_frame();
+    frame = av_frame_alloc();
     if (!frame) {
         fprintf(stderr, "Could not allocate audio frame\n");
         exit(1);
@@ -170,6 +170,10 @@ static void audio_encode_example(const char *filename)
      * we calculate the size of the samples buffer in bytes */
     buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
                                              c->sample_fmt, 0);
+    if (buffer_size < 0) {
+        fprintf(stderr, "Could not get sample buffer size\n");
+        exit(1);
+    }
     samples = av_malloc(buffer_size);
     if (!samples) {
         fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
@@ -187,7 +191,7 @@ static void audio_encode_example(const char *filename)
     /* encode a single tone sound */
     t = 0;
     tincr = 2 * M_PI * 440.0 / c->sample_rate;
-    for(i=0;i<200;i++) {
+    for (i = 0; i < 200; i++) {
         av_init_packet(&pkt);
         pkt.data = NULL; // packet data will be allocated by the encoder
         pkt.size = 0;
@@ -227,7 +231,7 @@ static void audio_encode_example(const char *filename)
     fclose(f);
 
     av_freep(&samples);
-    avcodec_free_frame(&frame);
+    av_frame_free(&frame);
     avcodec_close(c);
     av_free(c);
 }
@@ -287,12 +291,11 @@ static void audio_decode_example(const char *outfilename, const char *filename)
         int got_frame = 0;
 
         if (!decoded_frame) {
-            if (!(decoded_frame = avcodec_alloc_frame())) {
+            if (!(decoded_frame = av_frame_alloc())) {
                 fprintf(stderr, "Could not allocate audio frame\n");
                 exit(1);
             }
-        } else
-            avcodec_get_frame_defaults(decoded_frame);
+        }
 
         len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
         if (len < 0) {
@@ -304,6 +307,11 @@ static void audio_decode_example(const char *outfilename, const char *filename)
             int data_size = av_samples_get_buffer_size(NULL, c->channels,
                                                        decoded_frame->nb_samples,
                                                        c->sample_fmt, 1);
+            if (data_size < 0) {
+                /* This should not occur, checking just for paranoia */
+                fprintf(stderr, "Failed to calculate data size\n");
+                exit(1);
+            }
             fwrite(decoded_frame->data[0], 1, data_size, outfile);
         }
         avpkt.size -= len;
@@ -329,7 +337,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
 
     avcodec_close(c);
     av_free(c);
-    avcodec_free_frame(&decoded_frame);
+    av_frame_free(&decoded_frame);
 }
 
 /*
@@ -366,12 +374,12 @@ static void video_encode_example(const char *filename, int codec_id)
     c->width = 352;
     c->height = 288;
     /* frames per second */
-    c->time_base= (AVRational){1,25};
+    c->time_base = (AVRational){1,25};
     c->gop_size = 10; /* emit one intra frame every ten frames */
-    c->max_b_frames=1;
+    c->max_b_frames = 1;
     c->pix_fmt = AV_PIX_FMT_YUV420P;
 
-    if(codec_id == AV_CODEC_ID_H264)
+    if (codec_id == AV_CODEC_ID_H264)
         av_opt_set(c->priv_data, "preset", "slow", 0);
 
     /* open it */
@@ -386,7 +394,7 @@ static void video_encode_example(const char *filename, int codec_id)
         exit(1);
     }
 
-    frame = avcodec_alloc_frame();
+    frame = av_frame_alloc();
     if (!frame) {
         fprintf(stderr, "Could not allocate video frame\n");
         exit(1);
@@ -405,7 +413,7 @@ static void video_encode_example(const char *filename, int codec_id)
     }
 
     /* encode 1 second of video */
-    for(i=0;i<25;i++) {
+    for (i = 0; i < 25; i++) {
         av_init_packet(&pkt);
         pkt.data = NULL;    // packet data will be allocated by the encoder
         pkt.size = 0;
@@ -413,15 +421,15 @@ static void video_encode_example(const char *filename, int codec_id)
         fflush(stdout);
         /* prepare a dummy image */
         /* Y */
-        for(y=0;y<c->height;y++) {
-            for(x=0;x<c->width;x++) {
+        for (y = 0; y < c->height; y++) {
+            for (x = 0; x < c->width; x++) {
                 frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
             }
         }
 
         /* Cb and Cr */
-        for(y=0;y<c->height/2;y++) {
-            for(x=0;x<c->width/2;x++) {
+        for (y = 0; y < c->height/2; y++) {
+            for (x = 0; x < c->width/2; x++) {
                 frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
                 frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
             }
@@ -467,7 +475,7 @@ static void video_encode_example(const char *filename, int codec_id)
     avcodec_close(c);
     av_free(c);
     av_freep(&frame->data[0]);
-    avcodec_free_frame(&frame);
+    av_frame_free(&frame);
     printf("\n");
 }
 
@@ -481,10 +489,10 @@ static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
     FILE *f;
     int i;
 
-    f=fopen(filename,"w");
-    fprintf(f,"P5\n%d %d\n%d\n",xsize,ysize,255);
-    for(i=0;i<ysize;i++)
-        fwrite(buf + i * wrap,1,xsize,f);
+    f = fopen(filename,"w");
+    fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
+    for (i = 0; i < ysize; i++)
+        fwrite(buf + i * wrap, 1, xsize, f);
     fclose(f);
 }
 
@@ -565,14 +573,14 @@ static void video_decode_example(const char *outfilename, const char *filename)
         exit(1);
     }
 
-    frame = avcodec_alloc_frame();
+    frame = av_frame_alloc();
     if (!frame) {
         fprintf(stderr, "Could not allocate video frame\n");
         exit(1);
     }
 
     frame_count = 0;
-    for(;;) {
+    for (;;) {
         avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
         if (avpkt.size == 0)
             break;
@@ -609,7 +617,7 @@ static void video_decode_example(const char *outfilename, const char *filename)
 
     avcodec_close(c);
     av_free(c);
-    avcodec_free_frame(&frame);
+    av_frame_free(&frame);
     printf("\n");
 }
 

doc/examples/avio_reading.c

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

doc/examples/demuxing_decoding.c

@@ -22,11 +22,11 @@
 
 /**
  * @file
- * libavformat demuxing API use example.
+ * Demuxing and decoding example.
  *
  * Show how to use the libavformat and libavcodec API to demux and
  * decode audio and video data.
- * @example doc/examples/demuxing.c
+ * @example demuxing_decoding.c
  */
 
 #include <libavutil/imgutils.h>
@@ -53,16 +53,30 @@ static AVPacket pkt;
 static int video_frame_count = 0;
 static int audio_frame_count = 0;
 
+/* The different ways of decoding and managing data memory. You are not
+ * supposed to support all the modes in your application but pick the one most
+ * appropriate to your needs. Look for the use of api_mode in this example to
+ * see what are the differences of API usage between them */
+enum {
+    API_MODE_OLD                  = 0, /* old method, deprecated */
+    API_MODE_NEW_API_REF_COUNT    = 1, /* new method, using the frame reference counting */
+    API_MODE_NEW_API_NO_REF_COUNT = 2, /* new method, without reference counting */
+};
+
+static int api_mode = API_MODE_OLD;
+
 static int decode_packet(int *got_frame, int cached)
 {
     int ret = 0;
     int decoded = pkt.size;
 
+    *got_frame = 0;
+
     if (pkt.stream_index == video_stream_idx) {
         /* decode video frame */
         ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
         if (ret < 0) {
-            fprintf(stderr, "Error decoding video frame\n");
+            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
             return ret;
         }
 
@@ -85,7 +99,7 @@ static int decode_packet(int *got_frame, int cached)
         /* decode audio frame */
         ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
         if (ret < 0) {
-            fprintf(stderr, "Error decoding audio frame\n");
+            fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret));
             return ret;
         }
         /* Some audio decoders decode only part of the packet, and have to be
@@ -113,6 +127,11 @@ static int decode_packet(int *got_frame, int cached)
         }
     }
 
+    /* If we use the new API with reference counting, we own the data and need
+     * to de-reference it when we don't use it anymore */
+    if (*got_frame && api_mode == API_MODE_NEW_API_REF_COUNT)
+        av_frame_unref(frame);
+
     return decoded;
 }
 
@@ -123,6 +142,7 @@ static int open_codec_context(int *stream_idx,
     AVStream *st;
     AVCodecContext *dec_ctx = NULL;
     AVCodec *dec = NULL;
+    AVDictionary *opts = NULL;
 
     ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
     if (ret < 0) {
@@ -139,10 +159,13 @@ static int open_codec_context(int *stream_idx,
         if (!dec) {
             fprintf(stderr, "Failed to find %s codec\n",
                     av_get_media_type_string(type));
-            return ret;
+            return AVERROR(EINVAL);
         }
 
-        if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
+        /* Init the decoders, with or without reference counting */
+        if (api_mode == API_MODE_NEW_API_REF_COUNT)
+            av_dict_set(&opts, "refcounted_frames", "1", 0);
+        if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
             fprintf(stderr, "Failed to open %s codec\n",
                     av_get_media_type_string(type));
             return ret;
@@ -185,15 +208,31 @@ int main (int argc, char **argv)
 {
     int ret = 0, got_frame;
 
-    if (argc != 4) {
-        fprintf(stderr, "usage: %s input_file video_output_file audio_output_file\n"
+    if (argc != 4 && argc != 5) {
+        fprintf(stderr, "usage: %s [-refcount=<old|new_norefcount|new_refcount>] "
+                "input_file video_output_file audio_output_file\n"
                 "API example program to show how to read frames from an input file.\n"
                 "This program reads frames from a file, decodes them, and writes decoded\n"
                 "video frames to a rawvideo file named video_output_file, and decoded\n"
-                "audio frames to a rawaudio file named audio_output_file.\n"
+                "audio frames to a rawaudio file named audio_output_file.\n\n"
+                "If the -refcount option is specified, the program use the\n"
+                "reference counting frame system which allows keeping a copy of\n"
+                "the data for longer than one decode call. If unset, it's using\n"
+                "the classic old method.\n"
                 "\n", argv[0]);
         exit(1);
     }
+    if (argc == 5) {
+        const char *mode = argv[1] + strlen("-refcount=");
+        if      (!strcmp(mode, "old"))            api_mode = API_MODE_OLD;
+        else if (!strcmp(mode, "new_norefcount")) api_mode = API_MODE_NEW_API_NO_REF_COUNT;
+        else if (!strcmp(mode, "new_refcount"))   api_mode = API_MODE_NEW_API_REF_COUNT;
+        else {
+            fprintf(stderr, "unknow mode '%s'\n", mode);
+            exit(1);
+        }
+        argv++;
+    }
     src_filename = argv[1];
     video_dst_filename = argv[2];
     audio_dst_filename = argv[3];
@@ -255,7 +294,12 @@ int main (int argc, char **argv)
         goto end;
     }
 
-    frame = avcodec_alloc_frame();
+    /* When using the new API, you need to use the libavutil/frame.h API, while
+     * the classic frame management is available in libavcodec */
+    if (api_mode == API_MODE_OLD)
+        frame = avcodec_alloc_frame();
+    else
+        frame = av_frame_alloc();
     if (!frame) {
         fprintf(stderr, "Could not allocate frame\n");
         ret = AVERROR(ENOMEM);
@@ -325,16 +369,17 @@ int main (int argc, char **argv)
     }
 
 end:
-    if (video_dec_ctx)
-        avcodec_close(video_dec_ctx);
-    if (audio_dec_ctx)
-        avcodec_close(audio_dec_ctx);
+    avcodec_close(video_dec_ctx);
+    avcodec_close(audio_dec_ctx);
     avformat_close_input(&fmt_ctx);
     if (video_dst_file)
         fclose(video_dst_file);
     if (audio_dst_file)
         fclose(audio_dst_file);
-    av_free(frame);
+    if (api_mode == API_MODE_OLD)
+        avcodec_free_frame(&frame);
+    else
+        av_frame_free(&frame);
     av_free(video_dst_data[0]);
 
     return ret < 0;

doc/examples/filter_audio.c

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

doc/examples/filtering_audio.c

@@ -25,7 +25,7 @@
 /**
  * @file
  * API example for audio decoding and filtering
- * @example doc/examples/filtering_audio.c
+ * @example filtering_audio.c
  */
 
 #include <unistd.h>
@@ -85,7 +85,7 @@ static int open_input_file(const char *filename)
 static int init_filters(const char *filters_descr)
 {
     char args[512];
-    int ret;
+    int ret = 0;
     AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
     AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
     AVFilterInOut *outputs = avfilter_inout_alloc();
@@ -97,6 +97,10 @@ static int init_filters(const char *filters_descr)
     AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
 
     filter_graph = avfilter_graph_alloc();
+    if (!outputs || !inputs || !filter_graph) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
 
     /* buffer audio source: the decoded frames from the decoder will be inserted here. */
     if (!dec_ctx->channel_layout)
@@ -109,7 +113,7 @@ static int init_filters(const char *filters_descr)
                                        args, NULL, filter_graph);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
-        return ret;
+        goto end;
     }
 
     /* buffer audio sink: to terminate the filter chain. */
@@ -117,28 +121,28 @@ static int init_filters(const char *filters_descr)
                                        NULL, NULL, filter_graph);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
-        return ret;
+        goto end;
     }
 
     ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
                               AV_OPT_SEARCH_CHILDREN);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
-        return ret;
+        goto end;
     }
 
     ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
                               AV_OPT_SEARCH_CHILDREN);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
-        return ret;
+        goto end;
     }
 
     ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
                               AV_OPT_SEARCH_CHILDREN);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
-        return ret;
+        goto end;
     }
 
     /* Endpoints for the filter graph. */
@@ -153,11 +157,11 @@ static int init_filters(const char *filters_descr)
     inputs->next       = NULL;
 
     if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
-                                    &inputs, &outputs, NULL)) < 0)
-        return ret;
+                                        &inputs, &outputs, NULL)) < 0)
+        goto end;
 
     if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        return ret;
+        goto end;
 
     /* Print summary of the sink buffer
      * Note: args buffer is reused to store channel layout string */
@@ -168,7 +172,11 @@ static int init_filters(const char *filters_descr)
            (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
            args);
 
-    return 0;
+end:
+    avfilter_inout_free(&inputs);
+    avfilter_inout_free(&outputs);
+
+    return ret;
 }
 
 static void print_frame(const AVFrame *frame)
@@ -188,7 +196,7 @@ static void print_frame(const AVFrame *frame)
 int main(int argc, char **argv)
 {
     int ret;
-    AVPacket packet;
+    AVPacket packet0, packet;
     AVFrame *frame = av_frame_alloc();
     AVFrame *filt_frame = av_frame_alloc();
     int got_frame;
@@ -202,7 +210,6 @@ int main(int argc, char **argv)
         exit(1);
     }
 
-    avcodec_register_all();
     av_register_all();
     avfilter_register_all();
 
@@ -212,18 +219,24 @@ int main(int argc, char **argv)
         goto end;
 
     /* read all packets */
+    packet0.data = NULL;
+    packet.data = NULL;
     while (1) {
-        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
-            break;
+        if (!packet0.data) {
+            if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
+                break;
+            packet0 = packet;
+        }
 
         if (packet.stream_index == audio_stream_index) {
-            avcodec_get_frame_defaults(frame);
             got_frame = 0;
             ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
             if (ret < 0) {
                 av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
                 continue;
             }
+            packet.size -= ret;
+            packet.data += ret;
 
             if (got_frame) {
                 /* push the audio data from decoded frame into the filtergraph */
@@ -235,29 +248,31 @@ int main(int argc, char **argv)
                 /* pull filtered audio from the filtergraph */
                 while (1) {
                     ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
-                    if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
+                    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
                         break;
-                    if(ret < 0)
+                    if (ret < 0)
                         goto end;
                     print_frame(filt_frame);
                     av_frame_unref(filt_frame);
                 }
             }
+
+            if (packet.size <= 0)
+                av_free_packet(&packet0);
+        } else {
+            /* discard non-wanted packets */
+            av_free_packet(&packet0);
         }
-        av_free_packet(&packet);
     }
 end:
     avfilter_graph_free(&filter_graph);
-    if (dec_ctx)
-        avcodec_close(dec_ctx);
+    avcodec_close(dec_ctx);
     avformat_close_input(&fmt_ctx);
     av_frame_free(&frame);
     av_frame_free(&filt_frame);
 
     if (ret < 0 && ret != AVERROR_EOF) {
-        char buf[1024];
-        av_strerror(ret, buf, sizeof(buf));
-        fprintf(stderr, "Error occurred: %s\n", buf);
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
         exit(1);
     }
 

doc/examples/filtering_video.c

@@ -24,7 +24,7 @@
 /**
  * @file
  * API example for decoding and filtering
- * @example doc/examples/filtering_video.c
+ * @example filtering_video.c
  */
 
 #define _XOPEN_SOURCE 600 /* for usleep */
@@ -36,6 +36,7 @@
 #include <libavfilter/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
+#include <libavutil/opt.h>
 
 const char *filter_descr = "scale=78:24";
 
@@ -70,6 +71,7 @@ static int open_input_file(const char *filename)
     }
     video_stream_index = ret;
     dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
+    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
 
     /* init the video decoder */
     if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -83,15 +85,18 @@ static int open_input_file(const char *filename)
 static int init_filters(const char *filters_descr)
 {
     char args[512];
-    int ret;
+    int ret = 0;
     AVFilter *buffersrc  = avfilter_get_by_name("buffer");
     AVFilter *buffersink = avfilter_get_by_name("buffersink");
     AVFilterInOut *outputs = avfilter_inout_alloc();
     AVFilterInOut *inputs  = avfilter_inout_alloc();
     enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
-    AVBufferSinkParams *buffersink_params;
 
     filter_graph = avfilter_graph_alloc();
+    if (!outputs || !inputs || !filter_graph) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
 
     /* buffer video source: the decoded frames from the decoder will be inserted here. */
     snprintf(args, sizeof(args),
@@ -104,18 +109,22 @@ static int init_filters(const char *filters_descr)
                                        args, NULL, filter_graph);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
-        return ret;
+        goto end;
     }
 
     /* buffer video sink: to terminate the filter chain. */
-    buffersink_params = av_buffersink_params_alloc();
-    buffersink_params->pixel_fmts = pix_fmts;
     ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
-                                       NULL, buffersink_params, filter_graph);
-    av_free(buffersink_params);
+                                       NULL, NULL, filter_graph);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
-        return ret;
+        goto end;
+    }
+
+    ret = av_opt_set_int_list(buffersink_ctx, "pix_fmts", pix_fmts,
+                              AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
+        goto end;
     }
 
     /* Endpoints for the filter graph. */
@@ -131,11 +140,16 @@ static int init_filters(const char *filters_descr)
 
     if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
                                     &inputs, &outputs, NULL)) < 0)
-        return ret;
+        goto end;
 
     if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        return ret;
-    return 0;
+        goto end;
+
+end:
+    avfilter_inout_free(&inputs);
+    avfilter_inout_free(&outputs);
+
+    return ret;
 }
 
 static void display_frame(const AVFrame *frame, AVRational time_base)
@@ -186,7 +200,6 @@ int main(int argc, char **argv)
         exit(1);
     }
 
-    avcodec_register_all();
     av_register_all();
     avfilter_register_all();
 
@@ -201,7 +214,6 @@ int main(int argc, char **argv)
             break;
 
         if (packet.stream_index == video_stream_index) {
-            avcodec_get_frame_defaults(frame);
             got_frame = 0;
             ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet);
             if (ret < 0) {
@@ -228,22 +240,20 @@ int main(int argc, char **argv)
                     display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
                     av_frame_unref(filt_frame);
                 }
+                av_frame_unref(frame);
             }
         }
         av_free_packet(&packet);
     }
 end:
     avfilter_graph_free(&filter_graph);
-    if (dec_ctx)
-        avcodec_close(dec_ctx);
+    avcodec_close(dec_ctx);
     avformat_close_input(&fmt_ctx);
     av_frame_free(&frame);
     av_frame_free(&filt_frame);
 
     if (ret < 0 && ret != AVERROR_EOF) {
-        char buf[1024];
-        av_strerror(ret, buf, sizeof(buf));
-        fprintf(stderr, "Error occurred: %s\n", buf);
+        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
         exit(1);
     }
 

doc/examples/metadata.c

@@ -23,7 +23,7 @@
 /**
  * @file
  * Shows how the metadata API can be used in application programs.
- * @example doc/examples/metadata.c
+ * @example metadata.c
  */
 
 #include <stdio.h>

doc/examples/muxing.c

@@ -24,9 +24,9 @@
  * @file
  * libavformat API example.
  *
- * Output a media file in any supported libavformat format.
- * The default codecs are used.
- * @example doc/examples/muxing.c
+ * Output a media file in any supported libavformat format. The default
+ * codecs are used.
+ * @example muxing.c
  */
 
 #include <stdlib.h>
@@ -36,18 +36,43 @@
 
 #include <libavutil/opt.h>
 #include <libavutil/mathematics.h>
+#include <libavutil/timestamp.h>
 #include <libavformat/avformat.h>
 #include <libswscale/swscale.h>
 #include <libswresample/swresample.h>
 
-/* 5 seconds stream duration */
-#define STREAM_DURATION   200.0
+static int audio_is_eof, video_is_eof;
+
+#define STREAM_DURATION   10.0
 #define STREAM_FRAME_RATE 25 /* 25 images/s */
-#define STREAM_NB_FRAMES  ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
 #define STREAM_PIX_FMT    AV_PIX_FMT_YUV420P /* default pix_fmt */
 
 static int sws_flags = SWS_BICUBIC;
 
+static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
+{
+    AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
+
+    printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
+           av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
+           av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
+           av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
+           pkt->stream_index);
+}
+
+static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
+{
+    /* rescale output packet timestamp values from codec to stream timebase */
+    pkt->pts = av_rescale_q_rnd(pkt->pts, *time_base, st->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
+    pkt->dts = av_rescale_q_rnd(pkt->dts, *time_base, st->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
+    pkt->duration = av_rescale_q(pkt->duration, *time_base, st->time_base);
+    pkt->stream_index = st->index;
+
+    /* Write the compressed frame to the media file. */
+    log_packet(fmt_ctx, pkt);
+    return av_interleaved_write_frame(fmt_ctx, pkt);
+}
+
 /* Add an output stream. */
 static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
                             enum AVCodecID codec_id)
@@ -73,7 +98,8 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
 
     switch ((*codec)->type) {
     case AVMEDIA_TYPE_AUDIO:
-        c->sample_fmt  = AV_SAMPLE_FMT_FLTP;
+        c->sample_fmt  = (*codec)->sample_fmts ?
+            (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
         c->bit_rate    = 64000;
         c->sample_rate = 44100;
         c->channels    = 2;
@@ -122,6 +148,7 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
 
 static float t, tincr, tincr2;
 
+AVFrame *audio_frame;
 static uint8_t **src_samples_data;
 static int       src_samples_linesize;
 static int       src_nb_samples;
@@ -130,6 +157,7 @@ static int max_dst_nb_samples;
 uint8_t **dst_samples_data;
 int       dst_samples_linesize;
 int       dst_samples_size;
+int samples_count;
 
 struct SwrContext *swr_ctx = NULL;
 
@@ -140,6 +168,13 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
 
     c = st->codec;
 
+    /* allocate and init a re-usable frame */
+    audio_frame = av_frame_alloc();
+    if (!audio_frame) {
+        fprintf(stderr, "Could not allocate audio frame\n");
+        exit(1);
+    }
+
     /* open it */
     ret = avcodec_open2(c, codec, NULL);
     if (ret < 0) {
@@ -157,12 +192,17 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
         10000 : c->frame_size;
 
     ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels,
-                                             src_nb_samples, c->sample_fmt, 0);
+                                             src_nb_samples, AV_SAMPLE_FMT_S16, 0);
     if (ret < 0) {
         fprintf(stderr, "Could not allocate source samples\n");
         exit(1);
     }
 
+    /* compute the number of converted samples: buffering is avoided
+     * ensuring that the output buffer will contain at least all the
+     * converted input samples */
+    max_dst_nb_samples = src_nb_samples;
+
     /* create resampler context */
     if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
         swr_ctx = swr_alloc();
@@ -184,17 +224,15 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
             fprintf(stderr, "Failed to initialize the resampling context\n");
             exit(1);
         }
-    }
 
-    /* compute the number of converted samples: buffering is avoided
-     * ensuring that the output buffer will contain at least all the
-     * converted input samples */
-    max_dst_nb_samples = src_nb_samples;
-    ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels,
-                                             max_dst_nb_samples, c->sample_fmt, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate destination samples\n");
-        exit(1);
+        ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels,
+                                                 max_dst_nb_samples, c->sample_fmt, 0);
+        if (ret < 0) {
+            fprintf(stderr, "Could not allocate destination samples\n");
+            exit(1);
+        }
+    } else {
+        dst_samples_data = src_samples_data;
     }
     dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples,
                                                   c->sample_fmt, 0);
@@ -217,77 +255,83 @@ static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
     }
 }
 
-static void write_audio_frame(AVFormatContext *oc, AVStream *st)
+static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush)
 {
     AVCodecContext *c;
     AVPacket pkt = { 0 }; // data and size must be 0;
-    AVFrame *frame = avcodec_alloc_frame();
     int got_packet, ret, dst_nb_samples;
 
     av_init_packet(&pkt);
     c = st->codec;
 
-    get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
+    if (!flush) {
+        get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
 
-    /* convert samples from native format to destination codec format, using the resampler */
-    if (swr_ctx) {
-        /* compute destination number of samples */
-        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
-                                        c->sample_rate, c->sample_rate, AV_ROUND_UP);
-        if (dst_nb_samples > max_dst_nb_samples) {
-            av_free(dst_samples_data[0]);
-            ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
-                                   dst_nb_samples, c->sample_fmt, 0);
-            if (ret < 0)
+        /* convert samples from native format to destination codec format, using the resampler */
+        if (swr_ctx) {
+            /* compute destination number of samples */
+            dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
+                                            c->sample_rate, c->sample_rate, AV_ROUND_UP);
+            if (dst_nb_samples > max_dst_nb_samples) {
+                av_free(dst_samples_data[0]);
+                ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
+                                       dst_nb_samples, c->sample_fmt, 0);
+                if (ret < 0)
+                    exit(1);
+                max_dst_nb_samples = dst_nb_samples;
+                dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
+                                                              c->sample_fmt, 0);
+            }
+
+            /* convert to destination format */
+            ret = swr_convert(swr_ctx,
+                              dst_samples_data, dst_nb_samples,
+                              (const uint8_t **)src_samples_data, src_nb_samples);
+            if (ret < 0) {
+                fprintf(stderr, "Error while converting\n");
                 exit(1);
-            max_dst_nb_samples = dst_nb_samples;
-            dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
-                                                          c->sample_fmt, 0);
+            }
+        } else {
+            dst_nb_samples = src_nb_samples;
         }
 
-        /* convert to destination format */
-        ret = swr_convert(swr_ctx,
-                          dst_samples_data, dst_nb_samples,
-                          (const uint8_t **)src_samples_data, src_nb_samples);
-        if (ret < 0) {
-            fprintf(stderr, "Error while converting\n");
-            exit(1);
-        }
-    } else {
-        dst_samples_data[0] = src_samples_data[0];
-        dst_nb_samples = src_nb_samples;
+        audio_frame->nb_samples = dst_nb_samples;
+        audio_frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
+        avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt,
+                                 dst_samples_data[0], dst_samples_size, 0);
+        samples_count += dst_nb_samples;
     }
 
-    frame->nb_samples = dst_nb_samples;
-    avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
-                             dst_samples_data[0], dst_samples_size, 0);
-
-    ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
+    ret = avcodec_encode_audio2(c, &pkt, flush ? NULL : audio_frame, &got_packet);
     if (ret < 0) {
         fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
         exit(1);
     }
 
-    if (!got_packet)
+    if (!got_packet) {
+        if (flush)
+            audio_is_eof = 1;
         return;
+    }
 
-    pkt.stream_index = st->index;
-
-    /* Write the compressed frame to the media file. */
-    ret = av_interleaved_write_frame(oc, &pkt);
-    if (ret != 0) {
+    ret = write_frame(oc, &c->time_base, st, &pkt);
+    if (ret < 0) {
         fprintf(stderr, "Error while writing audio frame: %s\n",
                 av_err2str(ret));
         exit(1);
     }
-    avcodec_free_frame(&frame);
 }
 
 static void close_audio(AVFormatContext *oc, AVStream *st)
 {
     avcodec_close(st->codec);
+    if (dst_samples_data != src_samples_data) {
+        av_free(dst_samples_data[0]);
+        av_free(dst_samples_data);
+    }
     av_free(src_samples_data[0]);
-    av_free(dst_samples_data[0]);
+    av_free(src_samples_data);
+    av_frame_free(&audio_frame);
 }
 
 /**************************************************************/
@@ -310,11 +354,14 @@ static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
     }
 
     /* allocate and init a re-usable frame */
-    frame = avcodec_alloc_frame();
+    frame = av_frame_alloc();
     if (!frame) {
         fprintf(stderr, "Could not allocate video frame\n");
         exit(1);
     }
+    frame->format = c->pix_fmt;
+    frame->width = c->width;
+    frame->height = c->height;
 
     /* Allocate the encoded raw picture. */
     ret = avpicture_alloc(&dst_picture, c->pix_fmt, c->width, c->height);
@@ -361,17 +408,13 @@ static void fill_yuv_image(AVPicture *pict, int frame_index,
     }
 }
 
-static void write_video_frame(AVFormatContext *oc, AVStream *st)
+static void write_video_frame(AVFormatContext *oc, AVStream *st, int flush)
 {
     int ret;
     static struct SwsContext *sws_ctx;
     AVCodecContext *c = st->codec;
 
-    if (frame_count >= STREAM_NB_FRAMES) {
-        /* No more frames to compress. The codec has a latency of a few
-         * frames if using B-frames, so we get the last frames by
-         * passing the same picture again. */
-    } else {
+    if (!flush) {
         if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
             /* as we only generate a YUV420P picture, we must convert it
              * to the codec pixel format if needed */
@@ -394,7 +437,7 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)
         }
     }
 
-    if (oc->oformat->flags & AVFMT_RAWPICTURE) {
+    if (oc->oformat->flags & AVFMT_RAWPICTURE && !flush) {
         /* Raw video case - directly store the picture in the packet */
         AVPacket pkt;
         av_init_packet(&pkt);
@@ -411,23 +454,24 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)
         av_init_packet(&pkt);
 
         /* encode the image */
-        ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
+        frame->pts = frame_count;
+        ret = avcodec_encode_video2(c, &pkt, flush ? NULL : frame, &got_packet);
         if (ret < 0) {
             fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
             exit(1);
         }
         /* If size is zero, it means the image was buffered. */
 
-        if (!ret && got_packet && pkt.size) {
-            pkt.stream_index = st->index;
-
-            /* Write the compressed frame to the media file. */
-            ret = av_interleaved_write_frame(oc, &pkt);
+        if (got_packet) {
+            ret = write_frame(oc, &c->time_base, st, &pkt);
         } else {
+            if (flush)
+                video_is_eof = 1;
             ret = 0;
         }
     }
-    if (ret != 0) {
+
+    if (ret < 0) {
         fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
         exit(1);
     }
@@ -439,7 +483,7 @@ static void close_video(AVFormatContext *oc, AVStream *st)
     avcodec_close(st->codec);
     av_free(src_picture.data[0]);
     av_free(dst_picture.data[0]);
-    av_free(frame);
+    av_frame_free(&frame);
 }
 
 /**************************************************************/
@@ -453,7 +497,7 @@ int main(int argc, char **argv)
     AVStream *audio_st, *video_st;
     AVCodec *audio_codec, *video_codec;
     double audio_time, video_time;
-    int ret;
+    int flush, ret;
 
     /* Initialize libavcodec, and register all codecs and formats. */
     av_register_all();
@@ -477,9 +521,9 @@ int main(int argc, char **argv)
         printf("Could not deduce output format from file extension: using MPEG.\n");
         avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
     }
-    if (!oc) {
+    if (!oc)
         return 1;
-    }
+
     fmt = oc->oformat;
 
     /* Add the audio and video streams using the default format codecs
@@ -487,12 +531,10 @@ int main(int argc, char **argv)
     video_st = NULL;
     audio_st = NULL;
 
-    if (fmt->video_codec != AV_CODEC_ID_NONE) {
+    if (fmt->video_codec != AV_CODEC_ID_NONE)
         video_st = add_stream(oc, &video_codec, fmt->video_codec);
-    }
-    if (fmt->audio_codec != AV_CODEC_ID_NONE) {
+    if (fmt->audio_codec != AV_CODEC_ID_NONE)
         audio_st = add_stream(oc, &audio_codec, fmt->audio_codec);
-    }
 
     /* Now that all the parameters are set, we can open the audio and
      * video codecs and allocate the necessary encode buffers. */
@@ -521,23 +563,23 @@ int main(int argc, char **argv)
         return 1;
     }
 
-    if (frame)
-        frame->pts = 0;
-    for (;;) {
+    flush = 0;
+    while ((video_st && !video_is_eof) || (audio_st && !audio_is_eof)) {
         /* Compute current audio and video time. */
-        audio_time = audio_st ? audio_st->pts.val * av_q2d(audio_st->time_base) : 0.0;
-        video_time = video_st ? video_st->pts.val * av_q2d(video_st->time_base) : 0.0;
+        audio_time = (audio_st && !audio_is_eof) ? audio_st->pts.val * av_q2d(audio_st->time_base) : INFINITY;
+        video_time = (video_st && !video_is_eof) ? video_st->pts.val * av_q2d(video_st->time_base) : INFINITY;
 
-        if ((!audio_st || audio_time >= STREAM_DURATION) &&
-            (!video_st || video_time >= STREAM_DURATION))
-            break;
+        if (!flush &&
+            (!audio_st || audio_time >= STREAM_DURATION) &&
+            (!video_st || video_time >= STREAM_DURATION)) {
+            flush = 1;
+        }
 
         /* write interleaved audio and video frames */
-        if (!video_st || (video_st && audio_st && audio_time < video_time)) {
-            write_audio_frame(oc, audio_st);
-        } else {
-            write_video_frame(oc, video_st);
-            frame->pts += av_rescale_q(1, video_st->codec->time_base, video_st->time_base);
+        if (audio_st && !audio_is_eof && audio_time <= video_time) {
+            write_audio_frame(oc, audio_st, flush);
+        } else if (video_st && !video_is_eof && video_time < audio_time) {
+            write_video_frame(oc, video_st, flush);
         }
     }
 

doc/examples/remuxing.c

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

doc/examples/resampling_audio.c

@@ -21,7 +21,7 @@
  */
 
 /**
- * @example doc/examples/resampling_audio.c
+ * @example resampling_audio.c
  * libswresample API use example.
  */
 
@@ -62,7 +62,7 @@ static int get_format_from_sample_fmt(const char **fmt,
 /**
  * Fill dst buffer with nb_samples, generated starting from t.
  */
-void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
+static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
 {
     int i, j;
     double tincr = 1.0 / sample_rate, *dstp = dst;
@@ -184,6 +184,10 @@ int main(int argc, char **argv)
         }
         dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
                                                  ret, dst_sample_fmt, 1);
+        if (dst_bufsize < 0) {
+            fprintf(stderr, "Could not get sample buffer size\n");
+            goto end;
+        }
         printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
         fwrite(dst_data[0], 1, dst_bufsize, dst_file);
     } while (t < 10);

doc/examples/scaling_video.c

@@ -23,7 +23,7 @@
 /**
  * @file
  * libswscale API use example.
- * @example doc/examples/scaling_video.c
+ * @example scaling_video.c
  */
 
 #include <libavutil/imgutils.h>

doc/examples/transcode_aac.c

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

doc/faq.texi

@@ -368,26 +368,6 @@ ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
 rm temp[12].[av] all.[av]
 @end example
 
-@section -profile option fails when encoding H.264 video with AAC audio
-
-@command{ffmpeg} prints an error like
-
-@example
-Undefined constant or missing '(' in 'baseline'
-Unable to parse option value "baseline"
-Error setting option profile to value baseline.
-@end example
-
-Short answer: write @option{-profile:v} instead of @option{-profile}.
-
-Long answer: this happens because the @option{-profile} option can apply to both
-video and audio.  Specifically the AAC encoder also defines some profiles, none
-of which are named @var{baseline}.
-
-The solution is to apply the @option{-profile} option to the video stream only
-by using @url{http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1, Stream specifiers}.
-Appending @code{:v} to it will do exactly that.
-
 @section Using @option{-f lavfi}, audio becomes mono for no apparent reason.
 
 Use @option{-dumpgraph -} to find out exactly where the channel layout is

doc/ffmpeg-resampler.texi

@@ -14,7 +14,7 @@
 
 The FFmpeg resampler provides a high-level interface to the
 libswresample library audio resampling utilities. In particular it
-allows to perform audio resampling, audio channel layout rematrixing,
+allows one to perform audio resampling, audio channel layout rematrixing,
 and convert audio format and packing layout.
 
 @c man end DESCRIPTION

doc/ffmpeg-scaler.texi

@@ -13,7 +13,7 @@
 @c man begin DESCRIPTION
 
 The FFmpeg rescaler provides a high-level interface to the libswscale
-library image conversion utilities. In particular it allows to perform
+library image conversion utilities. In particular it allows one to perform
 image rescaling and pixel format conversion.
 
 @c man end DESCRIPTION

doc/ffmpeg.texi

@@ -80,11 +80,22 @@ The transcoding process in @command{ffmpeg} for each output can be described by
 the following diagram:
 
 @example
- _______              ______________               _________              ______________            ________
-|       |            |              |             |         |            |              |          |        |
-| input |  demuxer   | encoded data |   decoder   | decoded |  encoder   | encoded data |  muxer   | output |
-| file  | ---------> | packets      |  ---------> | frames  | ---------> | packets      | -------> | file   |
-|_______|            |______________|             |_________|            |______________|          |________|
+ _______              ______________
+|       |            |              |
+| input |  demuxer   | encoded data |   decoder
+| file  | ---------> | packets      | -----+
+|_______|            |______________|      |
+                                           v
+                                       _________
+                                      |         |
+                                      | decoded |
+                                      | frames  |
+ ________             ______________  |_________|
+|        |           |              |      |
+| output | <-------- | encoded data | <----+
+| file   |   muxer   | packets      |   encoder
+|________|           |______________|
+
 
 @end example
 
@@ -112,11 +123,11 @@ the same type. In the above diagram they can be represented by simply inserting
 an additional step between decoding and encoding:
 
 @example
- _________                        __________              ______________
-|         |                      |          |            |              |
-| decoded |  simple filtergraph  | filtered |  encoder   | encoded data |
-| frames  | -------------------> | frames   | ---------> | packets      |
-|_________|                      |__________|            |______________|
+ _________               __________              ______________
+|         |  simple     |          |            |              |
+| decoded |  fltrgrph   | filtered |  encoder   | encoded data |
+| frames  | ----------> | frames   | ---------> | packets      |
+|_________|             |__________|            |______________|
 
 @end example
 
@@ -125,10 +136,10 @@ Simple filtergraphs are configured with the per-stream @option{-filter} option
 A simple filtergraph for video can look for example like this:
 
 @example
- _______        _____________        _______        _____        ________
-|       |      |             |      |       |      |     |      |        |
-| input | ---> | deinterlace | ---> | scale | ---> | fps | ---> | output |
-|_______|      |_____________|      |_______|      |_____|      |________|
+ _______        _____________        _______        ________
+|       |      |             |      |       |      |        |
+| input | ---> | deinterlace | ---> | scale | ---> | output |
+|_______|      |_____________|      |_______|      |________|
 
 @end example
 
@@ -285,23 +296,20 @@ input until the timestamps reach @var{position}.
 @var{position} may be either in seconds or in @code{hh:mm:ss[.xxx]} form.
 
 @item -itsoffset @var{offset} (@emph{input})
-Set the input time offset in seconds.
-@code{[-]hh:mm:ss[.xxx]} syntax is also supported.
-The offset is added to the timestamps of the input files.
-Specifying a positive offset means that the corresponding
-streams are delayed by @var{offset} seconds.
+Set the input time offset.
 
-@item -timestamp @var{time} (@emph{output})
+@var{offset} must be a time duration specification,
+see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+
+The offset is added to the timestamps of the input files. Specifying
+a positive offset means that the corresponding streams are delayed by
+the time duration specified in @var{offset}.
+
+@item -timestamp @var{date} (@emph{output})
 Set the recording timestamp in the container.
-The syntax for @var{time} is:
-@example
-now|([(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...])|(HHMMSS[.m...]))[Z|z])
-@end example
-If the value is "now" it takes the current time.
-Time is local time unless 'Z' or 'z' is appended, in which case it is
-interpreted as UTC.
-If the year-month-day part is not specified it takes the current
-year-month-day.
+
+@var{date} must be a time duration specification,
+see @ref{date syntax,,the Date section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
 
 @item -metadata[:metadata_specifier] @var{key}=@var{value} (@emph{output,per-metadata})
 Set a metadata key/value pair.
@@ -348,8 +356,13 @@ Stop writing to the stream after @var{framecount} frames.
 
 @item -q[:@var{stream_specifier}] @var{q} (@emph{output,per-stream})
 @itemx -qscale[:@var{stream_specifier}] @var{q} (@emph{output,per-stream})
-Use fixed quality scale (VBR). The meaning of @var{q} is
+Use fixed quality scale (VBR). The meaning of @var{q}/@var{qscale} is
 codec-dependent.
+If @var{qscale} is used without a @var{stream_specifier} then it applies only
+to the video stream, this is to maintain compatibility with previous behavior
+and as specifying the same codec specific value to 2 different codecs that is
+audio and video generally is not what is intended when no stream_specifier is
+used.
 
 @anchor{filter_option}
 @item -filter[:@var{stream_specifier}] @var{filtergraph} (@emph{output,per-stream})
@@ -503,9 +516,6 @@ prefix is ``ffmpeg2pass''. The complete file name will be
 @file{PREFIX-N.log}, where N is a number specific to the output
 stream
 
-@item -vlang @var{code}
-Set the ISO 639 language code (3 letters) of the current video stream.
-
 @item -vf @var{filtergraph} (@emph{output})
 Create the filtergraph specified by @var{filtergraph} and use it to
 filter the stream.
@@ -616,6 +626,42 @@ would be more efficient.
 @item -copyinkf[:@var{stream_specifier}] (@emph{output,per-stream})
 When doing stream copy, copy also non-key frames found at the
 beginning.
+
+@item -hwaccel[:@var{stream_specifier}] @var{hwaccel} (@emph{input,per-stream})
+Use hardware acceleration to decode the matching stream(s). The allowed values
+of @var{hwaccel} are:
+@table @option
+@item none
+Do not use any hardware acceleration (the default).
+
+@item auto
+Automatically select the hardware acceleration method.
+
+@item vdpau
+Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration.
+@end table
+
+This option has no effect if the selected hwaccel is not available or not
+supported by the chosen decoder.
+
+Note that most acceleration methods are intended for playback and will not be
+faster than software decoding on modern CPUs. Additionally, @command{ffmpeg}
+will usually need to copy the decoded frames from the GPU memory into the system
+memory, resulting in further performance loss. This option is thus mainly
+useful for testing.
+
+@item -hwaccel_device[:@var{stream_specifier}] @var{hwaccel_device} (@emph{input,per-stream})
+Select a device to use for hardware acceleration.
+
+This option only makes sense when the @option{-hwaccel} option is also
+specified. Its exact meaning depends on the specific hardware acceleration
+method chosen.
+
+@table @option
+@item vdpau
+For VDPAU, this option specifies the X11 display/screen to use. If this option
+is not specified, the value of the @var{DISPLAY} environment variable is used
+@end table
 @end table
 
 @section Audio Options
@@ -668,8 +714,6 @@ stereo but not 6 channels as 5.1. The default is to always try to guess. Use
 @section Subtitle options:
 
 @table @option
-@item -slang @var{code}
-Set the ISO 639 language code (3 letters) of the current subtitle stream.
 @item -scodec @var{codec} (@emph{input/output})
 Set the subtitle codec. This is an alias for @code{-codec:s}.
 @item -sn (@emph{output})
@@ -998,7 +1042,7 @@ ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264
 ffmpeg -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt
 @end example
 
-@item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{per-stream})
+@item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{input/output,per-stream})
 Force a tag/fourcc for matching streams.
 
 @item -timecode @var{hh}:@var{mm}:@var{ss}SEP@var{ff}
@@ -1072,12 +1116,14 @@ transcoding. Use @option{-noaccurate_seek} to disable it, which may be useful
 e.g. when copying some streams and transcoding the others.
 
 @item -override_ffserver (@emph{global})
-Overrides the input specifications from ffserver. Using this option you can
-map any input stream to ffserver and control many aspects of the encoding from
-ffmpeg. Without this option ffmpeg will transmit to ffserver what is requested by
-ffserver.
+Overrides the input specifications from @command{ffserver}. Using this
+option you can map any input stream to @command{ffserver} and control
+many aspects of the encoding from @command{ffmpeg}. Without this
+option @command{ffmpeg} will transmit to @command{ffserver} what is
+requested by @command{ffserver}.
+
 The option is intended for cases where features are needed that cannot be
-specified to ffserver but can be to ffmpeg.
+specified to @command{ffserver} but can be to @command{ffmpeg}.
 
 @end table
 

doc/ffplay.texi

@@ -188,6 +188,12 @@ Cycle program.
 @item w
 Show audio waves.
 
+@item s
+Step to the next frame.
+
+Pause if the stream is not already paused, step to the next video
+frame, and pause.
+
 @item left/right
 Seek backward/forward 10 seconds.
 
@@ -195,6 +201,8 @@ Seek backward/forward 10 seconds.
 Seek backward/forward 1 minute.
 
 @item page down/page up
+Seek to the previous/next chapter.
+or if there are no chapters
 Seek backward/forward 10 minutes.
 
 @item mouse click
@@ -206,6 +214,7 @@ Seek to percentage in file corresponding to fraction of width.
 
 @include config.texi
 @ifset config-all
+@set config-readonly
 @ifset config-avutil
 @include utils.texi
 @end ifset

doc/ffprobe.texi

@@ -197,11 +197,11 @@ The information for each single packet is printed within a dedicated
 section with name "PACKET".
 
 @item -show_frames
-Show information about each frame contained in the input multimedia
-stream.
+Show information about each frame and subtitle contained in the input
+multimedia stream.
 
 The information for each single frame is printed within a dedicated
-section with name "FRAME".
+section with name "FRAME" or "SUBTITLE".
 
 @item -show_streams
 Show information about each media stream contained in the input
@@ -337,6 +337,39 @@ A writer may accept one or more arguments, which specify the options
 to adopt. The options are specified as a list of @var{key}=@var{value}
 pairs, separated by ":".
 
+All writers support the following options:
+
+@table @option
+@item string_validation, sv
+Set string validation mode.
+
+The following values are accepted.
+@table @samp
+@item fail
+The writer will fail immediately in case an invalid string (UTF-8)
+sequence or code point is found in the input. This is especially
+useful to validate input metadata.
+
+@item ignore
+Any validation error will be ignored. This will result in possibly
+broken output, especially with the json or xml writer.
+
+@item replace
+The writer will substitute invalid UTF-8 sequences or code points with
+the string specified with the @option{string_validation_replacement}.
+@end table
+
+Default value is @samp{replace}.
+
+@item string_validation_replacement, svr
+Set replacement string to use in case @option{string_validation} is
+set to @samp{replace}.
+
+In case the option is not specified, the writer will assume the empty
+string, that is it will remove the invalid sequences from the input
+strings.
+@end table
+
 A description of the currently available writers follows.
 
 @section default
@@ -566,6 +599,7 @@ DV, GXF and AVI timecodes are available in format metadata
 
 @include config.texi
 @ifset config-all
+@set config-readonly
 @ifset config-avutil
 @include utils.texi
 @end ifset

doc/ffprobe.xsd

@@ -28,7 +28,10 @@
 
     <xsd:complexType name="framesType">
         <xsd:sequence>
-            <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
+            <xsd:choice minOccurs="0" maxOccurs="unbounded">
+                <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
+                <xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
+            </xsd:choice>
         </xsd:sequence>
     </xsd:complexType>
 
@@ -58,6 +61,8 @@
       <xsd:attribute name="pkt_pts_time"  type="xsd:float"/>
       <xsd:attribute name="pkt_dts"       type="xsd:long" />
       <xsd:attribute name="pkt_dts_time"  type="xsd:float"/>
+      <xsd:attribute name="best_effort_timestamp"      type="xsd:long" />
+      <xsd:attribute name="best_effort_timestamp_time" type="xsd:float" />
       <xsd:attribute name="pkt_duration"  type="xsd:long" />
       <xsd:attribute name="pkt_duration_time" type="xsd:float"/>
       <xsd:attribute name="pkt_pos"       type="xsd:long" />
@@ -82,6 +87,16 @@
       <xsd:attribute name="repeat_pict"            type="xsd:int"   />
     </xsd:complexType>
 
+    <xsd:complexType name="subtitleType">
+      <xsd:attribute name="media_type"         type="xsd:string" fixed="subtitle" use="required"/>
+      <xsd:attribute name="pts"                type="xsd:long" />
+      <xsd:attribute name="pts_time"           type="xsd:float"/>
+      <xsd:attribute name="format"             type="xsd:int"  />
+      <xsd:attribute name="start_display_time" type="xsd:int"  />
+      <xsd:attribute name="end_display_time"   type="xsd:int"  />
+      <xsd:attribute name="num_rects"          type="xsd:int"  />
+    </xsd:complexType>
+
     <xsd:complexType name="streamsType">
         <xsd:sequence>
             <xsd:element name="stream" type="ffprobe:streamType" minOccurs="0" maxOccurs="unbounded"/>

doc/ffserver.conf

@@ -235,7 +235,7 @@ StartSendOnKey
 
 #<Stream test.ogg>
 #Feed feed1.ffm
-#Title "Stream title"
+#Metadata title "Stream title"
 #AudioBitRate 64
 #AudioChannels 2
 #AudioSampleRate 44100
@@ -280,10 +280,10 @@ StartSendOnKey
 #<Stream file.asf>
 #File "/usr/local/httpd/htdocs/test.asf"
 #NoAudio
-#Author "Me"
-#Copyright "Super MegaCorp"
-#Title "Test stream from disk"
-#Comment "Test comment"
+#Metadata author "Me"
+#Metadata copyright "Super MegaCorp"
+#Metadata title "Test stream from disk"
+#Metadata comment "Test comment"
 #</Stream>
 
 

doc/ffserver.texi

@@ -16,11 +16,14 @@ ffserver [@var{options}]
 @chapter Description
 @c man begin DESCRIPTION
 
-@command{ffserver} is a streaming server for both audio and video. It
-supports several live feeds, streaming from files and time shifting on
-live feeds (you can seek to positions in the past on each live feed,
-provided you specify a big enough feed storage in
-@file{ffserver.conf}).
+@command{ffserver} is a streaming server for both audio and video.
+It supports several live feeds, streaming from files and time shifting
+on live feeds. You can seek to positions in the past on each live
+feed, provided you specify a big enough feed storage.
+
+@command{ffserver} is configured through a configuration file, which
+is read at startup. If not explicitly specified, it will read from
+@file{/etc/ffserver.conf}.
 
 @command{ffserver} receives prerecorded files or FFM streams from some
 @command{ffmpeg} instance as input, then streams them over
@@ -39,10 +42,126 @@ For each feed you can have different output streams in various
 formats, each one specified by a @code{<Stream>} section in the
 configuration file.
 
+@chapter Detailed description
+
+@command{ffserver} works by forwarding streams encoded by
+@command{ffmpeg}, or pre-recorded streams which are read from disk.
+
+Precisely, @command{ffserver} acts as an HTTP server, accepting POST
+requests from @command{ffmpeg} to acquire the stream to publish, and
+serving RTSP clients or HTTP clients GET requests with the stream
+media content.
+
+A feed is an @ref{FFM} stream created by @command{ffmpeg}, and sent to
+a port where @command{ffserver} is listening.
+
+Each feed is identified by a unique name, corresponding to the name
+of the resource published on @command{ffserver}, and is configured by
+a dedicated @code{Feed} section in the configuration file.
+
+The feed publish URL is given by:
+@example
+http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name}
+@end example
+
+where @var{ffserver_ip_address} is the IP address of the machine where
+@command{ffserver} is installed, @var{http_port} is the port number of
+the HTTP server (configured through the @option{Port} option), and
+@var{feed_name} is the name of the corresponding feed defined in the
+configuration file.
+
+Each feed is associated to a file which is stored on disk. This stored
+file is used to allow to send pre-recorded data to a player as fast as
+possible when new content is added in real-time to the stream.
+
+A "live-stream" or "stream" is a resource published by
+@command{ffserver}, and made accessible through the HTTP protocol to
+clients.
+
+A stream can be connected to a feed, or to a file. In the first case,
+the published stream is forwarded from the corresponding feed
+generated by a running instance of @command{ffmpeg}, in the second
+case the stream is read from a pre-recorded file.
+
+Each stream is identified by a unique name, corresponding to the name
+of the resource served by @command{ffserver}, and is configured by
+a dedicated @code{Stream} section in the configuration file.
+
+The stream access HTTP URL is given by:
+@example
+http://@var{ffserver_ip_address}:@var{http_port}/@var{stream_name}[@var{options}]
+@end example
+
+The stream access RTSP URL is given by:
+@example
+http://@var{ffserver_ip_address}:@var{rtsp_port}/@var{stream_name}[@var{options}]
+@end example
+
+@var{stream_name} is the name of the corresponding stream defined in
+the configuration file. @var{options} is a list of options specified
+after the URL which affects how the stream is served by
+@command{ffserver}. @var{http_port} and @var{rtsp_port} are the HTTP
+and RTSP ports configured with the options @var{Port} and
+@var{RTSPPort} respectively.
+
+In case the stream is associated to a feed, the encoding parameters
+must be configured in the stream configuration. They are sent to
+@command{ffmpeg} when setting up the encoding. This allows
+@command{ffserver} to define the encoding parameters used by
+the @command{ffmpeg} encoders.
+
+The @command{ffmpeg} @option{override_ffserver} commandline option
+allows one to override the encoding parameters set by the server.
+
+Multiple streams can be connected to the same feed.
+
+For example, you can have a situation described by the following
+graph:
+@example
+               _________       __________
+              |         |     |          |
+ffmpeg 1 -----| feed 1  |-----| stream 1 |
+    \         |_________|\    |__________|
+     \                    \
+      \                    \   __________
+       \                    \ |          |
+        \                    \| stream 2 |
+         \                    |__________|
+          \
+           \   _________       __________
+            \ |         |     |          |
+             \| feed 2  |-----| stream 3 |
+              |_________|     |__________|
+
+               _________       __________
+              |         |     |          |
+ffmpeg 2 -----| feed 3  |-----| stream 4 |
+              |_________|     |__________|
+
+               _________       __________
+              |         |     |          |
+              | file 1  |-----| stream 5 |
+              |_________|     |__________|
+@end example
+
+@anchor{FFM}
+@section FFM, FFM2 formats
+
+FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
+video and audio streams and encoding options, and can store a moving time segment
+of an infinite movie or a whole movie.
+
+FFM is version specific, and there is limited compatibility of FFM files
+generated by one version of ffmpeg/ffserver and another version of
+ffmpeg/ffserver. It may work but it is not guaranteed to work.
+
+FFM2 is extensible while maintaining compatibility and should work between
+differing versions of tools. FFM2 is the default.
+
 @section Status stream
 
-ffserver supports an HTTP interface which exposes the current status
-of the server.
+@command{ffserver} supports an HTTP interface which exposes the
+current status of the server.
 
 Simply point your browser to the address of the special status stream
 specified in the configuration file.
@@ -61,27 +180,8 @@ ACL allow 192.168.0.0 192.168.255.255
 then the server will post a page with the status information when
 the special stream @file{status.html} is requested.
 
-@section What can this do?
-
-When properly configured and running, you can capture video and audio in real
-time from a suitable capture card, and stream it out over the Internet to
-either Windows Media Player or RealAudio player (with some restrictions).
-
-It can also stream from files, though that is currently broken. Very often, a
-web server can be used to serve up the files just as well.
-
-It can stream prerecorded video from .ffm files, though it is somewhat tricky
-to make it work correctly.
-
 @section How do I make it work?
 
-First, build the kit. It *really* helps to have installed LAME first. Then when
-you run the ffserver ./configure, make sure that you have the
-@code{--enable-libmp3lame} flag turned on.
-
-LAME is important as it allows for streaming audio to Windows Media Player.
-Don't ask why the other audio types do not work.
-
 As a simple test, just run the following two command lines where INPUTFILE
 is some file which you can decode with ffmpeg:
 
@@ -109,35 +209,6 @@ You should edit the ffserver.conf file to suit your needs (in terms of
 frame rates etc). Then install ffserver and ffmpeg, write a script to start
 them up, and off you go.
 
-@section Troubleshooting
-
-@subsection I don't hear any audio, but video is fine.
-
-Maybe you didn't install LAME, or got your ./configure statement wrong. Check
-the ffmpeg output to see if a line referring to MP3 is present. If not, then
-your configuration was incorrect. If it is, then maybe your wiring is not
-set up correctly. Maybe the sound card is not getting data from the right
-input source. Maybe you have a really awful audio interface (like I do)
-that only captures in stereo and also requires that one channel be flipped.
-If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before
-starting ffmpeg.
-
-@subsection The audio and video lose sync after a while.
-
-Yes, they do.
-
-@subsection After a long while, the video update rate goes way down in WMP.
-
-Yes, it does. Who knows why?
-
-@subsection WMP 6.4 behaves differently to WMP 7.
-
-Yes, it does. Any thoughts on this would be gratefully received. These
-differences extend to embedding WMP into a web page. [There are two
-object IDs that you can use: The old one, which does not play well, and
-the new one, which does (both tested on the same system). However,
-I suspect that the new one is not available unless you have installed WMP 7].
-
 @section What else can it do?
 
 You can replay video from .ffm files that was recorded earlier.
@@ -177,9 +248,6 @@ specify a time. In addition, ffserver will skip frames until a key_frame
 is found. This further reduces the startup delay by not transferring data
 that will be discarded.
 
-* You may want to adjust the MaxBandwidth in the ffserver.conf to limit
-the amount of bandwidth consumed by live streams.
-
 @section Why does the ?buffer / Preroll stop working after a time?
 
 It turns out that (on my machine at least) the number of frames successfully
@@ -213,19 +281,6 @@ You use this by adding the ?date= to the end of the URL for the stream.
 For example:   @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
 @c man end
 
-@section What is FFM, FFM2
-
-FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
-video and audio streams and encoding options, and can store a moving time segment
-of an infinite movie or a whole movie.
-
-FFM is version specific, and there is limited compatibility of FFM files
-generated by one version of ffmpeg/ffserver and another version of
-ffmpeg/ffserver. It may work but it is not guaranteed to work.
-
-FFM2 is extensible while maintaining compatibility and should work between
-differing versions of tools. FFM2 is the default.
-
 @chapter Options
 @c man begin OPTIONS
 
@@ -235,15 +290,540 @@ differing versions of tools. FFM2 is the default.
 
 @table @option
 @item -f @var{configfile}
-Use @file{configfile} instead of @file{/etc/ffserver.conf}.
+Read configuration file @file{configfile}. If not specified it will
+read by default from @file{/etc/ffserver.conf}.
+
 @item -n
-Enable no-launch mode. This option disables all the Launch directives
-within the various <Stream> sections. Since ffserver will not launch
-any ffmpeg instances, you will have to launch them manually.
+Enable no-launch mode. This option disables all the @code{Launch}
+directives within the various @code{<Feed>} sections. Since
+@command{ffserver} will not launch any @command{ffmpeg} instances, you
+will have to launch them manually.
+
 @item -d
-Enable debug mode. This option increases log verbosity, directs log
-messages to stdout.
+Enable debug mode. This option increases log verbosity, and directs
+log messages to stdout. When specified, the @option{CustomLog} option
+is ignored.
 @end table
+
+@chapter Configuration file syntax
+
+@command{ffserver} reads a configuration file containing global
+options and settings for each stream and feed.
+
+The configuration file consists of global options and dedicated
+sections, which must be introduced by "<@var{SECTION_NAME}
+@var{ARGS}>" on a separate line and must be terminated by a line in
+the form "</@var{SECTION_NAME}>". @var{ARGS} is optional.
+
+Currently the following sections are recognized: @samp{Feed},
+@samp{Stream}, @samp{Redirect}.
+
+A line starting with @code{#} is ignored and treated as a comment.
+
+Name of options and sections are case-insensitive.
+
+@section ACL syntax
+An ACL (Access Control List) specifies the address which are allowed
+to access a given stream, or to write a given feed.
+
+It accepts the folling forms
+@itemize
+@item
+Allow/deny access to @var{address}.
+@example
+ACL ALLOW <address>
+ACL DENY <address>
+@end example
+
+@item
+Allow/deny access to ranges of addresses from @var{first_address} to
+@var{last_address}.
+@example
+ACL ALLOW <first_address> <last_address>
+ACL DENY <first_address> <last_address>
+@end example
+@end itemize
+
+You can repeat the ACL allow/deny as often as you like. It is on a per
+stream basis. The first match defines the action. If there are no matches,
+then the default is the inverse of the last ACL statement.
+
+Thus 'ACL allow localhost' only allows access from localhost.
+'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
+allow everybody else.
+
+@section Global options
+@table @option
+@item Port @var{port_number}
+@item RTSPPort @var{port_number}
+
+Set TCP port number on which the HTTP/RTSP server is listening. You
+must select a different port from your standard HTTP web server if it
+is running on the same computer.
+
+If not specified, no corresponding server will be created.
+
+@item BindAddress @var{ip_address}
+@item RTSPBindAddress @var{ip_address}
+Set address on which the HTTP/RTSP server is bound. Only useful if you
+have several network interfaces.
+
+@item MaxHTTPConnections @var{n}
+Set number of simultaneous HTTP connections that can be handled. It
+has to be defined @emph{before} the @option{MaxClients} parameter,
+since it defines the @option{MaxClients} maximum limit.
+
+Default value is 2000.
+
+@item MaxClients @var{n}
+Set number of simultaneous requests that can be handled. Since
+@command{ffserver} is very fast, it is more likely that you will want
+to leave this high and use @option{MaxBandwidth}.
+
+Default value is 5.
+
+@item MaxBandwidth @var{kbps}
+Set the maximum amount of kbit/sec that you are prepared to consume
+when streaming to clients.
+
+Default value is 1000.
+
+@item CustomLog @var{filename}
+Set access log file (uses standard Apache log file format). '-' is the
+standard output.
+
+If not specified @command{ffserver} will produce no log.
+
+In case the commandline option @option{-d} is specified this option is
+ignored, and the log is written to standard output.
+
+@item NoDaemon
+Set no-daemon mode. This option is currently ignored since now
+@command{ffserver} will always work in no-daemon mode, and is
+deprecated.
+@end table
+
+@section Feed section
+
+A Feed section defines a feed provided to @command{ffserver}.
+
+Each live feed contains one video and/or audio sequence coming from an
+@command{ffmpeg} encoder or another @command{ffserver}. This sequence
+may be encoded simultaneously with several codecs at several
+resolutions.
+
+A feed instance specification is introduced by a line in the form:
+@example
+<Feed FEED_FILENAME>
+@end example
+
+where @var{FEED_FILENAME} specifies the unique name of the FFM stream.
+
+The following options are recognized within a Feed section.
+
+@table @option
+@item File @var{filename}
+@item ReadOnlyFile @var{filename}
+Set the path where the feed file is stored on disk.
+
+If not specified, the @file{/tmp/FEED.ffm} is assumed, where
+@var{FEED} is the feed name.
+
+If @option{ReadOnlyFile} is used the file is marked as read-only and
+it will not be deleted or updated.
+
+@item Truncate
+Truncate the feed file, rather than appending to it. By default
+@command{ffserver} will append data to the file, until the maximum
+file size value is reached (see @option{FileMaxSize} option).
+
+@item FileMaxSize @var{size}
+Set maximum size of the feed file in bytes. 0 means unlimited. The
+postfixes @code{K} (2^10), @code{M} (2^20), and @code{G} (2^30) are
+recognized.
+
+Default value is 5M.
+
+@item Launch @var{args}
+Launch an @command{ffmpeg} command when creating @command{ffserver}.
+
+@var{args} must be a sequence of arguments to be provided to an
+@command{ffmpeg} instance. The first provided argument is ignored, and
+it is replaced by a path with the same dirname of the @command{ffserver}
+instance, followed by the remaining argument and terminated with a
+path corresponding to the feed.
+
+When the launched process exits, @command{ffserver} will launch
+another program instance.
+
+In case you need a more complex @command{ffmpeg} configuration,
+e.g. if you need to generate multiple FFM feeds with a single
+@command{ffmpeg} instance, you should launch @command{ffmpeg} by hand.
+
+This option is ignored in case the commandline option @option{-n} is
+specified.
+
+@item ACL @var{spec}
+Specify the list of IP address which are allowed or denied to write
+the feed. Multiple ACL options can be specified.
+@end table
+
+@section Stream section
+
+A Stream section defines a stream provided by @command{ffserver}, and
+identified by a single name.
+
+The stream is sent when answering a request containing the stream
+name.
+
+A stream section must be introduced by the line:
+@example
+<Stream STREAM_NAME>
+@end example
+
+where @var{STREAM_NAME} specifies the unique name of the stream.
+
+The following options are recognized within a Stream section.
+
+Encoding options are marked with the @emph{encoding} tag, and they are
+used to set the encoding parameters, and are mapped to libavcodec
+encoding options. Not all encoding options are supported, in
+particular it is not possible to set encoder private options. In order
+to override the encoding options specified by @command{ffserver}, you
+can use the @command{ffmpeg} @option{override_ffserver} commandline
+option.
+
+Only one of the @option{Feed} and @option{File} options should be set.
+
+@table @option
+@item Feed @var{feed_name}
+Set the input feed. @var{feed_name} must correspond to an existing
+feed defined in a @code{Feed} section.
+
+When this option is set, encoding options are used to setup the
+encoding operated by the remote @command{ffmpeg} process.
+
+@item File @var{filename}
+Set the filename of the pre-recorded input file to stream.
+
+When this option is set, encoding options are ignored and the input
+file content is re-streamed as is.
+
+@item Format @var{format_name}
+Set the format of the output stream.
+
+Must be the name of a format recognized by FFmpeg. If set to
+@samp{status}, it is treated as a status stream.
+
+@item InputFormat @var{format_name}
+Set input format. If not specified, it is automatically guessed.
+
+@item Preroll @var{n}
+Set this to the number of seconds backwards in time to start. Note that
+most players will buffer 5-10 seconds of video, and also you need to allow
+for a keyframe to appear in the data stream.
+
+Default value is 0.
+
+@item StartSendOnKey
+Do not send stream until it gets the first key frame. By default
+@command{ffserver} will send data immediately.
+
+@item MaxTime @var{n}
+Set the number of seconds to run. This value set the maximum duration
+of the stream a client will be able to receive.
+
+A value of 0 means that no limit is set on the stream duration.
+
+@item ACL @var{spec}
+Set ACL for the stream.
+
+@item DynamicACL @var{spec}
+
+@item RTSPOption @var{option}
+
+@item MulticastAddress @var{address}
+
+@item MulticastPort @var{port}
+
+@item MulticastTTL @var{integer}
+
+@item NoLoop
+
+@item FaviconURL @var{url}
+Set favicon (favourite icon) for the server status page. It is ignored
+for regular streams.
+
+@item Author @var{value}
+@item Comment @var{value}
+@item Copyright @var{value}
+@item Title @var{value}
+Set metadata corresponding to the option. All these options are
+deprecated in favor of @option{Metadata}.
+
+@item Metadata @var{key} @var{value}
+Set metadata value on the output stream.
+
+@item NoAudio
+@item NoVideo
+Suppress audio/video.
+
+@item AudioCodec @var{codec_name} (@emph{encoding,audio})
+Set audio codec.
+
+@item AudioBitRate @var{rate} (@emph{encoding,audio})
+Set bitrate for the audio stream in kbits per second.
+
+@item AudioChannels @var{n} (@emph{encoding,audio})
+Set number of audio channels.
+
+@item AudioSampleRate @var{n} (@emph{encoding,audio})
+Set sampling frequency for audio. When using low bitrates, you should
+lower this frequency to 22050 or 11025. The supported frequencies
+depend on the selected audio codec.
+
+@item AVOptionAudio @var{option} @var{value} (@emph{encoding,audio})
+Set generic option for audio stream.
+
+@item AVPresetAudio @var{preset} (@emph{encoding,audio})
+Set preset for audio stream.
+
+@item VideoCodec @var{codec_name} (@emph{encoding,video})
+Set video codec.
+
+@item VideoBitRate @var{n} (@emph{encoding,video})
+Set bitrate for the video stream in kbits per second.
+
+@item VideoBitRateRange @var{range} (@emph{encoding,video})
+Set video bitrate range.
+
+A range must be specified in the form @var{minrate}-@var{maxrate}, and
+specifies the @option{minrate} and @option{maxrate} encoding options
+expressed in kbits per second.
+
+@item VideoBitRateRangeTolerance @var{n} (@emph{encoding,video})
+Set video bitrate tolerance in kbits per second.
+
+@item PixelFormat @var{pixel_format} (@emph{encoding,video})
+Set video pixel format.
+
+@item Debug @var{integer} (@emph{encoding,video})
+Set video @option{debug} encoding option.
+
+@item Strict @var{integer} (@emph{encoding,video})
+Set video @option{strict} encoding option.
+
+@item VideoBufferSize @var{n} (@emph{encoding,video})
+Set ratecontrol buffer size, expressed in KB.
+
+@item VideoFrameRate @var{n} (@emph{encoding,video})
+Set number of video frames per second.
+
+@item VideoSize (@emph{encoding,video})
+Set size of the video frame, must be an abbreviation or in the form
+@var{W}x@var{H}.  See @ref{video size syntax,,the Video size section
+in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+
+Default value is @code{160x128}.
+
+@item VideoIntraOnly (@emph{encoding,video})
+Transmit only intra frames (useful for low bitrates, but kills frame rate).
+
+@item VideoGopSize @var{n} (@emph{encoding,video})
+If non-intra only, an intra frame is transmitted every VideoGopSize
+frames. Video synchronization can only begin at an intra frame.
+
+@item VideoTag @var{tag} (@emph{encoding,video})
+Set video tag.
+
+@item VideoHighQuality (@emph{encoding,video})
+@item Video4MotionVector (@emph{encoding,video})
+
+@item BitExact (@emph{encoding,video})
+Set bitexact encoding flag.
+
+@item IdctSimple (@emph{encoding,video})
+Set simple IDCT algorithm.
+
+@item Qscale @var{n} (@emph{encoding,video})
+Enable constant quality encoding, and set video qscale (quantization
+scale) value, expressed in @var{n} QP units.
+
+@item VideoQMin @var{n} (@emph{encoding,video})
+@item VideoQMax @var{n} (@emph{encoding,video})
+Set video qmin/qmax.
+
+@item VideoQDiff @var{integer} (@emph{encoding,video})
+Set video @option{qdiff} encoding option.
+
+@item LumiMask @var{float} (@emph{encoding,video})
+@item DarkMask @var{float} (@emph{encoding,video})
+Set @option{lumi_mask}/@option{dark_mask} encoding options.
+
+@item AVOptionVideo @var{option} @var{value} (@emph{encoding,video})
+Set generic option for video stream.
+
+@item AVPresetVideo @var{preset} (@emph{encoding,video})
+Set preset for video stream.
+
+@var{preset} must be the path of a preset file.
+@end table
+
+@subsection Server status stream
+
+A server status stream is a special stream which is used to show
+statistics about the @command{ffserver} operations.
+
+It must be specified setting the option @option{Format} to
+@samp{status}.
+
+@section Redirect section
+
+A redirect section specifies where to redirect the requested URL to
+another page.
+
+A redirect section must be introduced by the line:
+@example
+<Redirect NAME>
+@end example
+
+where @var{NAME} is the name of the page which should be redirected.
+
+It only accepts the option @option{URL}, which specify the redirection
+URL.
+
+@chapter Stream examples
+
+@itemize
+@item
+Multipart JPEG
+@example
+<Stream test.mjpg>
+Feed feed1.ffm
+Format mpjpeg
+VideoFrameRate 2
+VideoIntraOnly
+NoAudio
+Strict -1
+</Stream>
+@end example
+
+@item
+Single JPEG
+@example
+<Stream test.jpg>
+Feed feed1.ffm
+Format jpeg
+VideoFrameRate 2
+VideoIntraOnly
+VideoSize 352x240
+NoAudio
+Strict -1
+</Stream>
+@end example
+
+@item
+Flash
+@example
+<Stream test.swf>
+Feed feed1.ffm
+Format swf
+VideoFrameRate 2
+VideoIntraOnly
+NoAudio
+</Stream>
+@end example
+
+@item
+ASF compatible
+@example
+<Stream test.asf>
+Feed feed1.ffm
+Format asf
+VideoFrameRate 15
+VideoSize 352x240
+VideoBitRate 256
+VideoBufferSize 40
+VideoGopSize 30
+AudioBitRate 64
+StartSendOnKey
+</Stream>
+@end example
+
+@item
+MP3 audio
+@example
+<Stream test.mp3>
+Feed feed1.ffm
+Format mp2
+AudioCodec mp3
+AudioBitRate 64
+AudioChannels 1
+AudioSampleRate 44100
+NoVideo
+</Stream>
+@end example
+
+@item
+Ogg Vorbis audio
+@example
+<Stream test.ogg>
+Feed feed1.ffm
+Metadata title "Stream title"
+AudioBitRate 64
+AudioChannels 2
+AudioSampleRate 44100
+NoVideo
+</Stream>
+@end example
+
+@item
+Real with audio only at 32 kbits
+@example
+<Stream test.ra>
+Feed feed1.ffm
+Format rm
+AudioBitRate 32
+NoVideo
+</Stream>
+@end example
+
+@item
+Real with audio and video at 64 kbits
+@example
+<Stream test.rm>
+Feed feed1.ffm
+Format rm
+AudioBitRate 32
+VideoBitRate 128
+VideoFrameRate 25
+VideoGopSize 25
+</Stream>
+@end example
+
+@item
+For stream coming from a file: you only need to set the input filename
+and optionally a new format.
+
+@example
+<Stream file.rm>
+File "/usr/local/httpd/htdocs/tlive.rm"
+NoAudio
+</Stream>
+@end example
+
+@example
+<Stream file.asf>
+File "/usr/local/httpd/htdocs/test.asf"
+NoAudio
+Metadata author "Me"
+Metadata copyright "Super MegaCorp"
+Metadata title "Test stream from disk"
+Metadata comment "Test comment"
+</Stream>
+@end example
+@end itemize
+
 @c man end
 
 @include config.texi

doc/fftools-common-opts.texi

@@ -194,6 +194,13 @@ to a plain @code{%}
 Errors in parsing the environment variable are not fatal, and will not
 appear in the report.
 
+@item -hide_banner
+Suppress printing banner.
+
+All FFmpeg tools will normally show a copyright notice, build options
+and library versions. This option can be used to suppress printing
+this information.
+
 @item -cpuflags flags (@emph{global})
 Allows setting and clearing cpu flags. This option is intended
 for testing. Do not use it unless you know what you're doing.
@@ -250,6 +257,10 @@ Possible flags for this option are:
 @end table
 @end table
 
+@item -opencl_bench
+Benchmark all available OpenCL devices and show the results. This option
+is only available when FFmpeg has been compiled with @code{--enable-opencl}.
+
 @item -opencl_options options (@emph{global})
 Set OpenCL environment options. This option is only available when
 FFmpeg has been compiled with @code{--enable-opencl}.

doc/formats.texi

@@ -125,18 +125,26 @@ Consider things that a sane encoder should not do as an error.
 Use wallclock as timestamps.
 
 @item avoid_negative_ts @var{integer} (@emph{output})
-Shift timestamps to make them non-negative. A value of 1 enables shifting,
-a value of 0 disables it, the default value of -1 enables shifting
-when required by the target format.
+
+Possible values:
+@table @samp
+@item make_non_negative
+Shift timestamps to make them non-negative.
+Also note that this affects only leading negative timestamps, and not
+non-monotonic negative timestamps.
+@item make_zero
+Shift timestamps so that the first timestamp is 0.
+@item auto (default)
+Enables shifting when required by the target format.
+@item disabled
+Disables shifting of timestamp.
+@end table
 
 When shifting is enabled, all output timestamps are shifted by the
 same amount. Audio, video, and subtitles desynching and relative
 timestamp differences are preserved compared to how they would have
 been without shifting.
 
-Also note that this affects only leading negative timestamps, and not
-non-monotonic negative timestamps.
-
 @item skip_initial_bytes @var{integer} (@emph{input})
 Set number of bytes to skip before reading header and frames if set to 1.
 Default is 0.
@@ -148,6 +156,18 @@ Correct single timestamp overflows if set to 1. Default is 1.
 Flush the underlying I/O stream after each packet. Default 1 enables it, and
 has the effect of reducing the latency; 0 disables it and may slightly
 increase performance in some cases.
+
+@item output_ts_offset @var{offset} (@emph{output})
+Set the output time offset.
+
+@var{offset} must be a time duration specification,
+see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+
+The offset is added by the muxer to the output timestamps.
+
+Specifying a positive offset means that the corresponding streams are
+delayed bt the time duration specified in @var{offset}. Default value
+is @code{0} (meaning that no offset is applied).
 @end table
 
 @c man end FORMAT OPTIONS
@@ -183,6 +203,10 @@ The exact semantics of stream specifiers is defined by the
 @code{avformat_match_stream_specifier()} function declared in the
 @file{libavformat/avformat.h} header.
 
+@ifclear config-writeonly
 @include demuxers.texi
+@end ifclear
+@ifclear config-readonly
 @include muxers.texi
+@end ifclear
 @include metadata.texi

doc/general.texi

@@ -122,6 +122,20 @@ x264 is under the GNU Public License Version 2 or later
 details), you must upgrade FFmpeg's license to GPL in order to use it.
 @end float
 
+@section x265
+
+FFmpeg can make use of the x265 library for HEVC encoding.
+
+Go to @url{http://x265.org/developers.html} and follow the instructions
+for installing the library. Then pass @code{--enable-libx265} to configure
+to enable it.
+
+@float note
+x265 is under the GNU Public License Version 2 or later
+(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for
+details), you must upgrade FFmpeg's license to GPL in order to use it.
+@end float
+
 @section libilbc
 
 iLBC is a narrowband speech codec that has been made freely available
@@ -148,6 +162,27 @@ libzvbi is licensed under the GNU General Public License Version 2 or later
 you must upgrade FFmpeg's license to GPL in order to use it.
 @end float
 
+@section AviSynth
+
+FFmpeg can read AviSynth scripts as input. To enable support, pass
+@code{--enable-avisynth} to configure.  The correct headers are
+included in compat/avisynth/, which allows the user to enable support
+without needing to search for these headers themselves.
+
+For Windows, supported AviSynth variants are
+@url{http://avisynth.nl, AviSynth 2.5 or 2.6} for 32-bit builds and
+@url{http://avs-plus.net, AviSynth+ 0.1} for 32-bit and 64-bit builds.
+
+For Linux and OS X, the supported AviSynth variant is
+@url{https://github.com/avxsynth/avxsynth, AvxSynth}.
+
+@float NOTE
+AviSynth and AvxSynth are loaded dynamically.  Distributors can build FFmpeg
+with @code{--enable-avisynth}, and the binaries will work regardless of the
+end user having AviSynth or AvxSynth installed - they'll only need to be
+installed to use AviSynth scripts (obviously).
+@end float
+
 
 @chapter Supported File Formats, Codecs or Features
 
@@ -249,6 +284,8 @@ library:
 @item GXF                       @tab X @tab X
     @tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley
          playout servers.
+@item HNM @tab   @tab X
+    @tab Only version 4 supported, used in some games from Cryo Interactive
 @item iCEDraw File              @tab   @tab X
 @item ICO                       @tab X @tab X
     @tab Microsoft Windows ICO
@@ -299,6 +336,8 @@ library:
     @tab also known as DVB Transport Stream
 @item MPEG-4                    @tab X @tab X
     @tab MPEG-4 is a variant of QuickTime.
+@item Mirillis FIC video        @tab   @tab X
+    @tab No cursor rendering.
 @item MIME multipart JPEG       @tab X @tab
 @item MSN TCP webcam            @tab   @tab X
     @tab Used by MSN Messenger webcam streams.
@@ -338,6 +377,7 @@ library:
 @item raw H.261                 @tab X @tab X
 @item raw H.263                 @tab X @tab X
 @item raw H.264                 @tab X @tab X
+@item raw HEVC                  @tab X @tab X
 @item raw Ingenient MJPEG       @tab   @tab X
 @item raw MJPEG                 @tab X @tab X
 @item raw MLP                   @tab   @tab X
@@ -488,8 +528,8 @@ following image formats are supported:
     @tab YUV, JPEG and some extension is not supported yet.
 @item Truevision Targa  @tab X @tab X
     @tab Targa (.TGA) image format
-@item WebP         @tab @tab X
-    @tab WebP image format
+@item WebP         @tab E @tab X
+    @tab WebP image format, encoding supported through external library libwebp
 @item XBM  @tab X @tab X
     @tab X BitMap image format
 @item XFace @tab X @tab X
@@ -607,6 +647,9 @@ following image formats are supported:
 @item H.263+ / H.263-1998 / H.263 version 2  @tab  X  @tab  X
 @item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10  @tab  E  @tab  X
     @tab encoding supported through external library libx264
+@item HEVC                   @tab  X  @tab  X
+    @tab encoding supported through the external library libx265
+@item HNM version 4          @tab     @tab  X
 @item HuffYUV                @tab  X  @tab  X
 @item HuffYUV FFmpeg variant @tab  X  @tab  X
 @item IBM Ultimotion         @tab     @tab  X
@@ -658,7 +701,6 @@ following image formats are supported:
 @item Mobotix MxPEG video    @tab     @tab  X
 @item Motion Pixels video    @tab     @tab  X
 @item MPEG-1 video           @tab  X  @tab  X
-@item MPEG-1/2 video XvMC (X-Video Motion Compensation)  @tab     @tab  X
 @item MPEG-2 video           @tab  X  @tab  X
 @item MPEG-4 part 2          @tab  X  @tab  X
     @tab libxvidcore can be used alternatively for encoding.
@@ -829,6 +871,7 @@ following image formats are supported:
     @tab QuickTime fourcc 'alac'
 @item ATRAC1                 @tab     @tab  X
 @item ATRAC3                 @tab     @tab  X
+@item ATRAC3+                @tab     @tab  X
 @item Bink Audio             @tab     @tab  X
     @tab Used in Bink and Smacker files in many games.
 @item CELT                   @tab     @tab  E
@@ -939,7 +982,6 @@ following image formats are supported:
 @item Vorbis                 @tab  E  @tab  X
     @tab A native but very primitive encoder exists.
 @item Voxware MetaSound      @tab     @tab  X
-    @tab imperfect and incomplete support
 @item WavPack                @tab  X  @tab  X
 @item Westwood Audio (SND1)  @tab     @tab  X
 @item Windows Media Audio 1  @tab  X  @tab  X
@@ -993,6 +1035,7 @@ performance on systems without hardware floating point support).
 @multitable @columnfractions .4 .1
 @item Name         @tab Support
 @item file         @tab X
+@item FTP          @tab X
 @item Gopher       @tab X
 @item HLS          @tab X
 @item HTTP         @tab X
@@ -1008,6 +1051,7 @@ performance on systems without hardware floating point support).
 @item RTMPTS       @tab X
 @item RTP          @tab X
 @item SCTP         @tab X
+@item SFTP         @tab E
 @item TCP          @tab X
 @item TLS          @tab X
 @item UDP          @tab X
@@ -1027,13 +1071,14 @@ performance on systems without hardware floating point support).
 @item caca              @tab        @tab X
 @item DV1394            @tab X      @tab
 @item Lavfi virtual device @tab X   @tab
-@item Linux framebuffer @tab X      @tab
+@item Linux framebuffer @tab X      @tab X
 @item JACK              @tab X      @tab
 @item LIBCDIO           @tab X
 @item LIBDC1394         @tab X      @tab
 @item OpenAL            @tab X
+@item OpenGL            @tab        @tab X
 @item OSS               @tab X      @tab X
-@item Pulseaudio        @tab X      @tab
+@item PulseAudio        @tab X      @tab X
 @item SDL               @tab        @tab X
 @item Video4Linux2      @tab X      @tab X
 @item VfW capture       @tab X      @tab

doc/git-howto.texi

@@ -299,7 +299,7 @@ the current branch history.
 git commit --amend
 @end example
 
-allows to amend the last commit details quickly.
+allows one to amend the last commit details quickly.
 
 @example
 git rebase -i origin/master

doc/git-howto.txt

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

doc/indevs.texi

@@ -409,7 +409,7 @@ OpenAL is part of Core Audio, the official Mac OS X Audio interface.
 See @url{http://developer.apple.com/technologies/mac/audio-and-video.html}
 @end table
 
-This device allows to capture from an audio input device handled
+This device allows one to capture from an audio input device handled
 through OpenAL.
 
 You need to specify the name of the device to capture in the provided
@@ -617,7 +617,7 @@ Select the pixel format (only valid for raw video input).
 
 @item input_format
 Set the preferred pixel format (for raw video) or a codec name.
-This option allows to select the input format, when several are
+This option allows one to select the input format, when several are
 available.
 
 @item framerate
@@ -678,7 +678,7 @@ other filename will be interpreted as device number 0.
 
 X11 video input device.
 
-This device allows to capture a region of an X11 display.
+This device allows one to capture a region of an X11 display.
 
 The filename passed as input has the syntax:
 @example

doc/libavutil.texi

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

doc/muxers.texi

@@ -23,6 +23,8 @@ A description of some of the currently available muxers follows.
 
 Audio Interchange File Format muxer.
 
+@subsection Options
+
 It accepts the following options:
 
 @table @option
@@ -49,6 +51,10 @@ The output of the muxer consists of a single line of the form:
 CRC=0x@var{CRC}, where @var{CRC} is a hexadecimal number 0-padded to
 8 digits containing the CRC for all the decoded input frames.
 
+See also the @ref{framecrc} muxer.
+
+@subsection Examples
+
 For example to compute the CRC of the input, and store it in the file
 @file{out.crc}:
 @example
@@ -68,8 +74,6 @@ and the input video converted to MPEG-2 video, use the command:
 ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc -
 @end example
 
-See also the @ref{framecrc} muxer.
-
 @anchor{framecrc}
 @section framecrc
 
@@ -89,6 +93,8 @@ packet of the form:
 @var{CRC} is a hexadecimal number 0-padded to 8 digits containing the
 CRC of the packet.
 
+@subsection Examples
+
 For example to compute the CRC of the audio and video frames in
 @file{INPUT}, converted to raw audio and video packets, and store it
 in the file @file{out.crc}:
@@ -132,6 +138,8 @@ packet of the form:
 @var{MD5} is a hexadecimal number representing the computed MD5 hash
 for the packet.
 
+@subsection Examples
+
 For example to compute the MD5 of the audio and video frames in
 @file{INPUT}, converted to raw audio and video packets, and store it
 in the file @file{out.md5}:
@@ -146,30 +154,89 @@ ffmpeg -i INPUT -f framemd5 -
 
 See also the @ref{md5} muxer.
 
+@anchor{gif}
+@section gif
+
+Animated GIF muxer.
+
+It accepts the following options:
+
+@table @option
+@item loop
+Set the number of times to loop the output. Use @code{-1} for no loop, @code{0}
+for looping indefinitely (default).
+
+@item final_delay
+Force the delay (expressed in centiseconds) after the last frame. Each frame
+ends with a delay until the next frame. The default is @code{-1}, which is a
+special value to tell the muxer to re-use the previous delay. In case of a
+loop, you might want to customize this value to mark a pause for instance.
+@end table
+
+For example, to encode a gif looping 10 times, with a 5 seconds delay between
+the loops:
+@example
+ffmpeg -i INPUT -loop 10 -final_delay 500 out.gif
+@end example
+
+Note 1: if you wish to extract the frames in separate GIF files, you need to
+force the @ref{image2} muxer:
+@example
+ffmpeg -i INPUT -c:v gif -f image2 "out%d.gif"
+@end example
+
+Note 2: the GIF format has a very small time base: the delay between two frames
+can not be smaller than one centi second.
+
 @anchor{hls}
 @section hls
 
 Apple HTTP Live Streaming muxer that segments MPEG-TS according to
-the HTTP Live Streaming specification.
+the HTTP Live Streaming (HLS) specification.
 
 It creates a playlist file and numbered segment files. The output
 filename specifies the playlist filename; the segment filenames
 receive the same basename as the playlist, a sequential number and
 a .ts extension.
 
+For example, to convert an input file with @command{ffmpeg}:
 @example
 ffmpeg -i in.nut out.m3u8
 @end example
 
+See also the @ref{segment} muxer, which provides a more generic and
+flexible implementation of a segmenter, and can be used to perform HLS
+segmentation.
+
+@subsection Options
+
+This muxer supports the following options:
+
 @table @option
-@item -hls_time @var{seconds}
-Set the segment length in seconds.
-@item -hls_list_size @var{size}
-Set the maximum number of playlist entries.
-@item -hls_wrap @var{wrap}
-Set the number after which index wraps.
-@item -start_number @var{number}
-Start the sequence from @var{number}.
+@item hls_time @var{seconds}
+Set the segment length in seconds. Default value is 2.
+
+@item hls_list_size @var{size}
+Set the maximum number of playlist entries. If set to 0 the list file
+will contain all the segments. Default value is 5.
+
+@item hls_wrap @var{wrap}
+Set the number after which the segment filename number (the number
+specified in each segment file) wraps. If set to 0 the number will be
+never wrapped. Default value is 0.
+
+This option is useful to avoid to fill the disk with many segment
+files, and limits the maximum number of segment files written to disk
+to @var{wrap}.
+
+@item start_number @var{number}
+Start the playlist sequence number from @var{number}. Default value is
+0.
+
+Note that the playlist sequence number must be unique for each segment
+and it is not to be confused with the segment filename sequence number
+which can be cyclic, for example if the @option{wrap} option is
+specified.
 @end table
 
 @anchor{ico}
@@ -235,6 +302,8 @@ The pattern "img%%-%d.jpg" will specify a sequence of filenames of the
 form @file{img%-1.jpg}, @file{img%-2.jpg}, ..., @file{img%-10.jpg},
 etc.
 
+@subsection Examples
+
 The following example shows how to use @command{ffmpeg} for creating a
 sequence of files @file{img-001.jpeg}, @file{img-002.jpeg}, ...,
 taking one image every second from the input video:
@@ -257,16 +326,32 @@ Note also that the pattern must not necessarily contain "%d" or
 ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg
 @end example
 
+The @option{strftime} option allows you to expand the filename with
+date and time information. Check the documentation of
+the @code{strftime()} function for the syntax.
+
+For example to generate image files from the @code{strftime()}
+"%Y-%m-%d_%H-%M-%S" pattern, the following @command{ffmpeg} command
+can be used:
+@example
+ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg"
+@end example
+
+@subsection Options
+
 @table @option
-@item start_number @var{number}
-Start the sequence from @var{number}. Default value is 1. Must be a
-non-negative number.
+@item start_number
+Start the sequence from the specified number. Default value is 1. Must
+be a non-negative number.
 
-@item -update @var{number}
-If @var{number} is nonzero, the filename will always be interpreted as just a
-filename, not a pattern, and this file will be continuously overwritten with new
-images.
+@item update
+If set to 1, the filename will always be interpreted as just a
+filename, not a pattern, and the corresponding file will be continuously
+overwritten with new images. Default value is 0.
 
+@item strftime
+If set to 1, expand the filename with date and time information from
+@code{strftime()}. Default value is 0.
 @end table
 
 The image muxer supports the .Y.U.V image file format. This format is
@@ -281,25 +366,27 @@ Matroska container muxer.
 
 This muxer implements the matroska and webm container specs.
 
+@subsection Metadata
+
 The recognized metadata settings in this muxer are:
 
 @table @option
+@item title
+Set title name provided to a single track.
 
-@item title=@var{title name}
-Name provided to a single track
-@end table
+@item language
+Specify the language of the track in the Matroska languages form.
 
-@table @option
+The language can be either the 3 letters bibliographic ISO-639-2 (ISO
+639-2/B) form (like "fre" for French), or a language code mixed with a
+country code for specialities in languages (like "fre-ca" for Canadian
+French).
 
-@item language=@var{language name}
-Specifies the language of the track in the Matroska languages form
-@end table
+@item stereo_mode
+Set stereo 3D video layout of two views in a single video track.
 
-@table @option
-
-@item stereo_mode=@var{mode}
-Stereo 3D video layout of two views in a single video track
-@table @option
+The following values are recognized:
+@table @samp
 @item mono
 video is not stereo
 @item left_right
@@ -338,10 +425,11 @@ For example a 3D WebM clip can be created using the following command line:
 ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm
 @end example
 
+@subsection Options
+
 This muxer supports the following options:
 
 @table @option
-
 @item reserve_index_space
 By default, this muxer writes the index for seeking (called cues in Matroska
 terms) at the end of the file, because it cannot know in advance how much space
@@ -356,7 +444,6 @@ for most use cases should be about 50kB per hour of video.
 
 Note that cues are only written if the output is seekable and this option will
 have no effect if it is not.
-
 @end table
 
 @anchor{md5}
@@ -386,7 +473,9 @@ ffmpeg -i INPUT -f md5 -
 
 See also the @ref{framemd5} muxer.
 
-@section MOV/MP4/ISMV
+@section mov, mp4, ismv
+
+MOV/MP4/ISMV (Smooth Streaming) muxer.
 
 The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4
 file has all the metadata about all packets stored in one location
@@ -402,6 +491,8 @@ very long files (since writing normal MOV/MP4 files stores info about
 every single packet in memory until the file is closed). The downside
 is that it is less compatible with other applications.
 
+@subsection Options
+
 Fragmentation is enabled by setting one of the AVOptions that define
 how to cut the file into fragments:
 
@@ -458,6 +549,8 @@ as fragmented output, thus it is not enabled by default.
 Add RTP hinting tracks to the output file.
 @end table
 
+@subsection Example
+
 Smooth Streaming content can be pushed in real time to a publishing
 point on IIS with this muxer. Example:
 @example
@@ -468,12 +561,15 @@ ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe
 
 The MP3 muxer writes a raw MP3 stream with an ID3v2 header at the beginning and
 optionally an ID3v1 tag at the end. ID3v2.3 and ID3v2.4 are supported, the
-@code{id3v2_version} option controls which one is used. The legacy ID3v1 tag is
-not written by default, but may be enabled with the @code{write_id3v1} option.
+@code{id3v2_version} option controls which one is used. Setting
+@code{id3v2_version} to 0 will disable the ID3v2 header completely. The legacy
+ID3v1 tag is not written by default, but may be enabled with the
+@code{write_id3v1} option.
 
-For seekable output the muxer also writes a Xing frame at the beginning, which
-contains the number of frames in the file. It is useful for computing duration
-of VBR files.
+The muxer may also write a Xing frame at the beginning, which contains the
+number of frames in the file. It is useful for computing duration of VBR files.
+The Xing frame is written if the output stream is seekable and if the
+@code{write_xing} option is set to 1 (the default).
 
 The muxer supports writing ID3v2 attached pictures (APIC frames). The pictures
 are supplied to the muxer in form of a video stream with a single packet. There
@@ -500,12 +596,24 @@ ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1
 -metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3
 @end example
 
+Write a "clean" MP3 without any extra features:
+@example
+ffmpeg -i input.wav -write_xing 0 -id3v2_version 0 out.mp3
+@end example
+
 @section mpegts
 
 MPEG transport stream muxer.
 
 This muxer implements ISO 13818-1 and part of ETSI EN 300 468.
 
+The recognized metadata settings in mpegts muxer are @code{service_provider}
+and @code{service_name}. If they are not set the default for
+@code{service_provider} is "FFmpeg" and the default for
+@code{service_name} is "Service01".
+
+@subsection Options
+
 The muxer options are:
 
 @table @option
@@ -558,10 +666,7 @@ Reemit PAT/PMT before writing the next packet.
 Use LATM packetization for AAC.
 @end table
 
-The recognized metadata settings in mpegts muxer are @code{service_provider}
-and @code{service_name}. If they are not set the default for
-@code{service_provider} is "FFmpeg" and the default for
-@code{service_name} is "Service01".
+@subsection Example
 
 @example
 ffmpeg -i file.mpg -c copy \
@@ -612,11 +717,12 @@ situations, giving a small seek granularity at the cost of additional container
 overhead.
 @end table
 
+@anchor{segment}
 @section segment, stream_segment, ssegment
 
 Basic stream segmenter.
 
-The segmenter muxer outputs streams to a number of separate files of nearly
+This muxer outputs streams to a number of separate files of nearly
 fixed duration. Output filename pattern can be set in a fashion similar to
 @ref{image2}.
 
@@ -638,14 +744,21 @@ The segment muxer works best with a single constant frame rate video.
 
 Optionally it can generate a list of the created segments, by setting
 the option @var{segment_list}. The list type is specified by the
-@var{segment_list_type} option.
+@var{segment_list_type} option. The entry filenames in the segment
+list are set by default to the basename of the corresponding segment
+files.
+
+See also the @ref{hls} muxer, which provides a more specific
+implementation for HLS segmentation.
+
+@subsection Options
 
 The segment muxer supports the following options:
 
 @table @option
 @item reference_stream @var{specifier}
 Set the reference stream, as specified by the string @var{specifier}.
-If @var{specifier} is set to @code{auto}, the reference is choosen
+If @var{specifier} is set to @code{auto}, the reference is chosen
 automatically. Otherwise it must be a stream specifier (see the ``Stream
 specifiers'' chapter in the ffmpeg manual) which specifies the
 reference stream. The default value is @code{auto}.
@@ -670,13 +783,15 @@ Allow caching (only affects M3U8 list files).
 Allow live-friendly file generation.
 @end table
 
-Default value is @code{samp}.
-
 @item segment_list_size @var{size}
 Update the list file so that it contains at most the last @var{size}
 segments. If 0 the list file will contain all the segments. Default
 value is 0.
 
+@item segment_list_entry_prefix @var{prefix}
+Set @var{prefix} to prepend to the name of each entry filename. By
+default no prefix is applied.
+
 @item segment_list_type @var{type}
 Specify the format for the segment list file.
 
@@ -748,7 +863,7 @@ In particular may be used in combination with the @file{ffmpeg} option
 @var{force_key_frames} may not be set accurately because of rounding
 issues, with the consequence that a key frame time may result set just
 before the specified time. For constant frame rate videos a value of
-1/2*@var{frame_rate} should address the worst case mismatch between
+1/(2*@var{frame_rate}) should address the worst case mismatch between
 the specified time and the time set by @var{force_key_frames}.
 
 @item segment_times @var{times}
@@ -849,8 +964,8 @@ to feed the same packets to several muxers directly.
 The slave outputs are specified in the file name given to the muxer,
 separated by '|'. If any of the slave name contains the '|' separator,
 leading or trailing spaces or any special character, it must be
-escaped (see the ``Quoting and escaping'' section in the ffmpeg-utils
-manual).
+escaped (see @ref{quoting_and_escaping,,the "Quoting and escaping"
+section in the ffmpeg-utils(1) manual,ffmpeg-utils}).
 
 Muxer options can be specified for each slave by prepending them as a list of
 @var{key}=@var{value} pairs separated by ':', between square brackets. If
@@ -865,9 +980,12 @@ output name suffix.
 
 @item bsfs[/@var{spec}]
 Specify a list of bitstream filters to apply to the specified
-output. It is possible to specify to which streams a given bitstream
-filter applies, by appending a stream specifier to the option
-separated by @code{/}. If the stream specifier is not specified, the
+output.
+
+It is possible to specify to which streams a given bitstream filter
+applies, by appending a stream specifier to the option separated by
+@code{/}. @var{spec} must be a stream specifier (see @ref{Format
+stream specifiers}).  If the stream specifier is not specified, the
 bistream filters will be applied to all streams in the output.
 
 Several bitstream filters can be specified, separated by ",".
@@ -878,7 +996,8 @@ specified by a stream specifier. If not specified, this defaults to
 all the input streams.
 @end table
 
-Some examples follow.
+@subsection Examples
+
 @itemize
 @item
 Encode something and both archive it in a WebM file and stream it
@@ -899,6 +1018,15 @@ audio packets.
 ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental
        -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
 @end example
+
+@item
+As below, but select only stream @code{a:1} for the audio output. Note
+that a second level escaping must be performed, as ":" is a special
+character used to separate options.
+@example
+ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental
+       -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"
+@end example
 @end itemize
 
 Note: some codecs may need different options depending on the output format;

doc/outdevs.texi

@@ -22,11 +22,27 @@ A description of the currently available output devices follows.
 
 ALSA (Advanced Linux Sound Architecture) output device.
 
+@subsection Examples
+
+@itemize
+@item
+Play a file on default ALSA device:
+@example
+ffmpeg -i INPUT -f alsa default
+@end example
+
+@item
+Play a file on soundcard 1, audio device 7:
+@example
+ffmpeg -i INPUT -f alsa hw:1,7
+@end example
+@end itemize
+
 @section caca
 
 CACA output device.
 
-This output device allows to show a video stream in CACA window.
+This output device allows one to show a video stream in CACA window.
 Only one CACA window is allowed per application, so you can
 have only one instance of this output device in an application.
 
@@ -104,6 +120,68 @@ ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors -
 @end example
 @end itemize
 
+@section decklink
+
+The decklink output device provides playback capabilities for Blackmagic
+DeckLink devices.
+
+To enable this output device, you need the Blackmagic DeckLink SDK and you
+need to configure with the appropriate @code{--extra-cflags}
+and @code{--extra-ldflags}.
+On Windows, you need to run the IDL files through @command{widl}.
+
+DeckLink is very picky about the formats it supports. Pixel format is always
+uyvy422, framerate and video size must be determined for your device with
+@command{-list_formats 1}. Audio sample rate is always 48 kHz.
+
+@subsection Options
+
+@table @option
+
+@item list_devices
+If set to @option{true}, print a list of devices and exit.
+Defaults to @option{false}.
+
+@item list_formats
+If set to @option{true}, print a list of supported formats and exit.
+Defaults to @option{false}.
+
+@item preroll
+Amount of time to preroll video in seconds.
+Defaults to @option{0.5}.
+
+@end table
+
+@subsection Examples
+
+@itemize
+
+@item
+List output devices:
+@example
+ffmpeg -i test.avi -f decklink -list_devices 1 dummy
+@end example
+
+@item
+List supported formats:
+@example
+ffmpeg -i test.avi -f decklink -list_formats 1 'DeckLink Mini Monitor'
+@end example
+
+@item
+Play video clip:
+@example
+ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 'DeckLink Mini Monitor'
+@end example
+
+@item
+Play video clip with non-standard framerate or video size:
+@example
+ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLink Mini Monitor'
+@end example
+
+@end itemize
+
 @section fbdev
 
 Linux framebuffer output device.
@@ -133,6 +211,41 @@ ffmpeg -re -i INPUT -vcodec rawvideo -pix_fmt bgra -f fbdev /dev/fb0
 
 See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
 
+@section opengl
+OpenGL output device.
+
+To enable this output device you need to configure FFmpeg with @code{--enable-opengl}.
+
+This output device allows one to render to OpenGL context.
+Context may be provided by application or default SDL window is created.
+
+When device renders to external context, application must implement handlers for following messages:
+@code{AV_CTL_MESSAGE_CREATE_WINDOW_BUFFER} - create OpenGL context on current thread.
+@code{AV_CTL_MESSAGE_PREPARE_WINDOW_BUFFER} - make OpenGL context current.
+@code{AV_CTL_MESSAGE_DISPLAY_WINDOW_BUFFER} - swap buffers.
+@code{AV_CTL_MESSAGE_DESTROY_WINDOW_BUFFER} - destroy OpenGL context.
+Application is also required to inform a device about current resolution by sending @code{AV_DEVICE_WINDOW_RESIZED} message.
+
+@subsection Options
+@table @option
+
+@item background
+Set background color. Black is a default.
+@item no_window
+Disables default SDL window when set to non-zero value.
+Application must provide OpenGL context and both @code{window_size_cb} and @code{window_swap_buffers_cb} callbacks when set.
+@item window_title
+Set the SDL window title, if not specified default to the filename specified for the output device.
+Ignored when @option{no_window} is set.
+
+@end table
+
+@subsection Examples
+Play a file on SDL window using OpenGL rendering:
+@example
+ffmpeg  -i INPUT -f opengl "window title"
+@end example
+
 @section oss
 
 OSS (Open Sound System) output device.
@@ -164,6 +277,19 @@ by default it is set to the specified output name.
 Specify the device to use. Default device is used when not provided.
 List of output devices can be obtained with command @command{pactl list sinks}.
 
+@item buffer_size
+@item buffer_duration
+Control the size and duration of the PulseAudio buffer. A small buffer
+gives more control, but requires more frequent updates.
+
+@option{buffer_size} specifies size in bytes while
+@option{buffer_duration} specifies duration in milliseconds.
+
+When both options are provided then the highest value is used
+(duration is recalculated to bytes using stream parameters). If they
+are set to 0 (which is default), the device will use the default
+PulseAudio duration value. By default PulseAudio set buffer duration
+to around 2 seconds.
 @end table
 
 @subsection Examples
@@ -176,7 +302,7 @@ ffmpeg  -i INPUT -f pulse "stream name"
 
 SDL (Simple DirectMedia Layer) output device.
 
-This output device allows to show a video stream in an SDL
+This output device allows one to show a video stream in an SDL
 window. Only one SDL window is allowed per application, so you can
 have only one instance of this output device in an application.
 
@@ -206,7 +332,17 @@ downscaled according to the aspect ratio.
 
 @item window_fullscreen
 Set fullscreen mode when non-zero value is provided.
-Zero is a default.
+Default value is zero.
+@end table
+
+@subsection Interactive commands
+
+The window created by the device can be controlled through the
+following interactive commands.
+
+@table @key
+@item q, ESC
+Quit the device immediately.
 @end table
 
 @subsection Examples
@@ -225,7 +361,7 @@ sndio audio output device.
 
 XV (XVideo) output device.
 
-This output device allows to show a video stream in a X Window System
+This output device allows one to show a video stream in a X Window System
 window.
 
 @subsection Options

doc/platform.texi

@@ -51,8 +51,9 @@ The toolchain provided with Xcode is sufficient to build the basic
 unacelerated code.
 
 Mac OS X on PowerPC or ARM (iPhone) requires a preprocessor from
+@url{https://github.com/FFmpeg/gas-preprocessor} or
 @url{http://github.com/yuvi/gas-preprocessor} to build the optimized
-assembler functions. Just download the Perl script and put it somewhere
+assembler functions. Put the Perl script somewhere
 in your PATH, FFmpeg's configure will pick it up automatically.
 
 Mac OS X on amd64 and x86 requires @command{yasm} to build most of the
@@ -108,14 +109,16 @@ libavformat) as DLLs.
 
 @section Microsoft Visual C++ or Intel C++ Compiler for Windows
 
-FFmpeg can be built with MSVC or ICL using a C99-to-C89 conversion utility and
-wrapper. For ICL, only the wrapper is used, since ICL supports C99.
+FFmpeg can be built with MSVC 2012 or earlier using a C99-to-C89 conversion utility
+and wrapper, or with MSVC 2013 and ICL natively.
 
 You will need the following prerequisites:
 
 @itemize
-@item @uref{http://download.videolan.org/pub/contrib/c99-to-c89/, C99-to-C89 Converter & Wrapper}
+@item @uref{https://github.com/libav/c99-to-c89/, C99-to-C89 Converter & Wrapper}
+(if using MSVC 2012 or earlier)
 @item @uref{http://code.google.com/p/msinttypes/, msinttypes}
+(if using MSVC 2012 or earlier)
 @item @uref{http://www.mingw.org/, MSYS}
 @item @uref{http://yasm.tortall.net/, YASM}
 @item @uref{http://gnuwin32.sourceforge.net/packages/bc.htm, bc for Windows} if
@@ -125,14 +128,16 @@ you want to run @uref{fate.html, FATE}.
 To set up a proper environment in MSYS, you need to run @code{msys.bat} from
 the Visual Studio or Intel Compiler command prompt.
 
-Place @code{makedef}, @code{c99wrap.exe}, @code{c99conv.exe}, and @code{yasm.exe}
-somewhere in your @code{PATH}.
+Place @code{yasm.exe} somewhere in your @code{PATH}. If using MSVC 2012 or
+earlier, place @code{c99wrap.exe} and @code{c99conv.exe} somewhere in your
+@code{PATH} as well.
 
-Next, make sure @code{inttypes.h} and any other headers and libs you want to use
-are located in a spot that the compiler can see. Do so by modifying the @code{LIB}
-and @code{INCLUDE} environment variables to include the @strong{Windows} paths to
-these directories. Alternatively, you can try and use the
-@code{--extra-cflags}/@code{--extra-ldflags} configure options.
+Next, make sure any other headers and libs you want to use, such as zlib, are
+located in a spot that the compiler can see. Do so by modifying the @code{LIB}
+and @code{INCLUDE} environment variables to include the @strong{Windows-style}
+paths to these directories. Alternatively, you can try and use the
+@code{--extra-cflags}/@code{--extra-ldflags} configure options. If using MSVC
+2012 or earlier, place @code{inttypes.h} somewhere the compiler can see too.
 
 Finally, run:
 
@@ -182,7 +187,9 @@ can see.
 @itemize
 @item Visual Studio 2010 Pro and Express
 @item Visual Studio 2012 Pro and Express
+@item Visual Studio 2013 Pro and Express
 @item Intel Composer XE 2013
+@item Intel Composer XE 2013 SP1
 @end itemize
 Anything else is not officially supported.
 

doc/protocols.texi

@@ -117,7 +117,19 @@ ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP/////////////
 
 File access protocol.
 
-Allow to read from or read to a file.
+Allow to read from or write to a file.
+
+A file URL can have the form:
+@example
+file:@var{filename}
+@end example
+
+where @var{filename} is the path of the file to read.
+
+An URL that does not have a protocol prefix will be assumed to be a
+file URL. Depending on the build, an URL that looks like a Windows
+path with the drive letter at the beginning will also be assumed to be
+a file URL (usually not the case in builds for unix-like systems).
 
 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
 use the command:
@@ -125,10 +137,6 @@ use the command:
 ffmpeg -i file:input.mpeg output.mpeg
 @end example
 
-The ff* tools default to the file protocol, that is a resource
-specified with the name "FILE.mpeg" is interpreted as the URL
-"file:FILE.mpeg".
-
 This protocol accepts the following options:
 
 @table @option
@@ -205,7 +213,7 @@ m3u8 files.
 
 HTTP (Hyper Text Transfer Protocol).
 
-This protocol accepts the following options.
+This protocol accepts the following options:
 
 @table @option
 @item seekable
@@ -215,32 +223,33 @@ if set to -1 it will try to autodetect if it is seekable. Default
 value is -1.
 
 @item chunked_post
-If set to 1 use chunked transfer-encoding for posts, default is 1.
+If set to 1 use chunked Transfer-Encoding for posts, default is 1.
+
+@item content_type
+Set a specific content type for the POST messages.
 
 @item headers
 Set custom HTTP headers, can override built in default headers. The
 value must be a string encoding the headers.
 
-@item content_type
-Force a content type.
-
-@item user-agent
-Override User-Agent header. If not specified the protocol will use a
-string describing the libavformat build.
-
 @item multiple_requests
-Use persistent connections if set to 1. By default it is 0.
+Use persistent connections if set to 1, default is 0.
 
 @item post_data
 Set custom HTTP post data.
 
+@item user-agent
+@item user_agent
+Override the User-Agent header. If not specified the protocol will use a
+string describing the libavformat build. ("Lavf/<version>")
+
 @item timeout
 Set timeout of socket I/O operations used by the underlying low level
 operation. By default it is set to -1, which means that the timeout is
 not specified.
 
 @item mime_type
-Set MIME type.
+Export the MIME type.
 
 @item icy
 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
@@ -249,17 +258,25 @@ the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
 The default is 0.
 
 @item icy_metadata_headers
-If the server supports ICY metadata, this contains the ICY specific HTTP reply
-headers, separated with newline characters.
+If the server supports ICY metadata, this contains the ICY-specific HTTP reply
+headers, separated by newline characters.
 
 @item icy_metadata_packet
 If the server supports ICY metadata, and @option{icy} was set to 1, this
-contains the last non-empty metadata packet sent by the server.
+contains the last non-empty metadata packet sent by the server. It should be
+polled in regular intervals by applications interested in mid-stream metadata
+updates.
 
 @item cookies
 Set the cookies to be sent in future requests. The format of each cookie is the
 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
 delimited by a newline character.
+
+@item offset
+Set initial byte offset.
+
+@item end_offset
+Try to limit the request to bytes preceding this offset.
 @end table
 
 @subsection HTTP Cookies
@@ -544,6 +561,10 @@ is not specified.
 Truncate existing files on write, if set to 1. A value of 0 prevents
 truncating. Default value is 1.
 
+@item private_key
+Specify the path of the file containing private key to use during authorization.
+By default libssh searches for keys in the @file{~/.ssh/} directory.
+
 @end table
 
 Example: Play a file stored on remote server.
@@ -661,6 +682,8 @@ set to the the local RTP port value plus 1.
 
 @section rtsp
 
+Real-Time Streaming Protocol.
+
 RTSP is not technically a protocol handler in libavformat, it is a demuxer
 and muxer. The demuxer supports both normal RTSP (with data transferred
 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
@@ -675,14 +698,22 @@ The required syntax for a RTSP url is:
 rtsp://@var{hostname}[:@var{port}]/@var{path}
 @end example
 
-The following options (set on the @command{ffmpeg}/@command{ffplay} command
-line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
-are supported:
+Options can be set on the @command{ffmpeg}/@command{ffplay} command
+line, or set in code via @code{AVOption}s or in
+@code{avformat_open_input}.
 
-Flags for @code{rtsp_transport}:
+The following options are supported.
 
 @table @option
+@item initial_pause
+Do not start playing the stream immediately if set to 1. Default value
+is 0.
 
+@item rtsp_transport
+Set RTSP trasport protocols.
+
+It accepts the following values:
+@table @samp
 @item udp
 Use UDP as lower transport protocol.
 
@@ -700,17 +731,56 @@ passing proxies.
 
 Multiple lower transport protocols may be specified, in that case they are
 tried one at a time (if the setup of one fails, the next one is tried).
-For the muxer, only the @code{tcp} and @code{udp} options are supported.
+For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
 
-Flags for @code{rtsp_flags}:
+@item rtsp_flags
+Set RTSP flags.
 
-@table @option
+The following values are accepted:
+@table @samp
 @item filter_src
 Accept packets only from negotiated peer address and port.
 @item listen
 Act as a server, listening for an incoming connection.
 @end table
 
+Default value is @samp{none}.
+
+@item allowed_media_types
+Set media types to accept from the server.
+
+The following flags are accepted:
+@table @samp
+@item video
+@item audio
+@item data
+@end table
+
+By default it accepts all media types.
+
+@item min_port
+Set minimum local UDP port. Default value is 5000.
+
+@item max_port
+Set maximum local UDP port. Default value is 65000.
+
+@item timeout
+Set maximum timeout (in seconds) to wait for incoming connections.
+
+A value of -1 mean infinite (default). This option implies the
+@option{rtsp_flags} set to @samp{listen}.
+
+@item reorder_queue_size
+Set number of packets to buffer for handling of reordered packets.
+
+@item stimeout
+Set socket TCP I/O timeout in micro seconds.
+
+@item user-agent
+Override User-Agent header. If not specified, it default to the
+libavformat identifier string.
+@end table
+
 When receiving data over UDP, the demuxer tries to reorder received packets
 (since they may arrive out of order, or packets may get lost totally). This
 can be disabled by setting the maximum demuxing delay to zero (via
@@ -721,36 +791,36 @@ streams to display can be chosen with @code{-vst} @var{n} and
 @code{-ast} @var{n} for video and audio respectively, and can be switched
 on the fly by pressing @code{v} and @code{a}.
 
-Example command lines:
+@subsection Examples
 
-To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
+The following examples all make use of the @command{ffplay} and
+@command{ffmpeg} tools.
 
+@itemize
+@item
+Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
 @example
 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
 @end example
 
-To watch a stream tunneled over HTTP:
-
+@item
+Watch a stream tunneled over HTTP:
 @example
 ffplay -rtsp_transport http rtsp://server/video.mp4
 @end example
 
-To send a stream in realtime to a RTSP server, for others to watch:
-
+@item
+Send a stream in realtime to a RTSP server, for others to watch:
 @example
 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
 @end example
 
-To receive a stream in realtime:
-
+@item
+Receive a stream in realtime:
 @example
 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
 @end example
-
-@table @option
-@item stimeout
-Socket IO timeout in micro seconds.
-@end table
+@end itemize
 
 @section sap
 
@@ -890,30 +960,39 @@ used as master salt.
 
 @section tcp
 
-Trasmission Control Protocol.
+Transmission Control Protocol.
 
 The required syntax for a TCP url is:
 @example
 tcp://@var{hostname}:@var{port}[?@var{options}]
 @end example
 
-@table @option
+@var{options} contains a list of &-separated options of the form
+@var{key}=@var{val}.
 
-@item listen
-Listen for an incoming connection
+The list of supported options follows.
+
+@table @option
+@item listen=@var{1|0}
+Listen for an incoming connection. Default value is 0.
 
 @item timeout=@var{microseconds}
-In read mode: if no data arrived in more than this time interval, raise error.
-In write mode: if socket cannot be written in more than this time interval, raise error.
-This also sets timeout on TCP connection establishing.
+Set raise error timeout, expressed in microseconds.
 
+This option is only relevant in read mode: if no data arrived in more
+than this time interval, raise error.
+
+@item listen_timeout=@var{microseconds}
+Set listen timeout, expressed in microseconds.
+@end table
+
+The following example shows how to setup a listening TCP connection
+with @command{ffmpeg}, which is then accessed with @command{ffplay}:
 @example
 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
 ffplay tcp://@var{hostname}:@var{port}
 @end example
 
-@end table
-
 @section tls
 
 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
@@ -979,7 +1058,7 @@ ffplay tls://@var{hostname}:@var{port}
 
 User Datagram Protocol.
 
-The required syntax for a UDP url is:
+The required syntax for an UDP URL is:
 @example
 udp://@var{hostname}:@var{port}[?@var{options}]
 @end example
@@ -987,14 +1066,13 @@ udp://@var{hostname}:@var{port}[?@var{options}]
 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
 
 In case threading is enabled on the system, a circular buffer is used
-to store the incoming data, which allows to reduce loss of data due to
+to store the incoming data, which allows one to reduce loss of data due to
 UDP socket buffer overruns. The @var{fifo_size} and
 @var{overrun_nonfatal} options are related to this buffer.
 
 The list of supported options follows.
 
 @table @option
-
 @item buffer_size=@var{size}
 Set the UDP socket buffer size in bytes. This is used both for the
 receiving and the sending buffer size.
@@ -1044,25 +1122,34 @@ Survive in case of UDP receiving circular buffer overrun. Default
 value is 0.
 
 @item timeout=@var{microseconds}
-In read mode: if no data arrived in more than this time interval, raise error.
+Set raise error timeout, expressed in microseconds.
+
+This option is only relevant in read mode: if no data arrived in more
+than this time interval, raise error.
 @end table
 
-Some usage examples of the UDP protocol with @command{ffmpeg} follow.
+@subsection Examples
 
-To stream over UDP to a remote endpoint:
+@itemize
+@item
+Use @command{ffmpeg} to stream over UDP to a remote endpoint:
 @example
 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
 @end example
 
-To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
+@item
+Use @command{ffmpeg} to stream in mpegts format over UDP using 188
+sized UDP packets, using a large input buffer:
 @example
 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
 @end example
 
-To receive over UDP from a remote endpoint:
+@item
+Use @command{ffmpeg} to receive over UDP from a remote endpoint:
 @example
-ffmpeg -i udp://[@var{multicast-address}]:@var{port}
+ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
 @end example
+@end itemize
 
 @section unix
 

doc/scaler.texi

@@ -35,7 +35,7 @@ Select nearest neighbor rescaling algorithm.
 @item area
 Select averaging area rescaling algorithm.
 
-@item bicubiclin
+@item bicublin
 Select bicubic scaling algorithm for the luma component, bilinear for
 chroma components.
 

doc/soc.txt

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

doc/texi2pod.pl

@@ -1,4 +1,4 @@
-#! /usr/bin/perl
+#!/usr/bin/env perl
 
 #   Copyright (C) 1999, 2000, 2001 Free Software Foundation, Inc.
 
@@ -327,6 +327,9 @@ die "No filename or title\n" unless defined $fn && defined $tl;
 $chapters{NAME} = "$fn \- $tl\n";
 $chapters{FOOTNOTES} .= "=back\n" if exists $chapters{FOOTNOTES};
 
+# always use utf8
+print "=encoding utf8\n\n";
+
 unshift @chapters_sequence, "NAME";
 for $chapter (@chapters_sequence) {
     if (exists $chapters{$chapter}) {
@@ -377,8 +380,8 @@ sub postprocess
     s/\(?\@xref\{(?:[^\}]*)\}(?:[^.<]|(?:<[^<>]*>))*\.\)?//g;
     s/\s+\(\@pxref\{(?:[^\}]*)\}\)//g;
     s/;\s+\@pxref\{(?:[^\}]*)\}//g;
-    s/\@ref\{(?:[^,\}]*,)(?:[^,\}]*,)([^,\}]*).*\}/$1/g;
-    s/\@ref\{([^\}]*)\}/$1/g;
+    s/\@ref\{(?:[^,\}]*,)(?:[^,\}]*,)([^,\}]*).*\}/B<$1>/g;
+    s/\@ref\{([^\}]*)\}/B<$1>/g;
     s/\@noindent\s*//g;
     s/\@refill//g;
     s/\@gol//g;

doc/texidep.pl

@@ -0,0 +1,32 @@
+#! /usr/bin/env perl
+
+# This script will print the dependency of a Texinfo file to stdout.
+# texidep.pl <src-path> <input.texi> <output.ext>
+
+use warnings;
+use strict;
+
+die unless @ARGV == 3;
+
+my ($src_path, $root, $target) = @ARGV;
+
+sub print_deps {
+    my ($file, $deps) = @_;
+    $deps->{$file} = 1;
+
+    open(my $fh, "<", "$file") or die "Cannot open file '$file': $!";
+    while (<$fh>) {
+        if (my ($i) = /^\@(?:verbatim)?include\s+(\S+)/) {
+            die "Circular dependency found in file $root\n" if exists $deps->{"doc/$1"};
+            print "$target: doc/$1\n";
+
+            # skip looking for config.texi dependencies, since it has
+            # none, and is not located in the source tree
+            if ("$1" ne "config.texi") {
+                print_deps("$src_path/doc/$1", {%$deps});
+            }
+        }
+    }
+}
+
+print_deps($root, {});

doc/utils.texi

@@ -1051,13 +1051,13 @@ See reference "OpenCL Specification Version: 1.2 chapter 5.6.4".
 Select the index of the platform to run OpenCL code.
 
 The specified index must be one of the indexes in the device list
-which can be obtained with @code{av_opencl_get_device_list()}.
+which can be obtained with @code{ffmpeg -opencl_bench} or @code{av_opencl_get_device_list()}.
 
 @item device_idx
 Select the index of the device used to run OpenCL code.
 
-The specifed index must be one of the indexes in the device list which
-can be obtained with @code{av_opencl_get_device_list()}.
+The specified index must be one of the indexes in the device list which
+can be obtained with @code{ffmpeg -opencl_bench} or @code{av_opencl_get_device_list()}.
 
 @end table
 

doc/viterbi.txt

@@ -1,109 +0,0 @@
-This is a quick description of the viterbi aka dynamic programing
-algorthm.
-
-Its reason for existence is that wikipedia has become very poor on
-describing algorithms in a way that makes it useable for understanding
-them or anything else actually. It tends now to describe the very same
-algorithm under 50 different names and pages with few understandable
-by even people who fully understand the algorithm and the theory behind.
-
-Problem description: (that is what it can solve)
-assume we have a 2d table, or you could call it a graph or matrix if you
-prefer
-
-    O   O   O   O   O   O   O
-
-    O   O   O   O   O   O   O
-
-    O   O   O   O   O   O   O
-
-    O   O   O   O   O   O   O
-
-
-That table has edges connecting points from each column to the next column
-and each edge has a score like: (only some edge and scores shown to keep it
-readable)
-
-
-    O--5--O-----O-----O-----O-----O
-     2   / 7   / \   / \   / \   /
-      \ /   \ /   \ /   \ /   \ /
-    O7-/--O--/--O--/--O--/--O--/--O
-     \/ \/ 1/ \/ \/ \/ \/ \/ \/ \/
-     /\ /\ 2\ /\ /\ /\ /\ /\ /\ /\
-    O3-/--O--/--O--/--O--/--O--/--O
-      / \   / \   / \   / \   / \
-     1   \ 9   \ /   \ /   \ /   \
-    O--2--O--1--O--5--O--3--O--8--O
-
-
-
-Our goal is to find a path from left to right through it which
-minimizes the sum of the score of all edges.
-(and of course left/right is just a convention here it could be top down too)
-Similarly the minimum could be the maximum by just fliping the sign,
-Example of a path with scores:
-
-    O   O   O   O   O   O   O
-
->---O.  O   O  .O-2-O   O   O
-      5.     .7      .
-    O   O-1-O   O   O 8 O   O
-                       .
-    O   O   O   O   O   O-1-O---> (sum here is 24)
-
-
-The viterbi algorthm now solves this simply column by column
-For the previous column each point has a best path and a associated
-score:
-
-    O-----5     O
-     \
-      \
-    O  \  1     O
-        \/
-        /\
-    O  /  2     O
-      /
-     /
-    O-----2     O
-
-
-To move one column forward we just need to find the best path and associated
-scores for the next column
-here are some edges we could choose from:
-
-
-    O-----5--3--O
-     \      \8
-      \       \
-    O  \  1--9--O
-        \/  \3
-        /\     \
-    O  /  2--1--O
-      /     \2
-     /        \
-    O-----2--4--O
-
-Finding the new best paths and scores for each point of our new column is
-trivial given we know the previous column best paths and scores:
-
-    O-----0-----8
-     \
-      \
-    O  \  0----10
-        \/
-        /\
-    O  /  0-----3
-      /     \
-     /        \
-    O     0     4
-
-
-the viterbi algorthm continues exactly like this column for column until the
-end and then just picks the path with the best score (above that would be the
-one with score 3)
-
-
-Author: Michael niedermayer
-Copyright LGPL

ffmpeg.c

@@ -30,6 +30,8 @@
 #include <stdlib.h>
 #include <errno.h>
 #include <limits.h>
+#include <stdint.h>
+
 #if HAVE_ISATTY
 #if HAVE_IO_H
 #include <io.h>
@@ -38,9 +40,9 @@
 #include <unistd.h>
 #endif
 #endif
+
 #include "libavformat/avformat.h"
 #include "libavdevice/avdevice.h"
-#include "libswscale/swscale.h"
 #include "libswresample/swresample.h"
 #include "libavutil/opt.h"
 #include "libavutil/channel_layout.h"
@@ -123,6 +125,7 @@ static int64_t getmaxrss(void);
 static int run_as_daemon  = 0;
 static int64_t video_size = 0;
 static int64_t audio_size = 0;
+static int64_t data_size = 0;
 static int64_t subtitle_size = 0;
 static int64_t extra_size = 0;
 static int nb_frames_dup = 0;
@@ -311,6 +314,7 @@ void term_exit(void)
 
 static volatile int received_sigterm = 0;
 static volatile int received_nb_signals = 0;
+static int main_return_code = 0;
 
 static void
 sigterm_handler(int sig)
@@ -319,7 +323,7 @@ sigterm_handler(int sig)
     received_nb_signals++;
     term_exit();
     if(received_nb_signals > 3)
-        exit_program(123);
+        exit(123);
 }
 
 void term_init(void)
@@ -466,7 +470,9 @@ static void ffmpeg_cleanup(int ret)
             bsfc = next;
         }
         output_streams[i]->bitstream_filters = NULL;
-        avcodec_free_frame(&output_streams[i]->filtered_frame);
+        av_frame_free(&output_streams[i]->filtered_frame);
+
+        av_parser_close(output_streams[i]->parser);
 
         av_freep(&output_streams[i]->forced_keyframes);
         av_expr_free(output_streams[i]->forced_keyframes_pexpr);
@@ -488,6 +494,7 @@ static void ffmpeg_cleanup(int ret)
         avsubtitle_free(&input_streams[i]->prev_sub.subtitle);
         av_frame_free(&input_streams[i]->sub2video.frame);
         av_freep(&input_streams[i]->filters);
+        av_freep(&input_streams[i]->hwaccel_device);
         av_freep(&input_streams[i]);
     }
 
@@ -507,6 +514,8 @@ static void ffmpeg_cleanup(int ret)
     if (received_sigterm) {
         av_log(NULL, AV_LOG_INFO, "Received signal %d: terminating.\n",
                (int) received_sigterm);
+    } else if (ret) {
+        av_log(NULL, AV_LOG_INFO, "Conversion failed!\n");
     }
     term_exit();
 }
@@ -542,6 +551,15 @@ static void update_benchmark(const char *fmt, ...)
     }
 }
 
+static void close_all_output_streams(OutputStream *ost, OSTFinished this_stream, OSTFinished others)
+{
+    int i;
+    for (i = 0; i < nb_output_streams; i++) {
+        OutputStream *ost2 = output_streams[i];
+        ost2->finished |= ost == ost2 ? this_stream : others;
+    }
+}
+
 static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
 {
     AVBitStreamFilterContext *bsfc = ost->bitstream_filters;
@@ -642,7 +660,8 @@ static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
     ret = av_interleaved_write_frame(s, pkt);
     if (ret < 0) {
         print_error("av_interleaved_write_frame()", ret);
-        exit_program(1);
+        main_return_code = 1;
+        close_all_output_streams(ost, MUXER_FINISHED | ENCODER_FINISHED, ENCODER_FINISHED);
     }
 }
 
@@ -650,7 +669,7 @@ static void close_output_stream(OutputStream *ost)
 {
     OutputFile *of = output_files[ost->file_index];
 
-    ost->finished = 1;
+    ost->finished |= ENCODER_FINISHED;
     if (of->shortest) {
         int64_t end = av_rescale_q(ost->sync_opts - ost->first_pts, ost->st->codec->time_base, AV_TIME_BASE_Q);
         of->recording_time = FFMIN(of->recording_time, end);
@@ -690,6 +709,12 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost,
 
     av_assert0(pkt.size || !pkt.data);
     update_benchmark(NULL);
+    if (debug_ts) {
+        av_log(NULL, AV_LOG_INFO, "encoder <- type:audio "
+               "frame_pts:%s frame_pts_time:%s time_base:%d/%d\n",
+               av_ts2str(frame->pts), av_ts2timestr(frame->pts, &enc->time_base),
+               enc->time_base.num, enc->time_base.den);
+    }
     if (avcodec_encode_audio2(enc, &pkt, frame, &got_packet) < 0) {
         av_log(NULL, AV_LOG_FATAL, "Audio encoding failed (avcodec_encode_audio2)\n");
         exit_program(1);
@@ -817,10 +842,29 @@ static void do_video_out(AVFormatContext *s,
     nb_frames = 1;
 
     format_video_sync = video_sync_method;
-    if (format_video_sync == VSYNC_AUTO)
-        format_video_sync = (s->oformat->flags & AVFMT_VARIABLE_FPS) ? ((s->oformat->flags & AVFMT_NOTIMESTAMPS) ? VSYNC_PASSTHROUGH : VSYNC_VFR) : VSYNC_CFR;
+    if (format_video_sync == VSYNC_AUTO) {
+        if(!strcmp(s->oformat->name, "avi")) {
+            format_video_sync = VSYNC_VFR;
+        } else
+            format_video_sync = (s->oformat->flags & AVFMT_VARIABLE_FPS) ? ((s->oformat->flags & AVFMT_NOTIMESTAMPS) ? VSYNC_PASSTHROUGH : VSYNC_VFR) : VSYNC_CFR;
+        if (   ist
+            && format_video_sync == VSYNC_CFR
+            && input_files[ist->file_index]->ctx->nb_streams == 1
+            && input_files[ist->file_index]->input_ts_offset == 0) {
+            format_video_sync = VSYNC_VSCFR;
+        }
+        if (format_video_sync == VSYNC_CFR && copy_ts) {
+            format_video_sync = VSYNC_VSCFR;
+        }
+    }
 
     switch (format_video_sync) {
+    case VSYNC_VSCFR:
+        if (ost->frame_number == 0 && delta - duration >= 0.5) {
+            av_log(NULL, AV_LOG_DEBUG, "Not duplicating %d initial frames\n", (int)lrintf(delta - duration));
+            delta = duration;
+            ost->sync_opts = lrint(sync_ipts);
+        }
     case VSYNC_CFR:
         // FIXME set to 0.5 after we fix some dts/pts bugs like in avidec.c
         if (delta < -1.1)
@@ -945,6 +989,13 @@ static void do_video_out(AVFormatContext *s,
         }
 
         update_benchmark(NULL);
+        if (debug_ts) {
+            av_log(NULL, AV_LOG_INFO, "encoder <- type:video "
+                   "frame_pts:%s frame_pts_time:%s time_base:%d/%d\n",
+                   av_ts2str(in_picture->pts), av_ts2timestr(in_picture->pts, &enc->time_base),
+                   enc->time_base.num, enc->time_base.den);
+        }
+
         ret = avcodec_encode_video2(enc, &pkt, in_picture, &got_packet);
         update_benchmark("encode_video %d.%d", ost->file_index, ost->index);
         if (ret < 0) {
@@ -953,6 +1004,13 @@ static void do_video_out(AVFormatContext *s,
         }
 
         if (got_packet) {
+            if (debug_ts) {
+                av_log(NULL, AV_LOG_INFO, "encoder -> type:video "
+                       "pkt_pts:%s pkt_pts_time:%s pkt_dts:%s pkt_dts_time:%s\n",
+                       av_ts2str(pkt.pts), av_ts2timestr(pkt.pts, &enc->time_base),
+                       av_ts2str(pkt.dts), av_ts2timestr(pkt.dts, &enc->time_base));
+            }
+
             if (pkt.pts == AV_NOPTS_VALUE && !(enc->codec->capabilities & CODEC_CAP_DELAY))
                 pkt.pts = ost->sync_opts;
 
@@ -1049,19 +1107,21 @@ static int reap_filters(void)
     for (i = 0; i < nb_output_streams; i++) {
         OutputStream *ost = output_streams[i];
         OutputFile    *of = output_files[ost->file_index];
+        AVFilterContext *filter;
+        AVCodecContext *enc = ost->st->codec;
         int ret = 0;
 
         if (!ost->filter)
             continue;
+        filter = ost->filter->filter;
 
-        if (!ost->filtered_frame && !(ost->filtered_frame = avcodec_alloc_frame())) {
+        if (!ost->filtered_frame && !(ost->filtered_frame = av_frame_alloc())) {
             return AVERROR(ENOMEM);
-        } else
-            avcodec_get_frame_defaults(ost->filtered_frame);
+        }
         filtered_frame = ost->filtered_frame;
 
         while (1) {
-            ret = av_buffersink_get_frame_flags(ost->filter->filter, filtered_frame,
+            ret = av_buffersink_get_frame_flags(filter, filtered_frame,
                                                AV_BUFFERSINK_FLAG_NO_REQUEST);
             if (ret < 0) {
                 if (ret != AVERROR(EAGAIN) && ret != AVERROR_EOF) {
@@ -1070,32 +1130,38 @@ static int reap_filters(void)
                 }
                 break;
             }
+            if (ost->finished) {
+                av_frame_unref(filtered_frame);
+                continue;
+            }
             frame_pts = AV_NOPTS_VALUE;
             if (filtered_frame->pts != AV_NOPTS_VALUE) {
                 int64_t start_time = (of->start_time == AV_NOPTS_VALUE) ? 0 : of->start_time;
-                filtered_frame->pts = frame_pts = av_rescale_q(filtered_frame->pts,
-                                                ost->filter->filter->inputs[0]->time_base,
-                                                ost->st->codec->time_base) -
-                                    av_rescale_q(start_time,
-                                                AV_TIME_BASE_Q,
-                                                ost->st->codec->time_base);
+                filtered_frame->pts = frame_pts =
+                    av_rescale_q(filtered_frame->pts, filter->inputs[0]->time_base, enc->time_base) -
+                    av_rescale_q(start_time, AV_TIME_BASE_Q, enc->time_base);
             }
             //if (ost->source_index >= 0)
             //    *filtered_frame= *input_streams[ost->source_index]->decoded_frame; //for me_threshold
 
-
-            switch (ost->filter->filter->inputs[0]->type) {
+            switch (filter->inputs[0]->type) {
             case AVMEDIA_TYPE_VIDEO:
                 filtered_frame->pts = frame_pts;
                 if (!ost->frame_aspect_ratio.num)
-                    ost->st->codec->sample_aspect_ratio = filtered_frame->sample_aspect_ratio;
+                    enc->sample_aspect_ratio = filtered_frame->sample_aspect_ratio;
+
+                if (debug_ts) {
+                    av_log(NULL, AV_LOG_INFO, "filter -> pts:%s pts_time:%s time_base:%d/%d\n",
+                            av_ts2str(filtered_frame->pts), av_ts2timestr(filtered_frame->pts, &enc->time_base),
+                            enc->time_base.num, enc->time_base.den);
+                }
 
                 do_video_out(of->ctx, ost, filtered_frame);
                 break;
             case AVMEDIA_TYPE_AUDIO:
                 filtered_frame->pts = frame_pts;
-                if (!(ost->st->codec->codec->capabilities & CODEC_CAP_PARAM_CHANGE) &&
-                    ost->st->codec->channels != av_frame_get_channels(filtered_frame)) {
+                if (!(enc->codec->capabilities & CODEC_CAP_PARAM_CHANGE) &&
+                    enc->channels != av_frame_get_channels(filtered_frame)) {
                     av_log(NULL, AV_LOG_ERROR,
                            "Audio filter graph output is not normalized and encoder does not support parameter changes\n");
                     break;
@@ -1276,16 +1342,21 @@ static void print_report(int is_last_report, int64_t timer_start, int64_t cur_ti
     }
 
     if (is_last_report) {
-        int64_t raw= audio_size + video_size + subtitle_size + extra_size;
+        int64_t raw   = audio_size + video_size + data_size + subtitle_size + extra_size;
+        float percent = 0.0;
+
+        if (raw)
+            percent = 100.0 * (total_size - raw) / raw;
+
         av_log(NULL, AV_LOG_INFO, "\n");
-        av_log(NULL, AV_LOG_INFO, "video:%1.0fkB audio:%1.0fkB subtitle:%1.0f global headers:%1.0fkB muxing overhead %f%%\n",
+        av_log(NULL, AV_LOG_INFO, "video:%1.0fkB audio:%1.0fkB subtitle:%1.0f data:%1.0f global headers:%1.0fkB muxing overhead %f%%\n",
                video_size / 1024.0,
                audio_size / 1024.0,
                subtitle_size / 1024.0,
+               data_size / 1024.0,
                extra_size / 1024.0,
-               100.0 * (total_size - raw) / raw
-        );
-        if(video_size + audio_size + subtitle_size + extra_size == 0){
+               percent);
+        if(video_size + data_size + audio_size + subtitle_size + extra_size == 0){
             av_log(NULL, AV_LOG_WARNING, "Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)\n");
         }
     }
@@ -1331,6 +1402,7 @@ static void flush_encoders(void)
 
             if (encode) {
                 AVPacket pkt;
+                int pkt_size;
                 int got_packet;
                 av_init_packet(&pkt);
                 pkt.data = NULL;
@@ -1343,7 +1415,6 @@ static void flush_encoders(void)
                     av_log(NULL, AV_LOG_FATAL, "%s encoding failed\n", desc);
                     exit_program(1);
                 }
-                *size += pkt.size;
                 if (ost->logfile && enc->stats_out) {
                     fprintf(ost->logfile, "%s", enc->stats_out);
                 }
@@ -1351,15 +1422,21 @@ static void flush_encoders(void)
                     stop_encoding = 1;
                     break;
                 }
+                if (ost->finished & MUXER_FINISHED) {
+                    av_free_packet(&pkt);
+                    continue;
+                }
+                *size += pkt.size;
                 if (pkt.pts != AV_NOPTS_VALUE)
                     pkt.pts = av_rescale_q(pkt.pts, enc->time_base, ost->st->time_base);
                 if (pkt.dts != AV_NOPTS_VALUE)
                     pkt.dts = av_rescale_q(pkt.dts, enc->time_base, ost->st->time_base);
                 if (pkt.duration > 0)
                     pkt.duration = av_rescale_q(pkt.duration, enc->time_base, ost->st->time_base);
+                pkt_size = pkt.size;
                 write_frame(os, &pkt, ost);
                 if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && vstats_filename) {
-                    do_video_stats(ost, pkt.size);
+                    do_video_stats(ost, pkt_size);
                 }
             }
 
@@ -1380,6 +1457,9 @@ static int check_output_constraints(InputStream *ist, OutputStream *ost)
     if (ost->source_index != ist_index)
         return 0;
 
+    if (ost->finished)
+        return 0;
+
     if (of->start_time != AV_NOPTS_VALUE && ist->pts < of->start_time)
         return 0;
 
@@ -1434,6 +1514,8 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
     else if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
         video_size += pkt->size;
         ost->sync_opts++;
+    } else if (ost->st->codec->codec_type == AVMEDIA_TYPE_DATA) {
+        data_size += pkt->size;
     } else if (ost->st->codec->codec_type == AVMEDIA_TYPE_SUBTITLE) {
         subtitle_size += pkt->size;
     }
@@ -1467,7 +1549,10 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
        && ost->st->codec->codec_id != AV_CODEC_ID_MPEG2VIDEO
        && ost->st->codec->codec_id != AV_CODEC_ID_VC1
        ) {
-        if (av_parser_change(ist->st->parser, ost->st->codec, &opkt.data, &opkt.size, pkt->data, pkt->size, pkt->flags & AV_PKT_FLAG_KEY)) {
+        if (av_parser_change(ost->parser, ost->st->codec,
+                             &opkt.data, &opkt.size,
+                             pkt->data, pkt->size,
+                             pkt->flags & AV_PKT_FLAG_KEY)) {
             opkt.buf = av_buffer_create(opkt.data, opkt.size, av_buffer_default_free, NULL, 0);
             if (!opkt.buf)
                 exit_program(1);
@@ -1476,6 +1561,7 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
         opkt.data = pkt->data;
         opkt.size = pkt->size;
     }
+    av_copy_packet_side_data(&opkt, pkt);
 
     if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && (of->ctx->oformat->flags & AVFMT_RAWPICTURE)) {
         /* store AVPicture in AVPacket, as expected by the output format */
@@ -1516,7 +1602,7 @@ static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
     int i, ret, err = 0, resample_changed;
     AVRational decoded_frame_tb;
 
-    if (!ist->decoded_frame && !(ist->decoded_frame = avcodec_alloc_frame()))
+    if (!ist->decoded_frame && !(ist->decoded_frame = av_frame_alloc()))
         return AVERROR(ENOMEM);
     if (!ist->filter_frame && !(ist->filter_frame = av_frame_alloc()))
         return AVERROR(ENOMEM);
@@ -1652,7 +1738,6 @@ static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
 static int decode_video(InputStream *ist, AVPacket *pkt, int *got_output)
 {
     AVFrame *decoded_frame, *f;
-    void *buffer_to_free = NULL;
     int i, ret = 0, err = 0, resample_changed;
     int64_t best_effort_timestamp;
     AVRational *frame_sample_aspect;
@@ -1687,18 +1772,26 @@ static int decode_video(InputStream *ist, AVPacket *pkt, int *got_output)
     if(ist->top_field_first>=0)
         decoded_frame->top_field_first = ist->top_field_first;
 
+    if (ist->hwaccel_retrieve_data && decoded_frame->format == ist->hwaccel_pix_fmt) {
+        err = ist->hwaccel_retrieve_data(ist->st->codec, decoded_frame);
+        if (err < 0)
+            goto fail;
+    }
+    ist->hwaccel_retrieved_pix_fmt = decoded_frame->format;
+
     best_effort_timestamp= av_frame_get_best_effort_timestamp(decoded_frame);
     if(best_effort_timestamp != AV_NOPTS_VALUE)
         ist->next_pts = ist->pts = av_rescale_q(decoded_frame->pts = best_effort_timestamp, ist->st->time_base, AV_TIME_BASE_Q);
 
     if (debug_ts) {
         av_log(NULL, AV_LOG_INFO, "decoder -> ist_index:%d type:video "
-                "frame_pts:%s frame_pts_time:%s best_effort_ts:%"PRId64" best_effort_ts_time:%s keyframe:%d frame_type:%d \n",
-                ist->st->index, av_ts2str(decoded_frame->pts),
-                av_ts2timestr(decoded_frame->pts, &ist->st->time_base),
-                best_effort_timestamp,
-                av_ts2timestr(best_effort_timestamp, &ist->st->time_base),
-                decoded_frame->key_frame, decoded_frame->pict_type);
+               "frame_pts:%s frame_pts_time:%s best_effort_ts:%"PRId64" best_effort_ts_time:%s keyframe:%d frame_type:%d time_base:%d/%d\n",
+               ist->st->index, av_ts2str(decoded_frame->pts),
+               av_ts2timestr(decoded_frame->pts, &ist->st->time_base),
+               best_effort_timestamp,
+               av_ts2timestr(best_effort_timestamp, &ist->st->time_base),
+               decoded_frame->key_frame, decoded_frame->pict_type,
+               ist->st->time_base.num, ist->st->time_base.den);
     }
 
     pkt->size = 0;
@@ -1751,9 +1844,9 @@ static int decode_video(InputStream *ist, AVPacket *pkt, int *got_output)
         }
     }
 
+fail:
     av_frame_unref(ist->filter_frame);
     av_frame_unref(decoded_frame);
-    av_free(buffer_to_free);
     return err < 0 ? err : ret;
 }
 
@@ -1773,25 +1866,32 @@ static int transcode_subtitles(InputStream *ist, AVPacket *pkt, int *got_output)
     }
 
     if (ist->fix_sub_duration) {
+        int end = 1;
         if (ist->prev_sub.got_output) {
-            int end = av_rescale(subtitle.pts - ist->prev_sub.subtitle.pts,
-                                 1000, AV_TIME_BASE);
+            end = av_rescale(subtitle.pts - ist->prev_sub.subtitle.pts,
+                             1000, AV_TIME_BASE);
             if (end < ist->prev_sub.subtitle.end_display_time) {
                 av_log(ist->st->codec, AV_LOG_DEBUG,
-                       "Subtitle duration reduced from %d to %d\n",
-                       ist->prev_sub.subtitle.end_display_time, end);
+                       "Subtitle duration reduced from %d to %d%s\n",
+                       ist->prev_sub.subtitle.end_display_time, end,
+                       end <= 0 ? ", dropping it" : "");
                 ist->prev_sub.subtitle.end_display_time = end;
             }
         }
         FFSWAP(int,        *got_output, ist->prev_sub.got_output);
         FFSWAP(int,        ret,         ist->prev_sub.ret);
         FFSWAP(AVSubtitle, subtitle,    ist->prev_sub.subtitle);
+        if (end <= 0)
+            goto out;
     }
 
+    if (!*got_output)
+        return ret;
+
     sub2video_update(ist, &subtitle);
 
-    if (!*got_output || !subtitle.num_rects)
-        return ret;
+    if (!subtitle.num_rects)
+        goto out;
 
     for (i = 0; i < nb_output_streams; i++) {
         OutputStream *ost = output_streams[i];
@@ -1802,6 +1902,7 @@ static int transcode_subtitles(InputStream *ist, AVPacket *pkt, int *got_output)
         do_subtitle_out(output_files[ost->file_index]->ctx, ost, ist, &subtitle);
     }
 
+out:
     avsubtitle_free(&subtitle);
     return ret;
 }
@@ -1963,6 +2064,63 @@ static void print_sdp(void)
     av_freep(&avc);
 }
 
+static const HWAccel *get_hwaccel(enum AVPixelFormat pix_fmt)
+{
+    int i;
+    for (i = 0; hwaccels[i].name; i++)
+        if (hwaccels[i].pix_fmt == pix_fmt)
+            return &hwaccels[i];
+    return NULL;
+}
+
+static enum AVPixelFormat get_format(AVCodecContext *s, const enum AVPixelFormat *pix_fmts)
+{
+    InputStream *ist = s->opaque;
+    const enum AVPixelFormat *p;
+    int ret;
+
+    for (p = pix_fmts; *p != -1; p++) {
+        const AVPixFmtDescriptor *desc = av_pix_fmt_desc_get(*p);
+        const HWAccel *hwaccel;
+
+        if (!(desc->flags & AV_PIX_FMT_FLAG_HWACCEL))
+            break;
+
+        hwaccel = get_hwaccel(*p);
+        if (!hwaccel ||
+            (ist->active_hwaccel_id && ist->active_hwaccel_id != hwaccel->id) ||
+            (ist->hwaccel_id != HWACCEL_AUTO && ist->hwaccel_id != hwaccel->id))
+            continue;
+
+        ret = hwaccel->init(s);
+        if (ret < 0) {
+            if (ist->hwaccel_id == hwaccel->id) {
+                av_log(NULL, AV_LOG_FATAL,
+                       "%s hwaccel requested for input stream #%d:%d, "
+                       "but cannot be initialized.\n", hwaccel->name,
+                       ist->file_index, ist->st->index);
+                exit_program(1);
+            }
+            continue;
+        }
+        ist->active_hwaccel_id = hwaccel->id;
+        ist->hwaccel_pix_fmt   = *p;
+        break;
+    }
+
+    return *p;
+}
+
+static int get_buffer(AVCodecContext *s, AVFrame *frame, int flags)
+{
+    InputStream *ist = s->opaque;
+
+    if (ist->hwaccel_get_buffer && frame->format == ist->hwaccel_pix_fmt)
+        return ist->hwaccel_get_buffer(s, frame, flags);
+
+    return avcodec_default_get_buffer2(s, frame, flags);
+}
+
 static int init_input_stream(int ist_index, char *error, int error_len)
 {
     int ret;
@@ -1976,21 +2134,23 @@ static int init_input_stream(int ist_index, char *error, int error_len)
             return AVERROR(EINVAL);
         }
 
+        ist->st->codec->opaque      = ist;
+        ist->st->codec->get_format  = get_format;
+        ist->st->codec->get_buffer2 = get_buffer;
+        ist->st->codec->thread_safe_callbacks = 1;
+
         av_opt_set_int(ist->st->codec, "refcounted_frames", 1, 0);
 
         if (!av_dict_get(ist->opts, "threads", NULL, 0))
             av_dict_set(&ist->opts, "threads", "auto", 0);
         if ((ret = avcodec_open2(ist->st->codec, codec, &ist->opts)) < 0) {
-            char errbuf[128];
             if (ret == AVERROR_EXPERIMENTAL)
                 abort_codec_experimental(codec, 0);
 
-            av_strerror(ret, errbuf, sizeof(errbuf));
-
             snprintf(error, error_len,
                      "Error while opening decoder for input stream "
                      "#%d:%d : %s",
-                     ist->file_index, ist->st->index, errbuf);
+                     ist->file_index, ist->st->index, av_err2str(ret));
             return ret;
         }
         assert_avoptions(ist->opts);
@@ -1998,7 +2158,6 @@ static int init_input_stream(int ist_index, char *error, int error_len)
 
     ist->next_pts = AV_NOPTS_VALUE;
     ist->next_dts = AV_NOPTS_VALUE;
-    ist->is_start = 1;
 
     return 0;
 }
@@ -2258,6 +2417,8 @@ static int transcode_init(void)
             av_reduce(&codec->time_base.num, &codec->time_base.den,
                         codec->time_base.num, codec->time_base.den, INT_MAX);
 
+            ost->parser = av_parser_init(codec->codec_id);
+
             switch (codec->codec_type) {
             case AVMEDIA_TYPE_AUDIO:
                 if (audio_volume != 256) {
@@ -2270,6 +2431,7 @@ static int transcode_init(void)
                 codec->frame_size         = icodec->frame_size;
                 codec->audio_service_type = icodec->audio_service_type;
                 codec->block_align        = icodec->block_align;
+                codec->delay              = icodec->delay;
                 if((codec->block_align == 1 || codec->block_align == 1152 || codec->block_align == 576) && codec->codec_id == AV_CODEC_ID_MP3)
                     codec->block_align= 0;
                 if(codec->codec_id == AV_CODEC_ID_AC3)
@@ -2336,12 +2498,25 @@ static int transcode_init(void)
                 if (ist && !ost->frame_rate.num)
                     ost->frame_rate = ist->framerate;
                 if (ist && !ost->frame_rate.num)
-                    ost->frame_rate = ist->st->r_frame_rate.num ? ist->st->r_frame_rate : (AVRational){25, 1};
+                    ost->frame_rate = ist->st->r_frame_rate;
+                if (ist && !ost->frame_rate.num) {
+                    ost->frame_rate = (AVRational){25, 1};
+                    av_log(NULL, AV_LOG_WARNING,
+                           "No information "
+                           "about the input framerate is available. Falling "
+                           "back to a default value of 25fps for output stream #%d:%d. Use the -r option "
+                           "if you want a different framerate.\n",
+                           ost->file_index, ost->index);
+                }
 //                    ost->frame_rate = ist->st->avg_frame_rate.num ? ist->st->avg_frame_rate : (AVRational){25, 1};
                 if (ost->enc && ost->enc->supported_framerates && !ost->force_fps) {
                     int idx = av_find_nearest_q_idx(ost->frame_rate, ost->enc->supported_framerates);
                     ost->frame_rate = ost->enc->supported_framerates[idx];
                 }
+                if (codec->codec_id == AV_CODEC_ID_MPEG4) {
+                    av_reduce(&ost->frame_rate.num, &ost->frame_rate.den,
+                              ost->frame_rate.num, ost->frame_rate.den, 65535);
+                }
             }
 
             switch (codec->codec_type) {
@@ -2357,7 +2532,7 @@ static int transcode_init(void)
                 if (ost->filter && !(codec->time_base.num && codec->time_base.den))
                     codec->time_base = ost->filter->filter->inputs[0]->time_base;
                 if (   av_q2d(codec->time_base) < 0.001 && video_sync_method != VSYNC_PASSTHROUGH
-                   && (video_sync_method == VSYNC_CFR || (video_sync_method == VSYNC_AUTO && !(oc->oformat->flags & AVFMT_VARIABLE_FPS)))){
+                   && (video_sync_method == VSYNC_CFR || video_sync_method == VSYNC_VSCFR || (video_sync_method == VSYNC_AUTO && !(oc->oformat->flags & AVFMT_VARIABLE_FPS)))){
                     av_log(oc, AV_LOG_WARNING, "Frame rate very high for a muxer not efficiently supporting it.\n"
                                                "Please consider specifying a lower framerate, a different muxer or -vsync 2\n");
                 }
@@ -2447,7 +2622,7 @@ static int transcode_init(void)
                         codec->stats_in = logbuffer;
                     }
                     if (codec->flags & CODEC_FLAG_PASS1) {
-                        f = fopen(logfilename, "wb");
+                        f = av_fopen_utf8(logfilename, "wb");
                         if (!f) {
                             av_log(NULL, AV_LOG_FATAL, "Cannot write log file '%s' for pass-1 encoding: %s\n",
                                 logfilename, strerror(errno));
@@ -2497,9 +2672,6 @@ static int transcode_init(void)
                 av_log(NULL, AV_LOG_WARNING, "The bitrate parameter is set too low."
                                              " It takes bits/s as argument, not kbits/s\n");
             extra_size += ost->st->codec->extradata_size;
-
-            if (ost->st->codec->me_threshold)
-                input_streams[ost->source_index]->st->codec->debug |= FF_DEBUG_MV;
         } else {
             av_opt_set_dict(ost->st->codec, &ost->opts);
         }
@@ -2536,12 +2708,10 @@ static int transcode_init(void)
         oc = output_files[i]->ctx;
         oc->interrupt_callback = int_cb;
         if ((ret = avformat_write_header(oc, &output_files[i]->opts)) < 0) {
-            char errbuf[128];
-            av_strerror(ret, errbuf, sizeof(errbuf));
             snprintf(error, sizeof(error),
                      "Could not write header for output file #%d "
                      "(incorrect codec parameters ?): %s",
-                     i, errbuf);
+                     i, av_err2str(ret));
             ret = AVERROR(EINVAL);
             goto dump_format;
         }
@@ -2795,11 +2965,15 @@ static void *input_thread(void *arg)
 
         av_dup_packet(&pkt);
         av_fifo_generic_write(f->fifo, &pkt, sizeof(pkt), NULL);
+        pthread_cond_signal(&f->fifo_cond);
 
         pthread_mutex_unlock(&f->fifo_lock);
     }
 
+    pthread_mutex_lock(&f->fifo_lock);
     f->finished = 1;
+    pthread_cond_signal(&f->fifo_cond);
+    pthread_mutex_unlock(&f->fifo_lock);
     return NULL;
 }
 
@@ -2851,6 +3025,10 @@ static int init_input_threads(void)
         if (!(f->fifo = av_fifo_alloc(8*sizeof(AVPacket))))
             return AVERROR(ENOMEM);
 
+        if (f->ctx->pb ? !f->ctx->pb->seekable :
+            strcmp(f->ctx->iformat->name, "lavfi"))
+            f->non_blocking = 1;
+
         pthread_mutex_init(&f->fifo_lock, NULL);
         pthread_cond_init (&f->fifo_cond, NULL);
 
@@ -2866,14 +3044,22 @@ static int get_input_packet_mt(InputFile *f, AVPacket *pkt)
 
     pthread_mutex_lock(&f->fifo_lock);
 
+    while (1) {
     if (av_fifo_size(f->fifo)) {
         av_fifo_generic_read(f->fifo, pkt, sizeof(*pkt), NULL);
         pthread_cond_signal(&f->fifo_cond);
+        break;
     } else {
-        if (f->finished)
+        if (f->finished) {
             ret = AVERROR_EOF;
-        else
+            break;
+        }
+        if (f->non_blocking) {
             ret = AVERROR(EAGAIN);
+            break;
+        }
+        pthread_cond_wait(&f->fifo_cond, &f->fifo_lock);
+    }
     }
 
     pthread_mutex_unlock(&f->fifo_lock);
@@ -3064,22 +3250,21 @@ static int process_input(int file_index)
         int64_t pkt_dts = av_rescale_q(pkt.dts, ist->st->time_base, AV_TIME_BASE_Q);
         int64_t delta   = pkt_dts - ist->next_dts;
         if (is->iformat->flags & AVFMT_TS_DISCONT) {
-        if(delta < -1LL*dts_delta_threshold*AV_TIME_BASE ||
-            (delta > 1LL*dts_delta_threshold*AV_TIME_BASE &&
-                ist->st->codec->codec_type != AVMEDIA_TYPE_SUBTITLE) ||
-            pkt_dts + AV_TIME_BASE/10 < ist->pts){
-            ifile->ts_offset -= delta;
-            av_log(NULL, AV_LOG_DEBUG,
-                   "timestamp discontinuity %"PRId64", new offset= %"PRId64"\n",
-                   delta, ifile->ts_offset);
-            pkt.dts -= av_rescale_q(delta, AV_TIME_BASE_Q, ist->st->time_base);
-            if (pkt.pts != AV_NOPTS_VALUE)
-                pkt.pts -= av_rescale_q(delta, AV_TIME_BASE_Q, ist->st->time_base);
-        }
+            if (delta < -1LL*dts_delta_threshold*AV_TIME_BASE ||
+                (delta > 1LL*dts_delta_threshold*AV_TIME_BASE &&
+                 ist->st->codec->codec_type != AVMEDIA_TYPE_SUBTITLE) ||
+                pkt_dts + AV_TIME_BASE/10 < FFMAX(ist->pts, ist->dts)) {
+                ifile->ts_offset -= delta;
+                av_log(NULL, AV_LOG_DEBUG,
+                       "timestamp discontinuity %"PRId64", new offset= %"PRId64"\n",
+                       delta, ifile->ts_offset);
+                pkt.dts -= av_rescale_q(delta, AV_TIME_BASE_Q, ist->st->time_base);
+                if (pkt.pts != AV_NOPTS_VALUE)
+                    pkt.pts -= av_rescale_q(delta, AV_TIME_BASE_Q, ist->st->time_base);
+            }
         } else {
             if ( delta < -1LL*dts_error_threshold*AV_TIME_BASE ||
-                (delta > 1LL*dts_error_threshold*AV_TIME_BASE && ist->st->codec->codec_type != AVMEDIA_TYPE_SUBTITLE)
-               ) {
+                (delta > 1LL*dts_error_threshold*AV_TIME_BASE && ist->st->codec->codec_type != AVMEDIA_TYPE_SUBTITLE)) {
                 av_log(NULL, AV_LOG_WARNING, "DTS %"PRId64", next:%"PRId64" st:%d invalid dropping\n", pkt.dts, ist->next_dts, pkt.stream_index);
                 pkt.dts = AV_NOPTS_VALUE;
             }
@@ -3087,8 +3272,7 @@ static int process_input(int file_index)
                 int64_t pkt_pts = av_rescale_q(pkt.pts, ist->st->time_base, AV_TIME_BASE_Q);
                 delta   = pkt_pts - ist->next_dts;
                 if ( delta < -1LL*dts_error_threshold*AV_TIME_BASE ||
-                    (delta > 1LL*dts_error_threshold*AV_TIME_BASE && ist->st->codec->codec_type != AVMEDIA_TYPE_SUBTITLE)
-                   ) {
+                    (delta > 1LL*dts_error_threshold*AV_TIME_BASE && ist->st->codec->codec_type != AVMEDIA_TYPE_SUBTITLE)) {
                     av_log(NULL, AV_LOG_WARNING, "PTS %"PRId64", next:%"PRId64" invalid dropping st:%d\n", pkt.pts, ist->next_dts, pkt.stream_index);
                     pkt.pts = AV_NOPTS_VALUE;
                 }
@@ -3112,10 +3296,8 @@ static int process_input(int file_index)
 
     ret = output_packet(ist, &pkt);
     if (ret < 0) {
-        char buf[128];
-        av_strerror(ret, buf, sizeof(buf));
         av_log(NULL, AV_LOG_ERROR, "Error while decoding stream #%d:%d: %s\n",
-                ist->file_index, ist->st->index, buf);
+               ist->file_index, ist->st->index, av_err2str(ret));
         if (exit_on_error)
             exit_program(1);
     }
@@ -3308,6 +3490,8 @@ static int transcode(void)
         ist = input_streams[i];
         if (ist->decoding_needed) {
             avcodec_close(ist->st->codec);
+            if (ist->hwaccel_uninit)
+                ist->hwaccel_uninit(ist->st->codec);
         }
     }
 
@@ -3447,6 +3631,6 @@ int main(int argc, char **argv)
     if ((decode_error_stat[0] + decode_error_stat[1]) * max_error_rate < decode_error_stat[1])
         exit_program(69);
 
-    exit_program(received_nb_signals ? 255 : 0);
-    return 0;
+    exit_program(received_nb_signals ? 255 : main_return_code);
+    return main_return_code;
 }

ffmpeg.h

@@ -51,10 +51,24 @@
 #define VSYNC_PASSTHROUGH 0
 #define VSYNC_CFR         1
 #define VSYNC_VFR         2
+#define VSYNC_VSCFR       0xfe
 #define VSYNC_DROP        0xff
 
 #define MAX_STREAMS 1024    /* arbitrary sanity check value */
 
+enum HWAccelID {
+    HWACCEL_NONE = 0,
+    HWACCEL_AUTO,
+    HWACCEL_VDPAU,
+};
+
+typedef struct HWAccel {
+    const char *name;
+    int (*init)(AVCodecContext *s);
+    enum HWAccelID id;
+    enum AVPixelFormat pix_fmt;
+} HWAccel;
+
 /* select an input stream for an output stream */
 typedef struct StreamMap {
     int disabled;           /* 1 is this mapping is disabled by a negative map */
@@ -99,6 +113,10 @@ typedef struct OptionsContext {
     int        nb_ts_scale;
     SpecifierOpt *dump_attachment;
     int        nb_dump_attachment;
+    SpecifierOpt *hwaccels;
+    int        nb_hwaccels;
+    SpecifierOpt *hwaccel_devices;
+    int        nb_hwaccel_devices;
 
     /* output options */
     StreamMap *stream_maps;
@@ -153,6 +171,8 @@ typedef struct OptionsContext {
     int        nb_intra_matrices;
     SpecifierOpt *inter_matrices;
     int        nb_inter_matrices;
+    SpecifierOpt *chroma_intra_matrices;
+    int        nb_chroma_intra_matrices;
     SpecifierOpt *top_field_first;
     int        nb_top_field_first;
     SpecifierOpt *metadata_map;
@@ -235,7 +255,6 @@ typedef struct InputStream {
     int64_t filter_in_rescale_delta_last;
 
     double ts_scale;
-    int is_start;            /* is 1 at the start and after a discontinuity */
     int saw_first_ts;
     int showed_multi_packet_warning;
     AVDictionary *opts;
@@ -274,6 +293,19 @@ typedef struct InputStream {
     int        nb_filters;
 
     int reinit_filters;
+
+    /* hwaccel options */
+    enum HWAccelID hwaccel_id;
+    char  *hwaccel_device;
+
+    /* hwaccel context */
+    enum HWAccelID active_hwaccel_id;
+    void  *hwaccel_ctx;
+    void (*hwaccel_uninit)(AVCodecContext *s);
+    int  (*hwaccel_get_buffer)(AVCodecContext *s, AVFrame *frame, int flags);
+    int  (*hwaccel_retrieve_data)(AVCodecContext *s, AVFrame *frame);
+    enum AVPixelFormat hwaccel_pix_fmt;
+    enum AVPixelFormat hwaccel_retrieved_pix_fmt;
 } InputStream;
 
 typedef struct InputFile {
@@ -281,6 +313,7 @@ typedef struct InputFile {
     int eof_reached;      /* true if eof reached */
     int eagain;           /* true if last read attempt returned EAGAIN */
     int ist_index;        /* index of first stream in input_streams */
+    int64_t input_ts_offset;
     int64_t ts_offset;
     int64_t last_ts;
     int64_t start_time;   /* user-specified start time in AV_TIME_BASE or AV_NOPTS_VALUE */
@@ -293,6 +326,7 @@ typedef struct InputFile {
 
 #if HAVE_PTHREADS
     pthread_t thread;           /* thread reading from this file */
+    int non_blocking;           /* reading packets from the thread should not block */
     int finished;               /* the thread has exited */
     int joined;                 /* the thread has been joined */
     pthread_mutex_t fifo_lock;  /* lock for access to fifo */
@@ -312,6 +346,11 @@ enum forced_keyframes_const {
 
 extern const char *const forced_keyframes_const_names[];
 
+typedef enum {
+    ENCODER_FINISHED = 1,
+    MUXER_FINISHED = 2,
+} OSTFinished ;
+
 typedef struct OutputStream {
     int file_index;          /* file index */
     int index;               /* stream index in the output file */
@@ -357,13 +396,15 @@ typedef struct OutputStream {
 
     OutputFilter *filter;
     char *avfilter;
+    char *filters;         ///< filtergraph associated to the -filter option
+    char *filters_script;  ///< filtergraph script associated to the -filter_script option
 
     int64_t sws_flags;
     AVDictionary *opts;
     AVDictionary *swr_opts;
     AVDictionary *resample_opts;
     char *apad;
-    int finished;        /* no more packets should be written for this stream */
+    OSTFinished finished;        /* no more packets should be written for this stream */
     int unavailable;                     /* true if the steram is unavailable (possibly temporarily) */
     int stream_copy;
     const char *attachment_filename;
@@ -371,6 +412,8 @@ typedef struct OutputStream {
     int copy_prior_start;
 
     int keep_pix_fmt;
+
+    AVCodecParserContext *parser;
 } OutputStream;
 
 typedef struct OutputFile {
@@ -425,6 +468,8 @@ extern float max_error_rate;
 extern const AVIOInterruptCB int_cb;
 
 extern const OptionDef options[];
+extern const HWAccel hwaccels[];
+
 
 void term_init(void);
 void term_exit(void);
@@ -448,4 +493,6 @@ FilterGraph *init_simple_filtergraph(InputStream *ist, OutputStream *ost);
 
 int ffmpeg_parse_options(int argc, char **argv);
 
+int vdpau_init(AVCodecContext *s);
+
 #endif /* FFMPEG_H */

ffmpeg_filter.c

@@ -18,6 +18,8 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
+#include <stdint.h>
+
 #include "ffmpeg.h"
 
 #include "libavfilter/avfilter.h"
@@ -42,12 +44,15 @@ enum AVPixelFormat choose_pixel_fmt(AVStream *st, AVCodec *codec, enum AVPixelFo
         const AVPixFmtDescriptor *desc = av_pix_fmt_desc_get(target);
         int has_alpha = desc ? desc->nb_components % 2 == 0 : 0;
         enum AVPixelFormat best= AV_PIX_FMT_NONE;
+        const enum AVPixelFormat mjpeg_formats[] = { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUV420P, AV_PIX_FMT_YUV422P, AV_PIX_FMT_NONE };
+        const enum AVPixelFormat ljpeg_formats[] = { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUVJ444P, AV_PIX_FMT_YUV420P,
+                                                     AV_PIX_FMT_YUV422P, AV_PIX_FMT_YUV444P, AV_PIX_FMT_BGRA, AV_PIX_FMT_NONE };
+
         if (st->codec->strict_std_compliance <= FF_COMPLIANCE_UNOFFICIAL) {
             if (st->codec->codec_id == AV_CODEC_ID_MJPEG) {
-                p = (const enum AVPixelFormat[]) { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUV420P, AV_PIX_FMT_YUV422P, AV_PIX_FMT_NONE };
+                p = mjpeg_formats;
             } else if (st->codec->codec_id == AV_CODEC_ID_LJPEG) {
-                p = (const enum AVPixelFormat[]) { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUVJ444P, AV_PIX_FMT_YUV420P,
-                                                 AV_PIX_FMT_YUV422P, AV_PIX_FMT_YUV444P, AV_PIX_FMT_BGRA, AV_PIX_FMT_NONE };
+                p =ljpeg_formats;
             }
         }
         for (; *p != AV_PIX_FMT_NONE; p++) {
@@ -654,7 +659,8 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
     av_bprintf(&args,
              "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:"
              "pixel_aspect=%d/%d:sws_param=flags=%d", ist->resample_width,
-             ist->resample_height, ist->resample_pix_fmt,
+             ist->resample_height,
+             ist->hwaccel_retrieve_data ? ist->hwaccel_retrieved_pix_fmt : ist->resample_pix_fmt,
              tb.num, tb.den, sar.num, sar.den,
              SWS_BILINEAR + ((ist->st->codec->flags&CODEC_FLAG_BITEXACT) ? SWS_BITEXACT:0));
     if (fr.num && fr.den)

ffmpeg_opt.c

@@ -62,6 +62,14 @@
             outvar = o->name[i].u.type;\
     }\
 }
+
+const HWAccel hwaccels[] = {
+#if HAVE_VDPAU_X11
+    { "vdpau", vdpau_init, HWACCEL_VDPAU, AV_PIX_FMT_VDPAU },
+#endif
+    { 0 },
+};
+
 char *vstats_filename;
 
 float audio_drift_threshold = 0.1;
@@ -551,13 +559,14 @@ static AVCodec *choose_decoder(OptionsContext *o, AVFormatContext *s, AVStream *
 static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
 {
     int i;
-    char *next, *codec_tag = NULL;
 
     for (i = 0; i < ic->nb_streams; i++) {
         AVStream *st = ic->streams[i];
         AVCodecContext *dec = st->codec;
         InputStream *ist = av_mallocz(sizeof(*ist));
-        char *framerate = NULL;
+        char *framerate = NULL, *hwaccel = NULL, *hwaccel_device = NULL;
+        char *codec_tag = NULL;
+        char *next;
 
         if (!ist)
             exit_program(1);
@@ -612,6 +621,41 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
             ist->top_field_first = -1;
             MATCH_PER_STREAM_OPT(top_field_first, i, ist->top_field_first, ic, st);
 
+            MATCH_PER_STREAM_OPT(hwaccels, str, hwaccel, ic, st);
+            if (hwaccel) {
+                if (!strcmp(hwaccel, "none"))
+                    ist->hwaccel_id = HWACCEL_NONE;
+                else if (!strcmp(hwaccel, "auto"))
+                    ist->hwaccel_id = HWACCEL_AUTO;
+                else {
+                    int i;
+                    for (i = 0; hwaccels[i].name; i++) {
+                        if (!strcmp(hwaccels[i].name, hwaccel)) {
+                            ist->hwaccel_id = hwaccels[i].id;
+                            break;
+                        }
+                    }
+
+                    if (!ist->hwaccel_id) {
+                        av_log(NULL, AV_LOG_FATAL, "Unrecognized hwaccel: %s.\n",
+                               hwaccel);
+                        av_log(NULL, AV_LOG_FATAL, "Supported hwaccels: ");
+                        for (i = 0; hwaccels[i].name; i++)
+                            av_log(NULL, AV_LOG_FATAL, "%s ", hwaccels[i].name);
+                        av_log(NULL, AV_LOG_FATAL, "\n");
+                        exit_program(1);
+                    }
+                }
+            }
+
+            MATCH_PER_STREAM_OPT(hwaccel_devices, str, hwaccel_device, ic, st);
+            if (hwaccel_device) {
+                ist->hwaccel_device = av_strdup(hwaccel_device);
+                if (!ist->hwaccel_device)
+                    exit_program(1);
+            }
+            ist->hwaccel_pix_fmt = AV_PIX_FMT_NONE;
+
             break;
         case AVMEDIA_TYPE_AUDIO:
             ist->guess_layout_max = INT_MAX;
@@ -654,10 +698,9 @@ static void assert_file_overwrite(const char *filename)
         exit_program(1);
     }
 
-    if (!file_overwrite &&
-        (strchr(filename, ':') == NULL || filename[1] == ':' ||
-         av_strstart(filename, "file:", NULL))) {
-        if (avio_check(filename, 0) == 0) {
+    if (!file_overwrite) {
+        const char *proto_name = avio_find_protocol_name(filename);
+        if (proto_name && !strcmp(proto_name, "file") && avio_check(filename, 0) == 0) {
             if (stdin_interaction && !no_file_overwrite) {
                 fprintf(stderr,"File '%s' already exists. Overwrite ? [y/N] ", filename);
                 fflush(stderr);
@@ -852,6 +895,7 @@ static int open_input_file(OptionsContext *o, const char *filename)
     f->ist_index  = nb_input_streams - ic->nb_streams;
     f->start_time = o->start_time;
     f->recording_time = o->recording_time;
+    f->input_ts_offset = o->input_ts_offset;
     f->ts_offset  = o->input_ts_offset - (copy_ts ? 0 : timestamp);
     f->nb_streams = ic->nb_streams;
     f->rate_emu   = o->rate_emu;
@@ -871,8 +915,13 @@ static int open_input_file(OptionsContext *o, const char *filename)
         const AVClass *class = avcodec_get_class();
         const AVOption *option = av_opt_find(&class, e->key, NULL, 0,
                                              AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ);
-        if (!option)
+        const AVClass *fclass = avformat_get_class();
+        const AVOption *foption = av_opt_find(&fclass, e->key, NULL, 0,
+                                             AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ);
+        if (!option || foption)
             continue;
+
+
         if (!(option->flags & AV_OPT_FLAG_DECODING_PARAM)) {
             av_log(NULL, AV_LOG_ERROR, "Codec AVOption %s (%s) specified for "
                    "input file #%d (%s) is not a decoding option.\n", e->key,
@@ -929,14 +978,14 @@ static uint8_t *get_line(AVIOContext *s)
 
 static int get_preset_file_2(const char *preset_name, const char *codec_name, AVIOContext **s)
 {
-    int i, ret = 1;
+    int i, ret = -1;
     char filename[1000];
     const char *base[3] = { getenv("AVCONV_DATADIR"),
                             getenv("HOME"),
                             AVCONV_DATADIR,
                             };
 
-    for (i = 0; i < FF_ARRAY_ELEMS(base) && ret; i++) {
+    for (i = 0; i < FF_ARRAY_ELEMS(base) && ret < 0; i++) {
         if (!base[i])
             continue;
         if (codec_name) {
@@ -944,7 +993,7 @@ static int get_preset_file_2(const char *preset_name, const char *codec_name, AV
                      i != 1 ? "" : "/.avconv", codec_name, preset_name);
             ret = avio_open2(s, filename, AVIO_FLAG_READ, &int_cb, NULL);
         }
-        if (ret) {
+        if (ret < 0) {
             snprintf(filename, sizeof(filename), "%s%s/%s.avpreset", base[i],
                      i != 1 ? "" : "/.avconv", preset_name);
             ret = avio_open2(s, filename, AVIO_FLAG_READ, &int_cb, NULL);
@@ -1151,26 +1200,36 @@ static char *get_ost_filters(OptionsContext *o, AVFormatContext *oc,
                              OutputStream *ost)
 {
     AVStream *st = ost->st;
-    char *filter = NULL, *filter_script = NULL;
 
-    MATCH_PER_STREAM_OPT(filter_scripts, str, filter_script, oc, st);
-    MATCH_PER_STREAM_OPT(filters, str, filter, oc, st);
-
-    if (filter_script && filter) {
+    if (ost->filters_script && ost->filters) {
         av_log(NULL, AV_LOG_ERROR, "Both -filter and -filter_script set for "
                "output stream #%d:%d.\n", nb_output_files, st->index);
         exit_program(1);
     }
 
-    if (filter_script)
-        return read_file(filter_script);
-    else if (filter)
-        return av_strdup(filter);
+    if (ost->filters_script)
+        return read_file(ost->filters_script);
+    else if (ost->filters)
+        return av_strdup(ost->filters);
 
     return av_strdup(st->codec->codec_type == AVMEDIA_TYPE_VIDEO ?
                      "null" : "anull");
 }
 
+static void check_streamcopy_filters(OptionsContext *o, AVFormatContext *oc,
+                                     const OutputStream *ost, enum AVMediaType type)
+{
+    if (ost->filters_script || ost->filters) {
+        av_log(NULL, AV_LOG_ERROR,
+               "%s '%s' was defined for %s output stream %d:%d but codec copy was selected.\n"
+               "Filtering and streamcopy cannot be used together.\n",
+               ost->filters ? "Filtergraph" : "Filtergraph script",
+               ost->filters ? ost->filters : ost->filters_script,
+               av_get_media_type_string(type), ost->file_index, ost->index);
+        exit_program(1);
+    }
+}
+
 static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, int source_index)
 {
     AVStream *st;
@@ -1199,11 +1258,15 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
         ost->frame_aspect_ratio = q;
     }
 
+    MATCH_PER_STREAM_OPT(filter_scripts, str, ost->filters_script, oc, st);
+    MATCH_PER_STREAM_OPT(filters,        str, ost->filters,        oc, st);
+
     if (!ost->stream_copy) {
         const char *p = NULL;
         char *frame_size = NULL;
         char *frame_pix_fmt = NULL;
         char *intra_matrix = NULL, *inter_matrix = NULL;
+        char *chroma_intra_matrix = NULL;
         int do_pass = 0;
         int i;
 
@@ -1236,6 +1299,16 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
             }
             parse_matrix_coeffs(video_enc->intra_matrix, intra_matrix);
         }
+        MATCH_PER_STREAM_OPT(chroma_intra_matrices, str, chroma_intra_matrix, oc, st);
+        if (chroma_intra_matrix) {
+            uint16_t *p = av_mallocz(sizeof(*video_enc->chroma_intra_matrix) * 64);
+            if (!p) {
+                av_log(NULL, AV_LOG_FATAL, "Could not allocate memory for intra matrix.\n");
+                exit_program(1);
+            }
+            av_codec_set_chroma_intra_matrix(video_enc, p);
+            parse_matrix_coeffs(p, chroma_intra_matrix);
+        }
         MATCH_PER_STREAM_OPT(inter_matrices, str, inter_matrix, oc, st);
         if (inter_matrix) {
             if (!(video_enc->inter_matrix = av_mallocz(sizeof(*video_enc->inter_matrix) * 64))) {
@@ -1311,6 +1384,9 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
         MATCH_PER_STREAM_OPT(copy_initial_nonkeyframes, i, ost->copy_initial_nonkeyframes, oc ,st);
     }
 
+    if (ost->stream_copy)
+        check_streamcopy_filters(o, oc, ost, AVMEDIA_TYPE_VIDEO);
+
     return ost;
 }
 
@@ -1327,6 +1403,9 @@ static OutputStream *new_audio_stream(OptionsContext *o, AVFormatContext *oc, in
     audio_enc = st->codec;
     audio_enc->codec_type = AVMEDIA_TYPE_AUDIO;
 
+    MATCH_PER_STREAM_OPT(filter_scripts, str, ost->filters_script, oc, st);
+    MATCH_PER_STREAM_OPT(filters,        str, ost->filters,        oc, st);
+
     if (!ost->stream_copy) {
         char *sample_fmt = NULL;
 
@@ -1364,6 +1443,9 @@ static OutputStream *new_audio_stream(OptionsContext *o, AVFormatContext *oc, in
         }
     }
 
+    if (ost->stream_copy)
+        check_streamcopy_filters(o, oc, ost, AVMEDIA_TYPE_AUDIO);
+
     return ost;
 }
 
@@ -1548,6 +1630,18 @@ static void init_output_filter(OutputFilter *ofilter, OptionsContext *o,
         exit_program(1);
     }
 
+    if (ost->avfilter && (ost->filters || ost->filters_script)) {
+        const char *opt = ost->filters ? "-vf/-af/-filter" : "-filter_script";
+        av_log(NULL, AV_LOG_ERROR,
+               "%s '%s' was specified through the %s option "
+               "for output stream %d:%d, which is fed from a complex filtergraph.\n"
+               "%s and -filter_complex cannot be used together for the same stream.\n",
+               ost->filters ? "Filtergraph" : "Filtergraph script",
+               ost->filters ? ost->filters : ost->filters_script,
+               opt, ost->file_index, ost->index, opt);
+        exit_program(1);
+    }
+
     if (configure_output_filter(ofilter->graph, ofilter, ofilter->out_tmp) < 0) {
         av_log(NULL, AV_LOG_FATAL, "Error configuring filter.\n");
         exit_program(1);
@@ -1727,7 +1821,7 @@ static int open_output_file(OptionsContext *o, const char *filename)
 
         /* subtitles: pick first */
         MATCH_PER_TYPE_OPT(codec_names, str, subtitle_codec_name, oc, "s");
-        if (!o->subtitle_disable && (oc->oformat->subtitle_codec != AV_CODEC_ID_NONE || subtitle_codec_name)) {
+        if (!o->subtitle_disable && (avcodec_find_encoder(oc->oformat->subtitle_codec) || subtitle_codec_name)) {
             for (i = 0; i < nb_input_streams; i++)
                 if (input_streams[i]->st->codec->codec_type == AVMEDIA_TYPE_SUBTITLE) {
                     new_subtitle_stream(o, oc, i);
@@ -2055,23 +2149,22 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
         opt_video_codec(o, "c:v", "mpeg1video");
         opt_audio_codec(o, "c:a", "mp2");
         parse_option(o, "f", "vcd", options);
-        av_dict_set(&o->g->codec_opts, "b:v", arg, 0);
 
         parse_option(o, "s", norm == PAL ? "352x288" : "352x240", options);
         parse_option(o, "r", frame_rates[norm], options);
-        av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", 0);
+        av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", AV_DICT_DONT_OVERWRITE);
 
-        av_dict_set(&o->g->codec_opts, "b:v", "1150000", 0);
-        av_dict_set(&o->g->codec_opts, "maxrate", "1150000", 0);
-        av_dict_set(&o->g->codec_opts, "minrate", "1150000", 0);
-        av_dict_set(&o->g->codec_opts, "bufsize", "327680", 0); // 40*1024*8;
+        av_dict_set(&o->g->codec_opts, "b:v", "1150000", AV_DICT_DONT_OVERWRITE);
+        av_dict_set(&o->g->codec_opts, "maxrate", "1150000", AV_DICT_DONT_OVERWRITE);
+        av_dict_set(&o->g->codec_opts, "minrate", "1150000", AV_DICT_DONT_OVERWRITE);
+        av_dict_set(&o->g->codec_opts, "bufsize", "327680", AV_DICT_DONT_OVERWRITE); // 40*1024*8;
 
-        av_dict_set(&o->g->codec_opts, "b:a", "224000", 0);
+        av_dict_set(&o->g->codec_opts, "b:a", "224000", AV_DICT_DONT_OVERWRITE);
         parse_option(o, "ar", "44100", options);
         parse_option(o, "ac", "2", options);
 
-        av_dict_set(&o->g->format_opts, "packetsize", "2324", 0);
-        av_dict_set(&o->g->format_opts, "muxrate", "1411200", 0); // 2352 * 75 * 8;
+        av_dict_set(&o->g->format_opts, "packetsize", "2324", AV_DICT_DONT_OVERWRITE);
+        av_dict_set(&o->g->format_opts, "muxrate", "1411200", AV_DICT_DONT_OVERWRITE); // 2352 * 75 * 8;
 
         /* We have to offset the PTS, so that it is consistent with the SCR.
            SCR starts at 36000, but the first two packs contain only padding
@@ -2088,18 +2181,18 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
         parse_option(o, "s", norm == PAL ? "480x576" : "480x480", options);
         parse_option(o, "r", frame_rates[norm], options);
         parse_option(o, "pix_fmt", "yuv420p", options);
-        av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", 0);
+        av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", AV_DICT_DONT_OVERWRITE);
 
-        av_dict_set(&o->g->codec_opts, "b:v", "2040000", 0);
-        av_dict_set(&o->g->codec_opts, "maxrate", "2516000", 0);
-        av_dict_set(&o->g->codec_opts, "minrate", "0", 0); // 1145000;
-        av_dict_set(&o->g->codec_opts, "bufsize", "1835008", 0); // 224*1024*8;
-        av_dict_set(&o->g->codec_opts, "scan_offset", "1", 0);
+        av_dict_set(&o->g->codec_opts, "b:v", "2040000", AV_DICT_DONT_OVERWRITE);
+        av_dict_set(&o->g->codec_opts, "maxrate", "2516000", AV_DICT_DONT_OVERWRITE);
+        av_dict_set(&o->g->codec_opts, "minrate", "0", AV_DICT_DONT_OVERWRITE); // 1145000;
+        av_dict_set(&o->g->codec_opts, "bufsize", "1835008", AV_DICT_DONT_OVERWRITE); // 224*1024*8;
+        av_dict_set(&o->g->codec_opts, "scan_offset", "1", AV_DICT_DONT_OVERWRITE);
 
-        av_dict_set(&o->g->codec_opts, "b:a", "224000", 0);
+        av_dict_set(&o->g->codec_opts, "b:a", "224000", AV_DICT_DONT_OVERWRITE);
         parse_option(o, "ar", "44100", options);
 
-        av_dict_set(&o->g->format_opts, "packetsize", "2324", 0);
+        av_dict_set(&o->g->format_opts, "packetsize", "2324", AV_DICT_DONT_OVERWRITE);
 
     } else if (!strcmp(arg, "dvd")) {
 
@@ -2110,17 +2203,17 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
         parse_option(o, "s", norm == PAL ? "720x576" : "720x480", options);
         parse_option(o, "r", frame_rates[norm], options);
         parse_option(o, "pix_fmt", "yuv420p", options);
-        av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", 0);
+        av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", AV_DICT_DONT_OVERWRITE);
 
-        av_dict_set(&o->g->codec_opts, "b:v", "6000000", 0);
-        av_dict_set(&o->g->codec_opts, "maxrate", "9000000", 0);
-        av_dict_set(&o->g->codec_opts, "minrate", "0", 0); // 1500000;
-        av_dict_set(&o->g->codec_opts, "bufsize", "1835008", 0); // 224*1024*8;
+        av_dict_set(&o->g->codec_opts, "b:v", "6000000", AV_DICT_DONT_OVERWRITE);
+        av_dict_set(&o->g->codec_opts, "maxrate", "9000000", AV_DICT_DONT_OVERWRITE);
+        av_dict_set(&o->g->codec_opts, "minrate", "0", AV_DICT_DONT_OVERWRITE); // 1500000;
+        av_dict_set(&o->g->codec_opts, "bufsize", "1835008", AV_DICT_DONT_OVERWRITE); // 224*1024*8;
 
-        av_dict_set(&o->g->format_opts, "packetsize", "2048", 0);  // from www.mpucoder.com: DVD sectors contain 2048 bytes of data, this is also the size of one pack.
-        av_dict_set(&o->g->format_opts, "muxrate", "10080000", 0); // from mplex project: data_rate = 1260000. mux_rate = data_rate * 8
+        av_dict_set(&o->g->format_opts, "packetsize", "2048", AV_DICT_DONT_OVERWRITE);  // from www.mpucoder.com: DVD sectors contain 2048 bytes of data, this is also the size of one pack.
+        av_dict_set(&o->g->format_opts, "muxrate", "10080000", AV_DICT_DONT_OVERWRITE); // from mplex project: data_rate = 1260000. mux_rate = data_rate * 8
 
-        av_dict_set(&o->g->codec_opts, "b:a", "448000", 0);
+        av_dict_set(&o->g->codec_opts, "b:a", "448000", AV_DICT_DONT_OVERWRITE);
         parse_option(o, "ar", "48000", options);
 
     } else if (!strncmp(arg, "dv", 2)) {
@@ -2688,7 +2781,7 @@ const OptionDef options[] = {
     { "frames",         OPT_INT64 | HAS_ARG | OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(max_frames) },
         "set the number of frames to record", "number" },
     { "tag",            OPT_STRING | HAS_ARG | OPT_SPEC |
-                        OPT_EXPERT | OPT_OUTPUT,                     { .off = OFFSET(codec_tags) },
+                        OPT_EXPERT | OPT_OUTPUT | OPT_INPUT,         { .off = OFFSET(codec_tags) },
         "force codec tag/fourcc", "fourcc/tag" },
     { "q",              HAS_ARG | OPT_EXPERT | OPT_DOUBLE |
                         OPT_SPEC | OPT_OUTPUT,                       { .off = OFFSET(qscale) },
@@ -2779,6 +2872,9 @@ const OptionDef options[] = {
     { "inter_matrix", OPT_VIDEO | HAS_ARG | OPT_EXPERT  | OPT_STRING | OPT_SPEC |
                       OPT_OUTPUT,                                                { .off = OFFSET(inter_matrices) },
         "specify inter matrix coeffs", "matrix" },
+    { "chroma_intra_matrix", OPT_VIDEO | HAS_ARG | OPT_EXPERT  | OPT_STRING | OPT_SPEC |
+                      OPT_OUTPUT,                                                { .off = OFFSET(chroma_intra_matrices) },
+        "specify intra matrix coeffs", "matrix" },
     { "top",          OPT_VIDEO | HAS_ARG | OPT_EXPERT  | OPT_INT| OPT_SPEC |
                       OPT_INPUT | OPT_OUTPUT,                                    { .off = OFFSET(top_field_first) },
         "top=1/bottom=0/auto=-1 field first", "" },
@@ -2800,6 +2896,12 @@ const OptionDef options[] = {
         "force key frames at specified timestamps", "timestamps" },
     { "b",            OPT_VIDEO | HAS_ARG | OPT_PERFILE | OPT_OUTPUT,            { .func_arg = opt_bitrate },
         "video bitrate (please use -b:v)", "bitrate" },
+    { "hwaccel",          OPT_VIDEO | OPT_STRING | HAS_ARG | OPT_EXPERT |
+                          OPT_SPEC | OPT_INPUT,                                  { .off = OFFSET(hwaccels) },
+        "use HW accelerated decoding", "hwaccel name" },
+    { "hwaccel_device",   OPT_VIDEO | OPT_STRING | HAS_ARG | OPT_EXPERT |
+                          OPT_SPEC | OPT_INPUT,                                  { .off = OFFSET(hwaccel_devices) },
+        "select a device for HW acceleration" "devicename" },
 
     /* audio options */
     { "aframes",        OPT_AUDIO | HAS_ARG  | OPT_PERFILE | OPT_OUTPUT,           { .func_arg = opt_audio_frames },

ffmpeg_vdpau.c

@@ -0,0 +1,335 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <stdint.h>
+
+#include <vdpau/vdpau.h>
+#include <vdpau/vdpau_x11.h>
+
+#include <X11/Xlib.h>
+
+#include "ffmpeg.h"
+
+#include "libavcodec/vdpau.h"
+
+#include "libavutil/avassert.h"
+#include "libavutil/buffer.h"
+#include "libavutil/frame.h"
+#include "libavutil/pixfmt.h"
+
+typedef struct VDPAUContext {
+    Display *dpy;
+
+    VdpDevice  device;
+    VdpDecoder decoder;
+    VdpGetProcAddress *get_proc_address;
+
+    VdpGetErrorString                               *get_error_string;
+    VdpGetInformationString                         *get_information_string;
+    VdpDeviceDestroy                                *device_destroy;
+    VdpDecoderCreate                                *decoder_create;
+    VdpDecoderDestroy                               *decoder_destroy;
+    VdpDecoderRender                                *decoder_render;
+    VdpVideoSurfaceCreate                           *video_surface_create;
+    VdpVideoSurfaceDestroy                          *video_surface_destroy;
+    VdpVideoSurfaceGetBitsYCbCr                     *video_surface_get_bits;
+    VdpVideoSurfaceGetParameters                    *video_surface_get_parameters;
+    VdpVideoSurfaceQueryGetPutBitsYCbCrCapabilities *video_surface_query;
+
+    AVFrame *tmp_frame;
+
+    enum AVPixelFormat pix_fmt;
+    VdpYCbCrFormat vdpau_format;
+} VDPAUContext;
+
+static void vdpau_uninit(AVCodecContext *s)
+{
+    InputStream  *ist = s->opaque;
+    VDPAUContext *ctx = ist->hwaccel_ctx;
+
+    ist->hwaccel_uninit        = NULL;
+    ist->hwaccel_get_buffer    = NULL;
+    ist->hwaccel_retrieve_data = NULL;
+
+    if (ctx->decoder_destroy)
+        ctx->decoder_destroy(ctx->decoder);
+
+    if (ctx->device_destroy)
+        ctx->device_destroy(ctx->device);
+
+    if (ctx->dpy)
+        XCloseDisplay(ctx->dpy);
+
+    av_frame_free(&ctx->tmp_frame);
+
+    av_freep(&ist->hwaccel_ctx);
+    av_freep(&s->hwaccel_context);
+}
+
+static void vdpau_release_buffer(void *opaque, uint8_t *data)
+{
+    VdpVideoSurface surface = *(VdpVideoSurface*)data;
+    VDPAUContext *ctx = opaque;
+
+    ctx->video_surface_destroy(surface);
+    av_freep(&data);
+}
+
+static int vdpau_get_buffer(AVCodecContext *s, AVFrame *frame, int flags)
+{
+    InputStream         *ist = s->opaque;
+    VDPAUContext        *ctx = ist->hwaccel_ctx;
+    VdpVideoSurface *surface;
+    VdpStatus err;
+
+    av_assert0(frame->format == AV_PIX_FMT_VDPAU);
+
+    surface = av_malloc(sizeof(*surface));
+    if (!surface)
+        return AVERROR(ENOMEM);
+
+    frame->buf[0] = av_buffer_create((uint8_t*)surface, sizeof(*surface),
+                                     vdpau_release_buffer, ctx,
+                                     AV_BUFFER_FLAG_READONLY);
+    if (!frame->buf[0]) {
+        av_freep(&surface);
+        return AVERROR(ENOMEM);
+    }
+
+    // properly we should keep a pool of surfaces instead of creating
+    // them anew for each frame, but since we don't care about speed
+    // much in this code, we don't bother
+    err = ctx->video_surface_create(ctx->device, VDP_CHROMA_TYPE_420,
+                                    frame->width, frame->height, surface);
+    if (err != VDP_STATUS_OK) {
+        av_log(NULL, AV_LOG_ERROR, "Error allocating a VDPAU video surface: %s\n",
+               ctx->get_error_string(err));
+        av_buffer_unref(&frame->buf[0]);
+        return AVERROR_UNKNOWN;
+    }
+
+    frame->data[3] = (uint8_t*)(uintptr_t)*surface;
+
+    return 0;
+}
+
+static int vdpau_retrieve_data(AVCodecContext *s, AVFrame *frame)
+{
+    VdpVideoSurface surface = (VdpVideoSurface)(uintptr_t)frame->data[3];
+    InputStream        *ist = s->opaque;
+    VDPAUContext       *ctx = ist->hwaccel_ctx;
+    VdpStatus err;
+    int ret, chroma_type;
+
+    err = ctx->video_surface_get_parameters(surface, &chroma_type,
+                                            &ctx->tmp_frame->width,
+                                            &ctx->tmp_frame->height);
+    if (err != VDP_STATUS_OK) {
+        av_log(NULL, AV_LOG_ERROR, "Error getting surface parameters: %s\n",
+               ctx->get_error_string(err));
+        return AVERROR_UNKNOWN;
+    }
+    ctx->tmp_frame->format = ctx->pix_fmt;
+
+    ret = av_frame_get_buffer(ctx->tmp_frame, 32);
+    if (ret < 0)
+        return ret;
+
+    ctx->tmp_frame->width  = frame->width;
+    ctx->tmp_frame->height = frame->height;
+
+    err = ctx->video_surface_get_bits(surface, ctx->vdpau_format,
+                                      (void * const *)ctx->tmp_frame->data,
+                                      ctx->tmp_frame->linesize);
+    if (err != VDP_STATUS_OK) {
+        av_log(NULL, AV_LOG_ERROR, "Error retrieving frame data from VDPAU: %s\n",
+               ctx->get_error_string(err));
+        ret = AVERROR_UNKNOWN;
+        goto fail;
+    }
+
+    if (ctx->vdpau_format == VDP_YCBCR_FORMAT_YV12)
+        FFSWAP(uint8_t*, ctx->tmp_frame->data[1], ctx->tmp_frame->data[2]);
+
+    ret = av_frame_copy_props(ctx->tmp_frame, frame);
+    if (ret < 0)
+        goto fail;
+
+    av_frame_unref(frame);
+    av_frame_move_ref(frame, ctx->tmp_frame);
+    return 0;
+
+fail:
+    av_frame_unref(ctx->tmp_frame);
+    return ret;
+}
+
+static const int vdpau_formats[][2] = {
+    { VDP_YCBCR_FORMAT_YV12, AV_PIX_FMT_YUV420P },
+    { VDP_YCBCR_FORMAT_NV12, AV_PIX_FMT_NV12 },
+    { VDP_YCBCR_FORMAT_YUYV, AV_PIX_FMT_YUYV422 },
+    { VDP_YCBCR_FORMAT_UYVY, AV_PIX_FMT_UYVY422 },
+};
+
+static int vdpau_alloc(AVCodecContext *s)
+{
+    InputStream  *ist = s->opaque;
+    int loglevel = (ist->hwaccel_id == HWACCEL_AUTO) ? AV_LOG_VERBOSE : AV_LOG_ERROR;
+    AVVDPAUContext *vdpau_ctx;
+    VDPAUContext *ctx;
+    const char *display, *vendor;
+    VdpStatus err;
+    int i;
+
+    ctx = av_mallocz(sizeof(*ctx));
+    if (!ctx)
+        return AVERROR(ENOMEM);
+
+    ist->hwaccel_ctx           = ctx;
+    ist->hwaccel_uninit        = vdpau_uninit;
+    ist->hwaccel_get_buffer    = vdpau_get_buffer;
+    ist->hwaccel_retrieve_data = vdpau_retrieve_data;
+
+    ctx->tmp_frame = av_frame_alloc();
+    if (!ctx->tmp_frame)
+        goto fail;
+
+    ctx->dpy = XOpenDisplay(ist->hwaccel_device);
+    if (!ctx->dpy) {
+        av_log(NULL, loglevel, "Cannot open the X11 display %s.\n",
+               XDisplayName(ist->hwaccel_device));
+        goto fail;
+    }
+    display = XDisplayString(ctx->dpy);
+
+    err = vdp_device_create_x11(ctx->dpy, XDefaultScreen(ctx->dpy), &ctx->device,
+                                &ctx->get_proc_address);
+    if (err != VDP_STATUS_OK) {
+        av_log(NULL, loglevel, "VDPAU device creation on X11 display %s failed.\n",
+               display);
+        goto fail;
+    }
+
+#define GET_CALLBACK(id, result)                                                \
+do {                                                                            \
+    void *tmp;                                                                  \
+    err = ctx->get_proc_address(ctx->device, id, &tmp);                         \
+    if (err != VDP_STATUS_OK) {                                                 \
+        av_log(NULL, loglevel, "Error getting the " #id " callback.\n");        \
+        goto fail;                                                              \
+    }                                                                           \
+    ctx->result = tmp;                                                          \
+} while (0)
+
+    GET_CALLBACK(VDP_FUNC_ID_GET_ERROR_STRING,               get_error_string);
+    GET_CALLBACK(VDP_FUNC_ID_GET_INFORMATION_STRING,         get_information_string);
+    GET_CALLBACK(VDP_FUNC_ID_DEVICE_DESTROY,                 device_destroy);
+    GET_CALLBACK(VDP_FUNC_ID_DECODER_CREATE,                 decoder_create);
+    GET_CALLBACK(VDP_FUNC_ID_DECODER_DESTROY,                decoder_destroy);
+    GET_CALLBACK(VDP_FUNC_ID_DECODER_RENDER,                 decoder_render);
+    GET_CALLBACK(VDP_FUNC_ID_VIDEO_SURFACE_CREATE,           video_surface_create);
+    GET_CALLBACK(VDP_FUNC_ID_VIDEO_SURFACE_DESTROY,          video_surface_destroy);
+    GET_CALLBACK(VDP_FUNC_ID_VIDEO_SURFACE_GET_BITS_Y_CB_CR, video_surface_get_bits);
+    GET_CALLBACK(VDP_FUNC_ID_VIDEO_SURFACE_GET_PARAMETERS,   video_surface_get_parameters);
+    GET_CALLBACK(VDP_FUNC_ID_VIDEO_SURFACE_QUERY_GET_PUT_BITS_Y_CB_CR_CAPABILITIES,
+                 video_surface_query);
+
+    for (i = 0; i < FF_ARRAY_ELEMS(vdpau_formats); i++) {
+        VdpBool supported;
+        err = ctx->video_surface_query(ctx->device, VDP_CHROMA_TYPE_420,
+                                       vdpau_formats[i][0], &supported);
+        if (err != VDP_STATUS_OK) {
+            av_log(NULL, loglevel,
+                   "Error querying VDPAU surface capabilities: %s\n",
+                   ctx->get_error_string(err));
+            goto fail;
+        }
+        if (supported)
+            break;
+    }
+    if (i == FF_ARRAY_ELEMS(vdpau_formats)) {
+        av_log(NULL, loglevel,
+               "No supported VDPAU format for retrieving the data.\n");
+        return AVERROR(EINVAL);
+    }
+    ctx->vdpau_format = vdpau_formats[i][0];
+    ctx->pix_fmt      = vdpau_formats[i][1];
+
+    vdpau_ctx = av_vdpau_alloc_context();
+    if (!vdpau_ctx)
+        goto fail;
+    vdpau_ctx->render = ctx->decoder_render;
+
+    s->hwaccel_context = vdpau_ctx;
+
+    ctx->get_information_string(&vendor);
+    av_log(NULL, AV_LOG_VERBOSE, "Using VDPAU -- %s -- on X11 display %s, "
+           "to decode input stream #%d:%d.\n", vendor,
+           display, ist->file_index, ist->st->index);
+
+    return 0;
+
+fail:
+    av_log(NULL, loglevel, "VDPAU init failed for stream #%d:%d.\n",
+           ist->file_index, ist->st->index);
+    vdpau_uninit(s);
+    return AVERROR(EINVAL);
+}
+
+int vdpau_init(AVCodecContext *s)
+{
+    InputStream *ist = s->opaque;
+    int loglevel = (ist->hwaccel_id == HWACCEL_AUTO) ? AV_LOG_VERBOSE : AV_LOG_ERROR;
+    AVVDPAUContext *vdpau_ctx;
+    VDPAUContext *ctx;
+    VdpStatus err;
+    int profile, ret;
+
+    if (!ist->hwaccel_ctx) {
+        ret = vdpau_alloc(s);
+        if (ret < 0)
+            return ret;
+    }
+    ctx       = ist->hwaccel_ctx;
+    vdpau_ctx = s->hwaccel_context;
+
+    ret = av_vdpau_get_profile(s, &profile);
+    if (ret < 0) {
+        av_log(NULL, loglevel, "No known VDPAU decoder profile for this stream.\n");
+        return AVERROR(EINVAL);
+    }
+
+    if (ctx->decoder)
+        ctx->decoder_destroy(ctx->decoder);
+
+    err = ctx->decoder_create(ctx->device, profile,
+                              s->coded_width, s->coded_height,
+                              16, &ctx->decoder);
+    if (err != VDP_STATUS_OK) {
+        av_log(NULL, loglevel, "Error creating the VDPAU decoder: %s\n",
+               ctx->get_error_string(err));
+        return AVERROR_UNKNOWN;
+    }
+
+    vdpau_ctx->decoder = ctx->decoder;
+
+    ist->hwaccel_get_buffer    = vdpau_get_buffer;
+    ist->hwaccel_retrieve_data = vdpau_retrieve_data;
+
+    return 0;
+}

ffplay.c

@@ -28,6 +28,8 @@
 #include <math.h>
 #include <limits.h>
 #include <signal.h>
+#include <stdint.h>
+
 #include "libavutil/avstring.h"
 #include "libavutil/colorspace.h"
 #include "libavutil/mathematics.h"
@@ -121,6 +123,7 @@ typedef struct PacketQueue {
 
 typedef struct VideoPicture {
     double pts;             // presentation timestamp for this picture
+    double duration;        // estimated duration based on frame rate
     int64_t pos;            // byte position in file
     SDL_Overlay *bmp;
     int width, height; /* source height & width */
@@ -142,6 +145,8 @@ typedef struct AudioParams {
     int channels;
     int64_t channel_layout;
     enum AVSampleFormat fmt;
+    int frame_size;
+    int bytes_per_sec;
 } AudioParams;
 
 typedef struct Clock {
@@ -242,13 +247,8 @@ typedef struct VideoState {
     SDL_cond *subpq_cond;
 
     double frame_timer;
-    double frame_last_pts;
-    double frame_last_duration;
-    double frame_last_dropped_pts;
     double frame_last_returned_time;
     double frame_last_filter_delay;
-    int64_t frame_last_dropped_pos;
-    int frame_last_dropped_serial;
     int video_stream;
     AVStream *video_st;
     PacketQueue videoq;
@@ -359,8 +359,6 @@ int64_t get_valid_channel_layout(int64_t channel_layout, int channels)
         return 0;
 }
 
-static int packet_queue_put(PacketQueue *q, AVPacket *pkt);
-
 static int packet_queue_put_private(PacketQueue *q, AVPacket *pkt)
 {
     MyAVPacketList *pkt1;
@@ -1068,20 +1066,24 @@ static void sigterm_handler(int sig)
     exit(123);
 }
 
+static void set_default_window_size(VideoPicture *vp)
+{
+    SDL_Rect rect;
+    calculate_display_rect(&rect, 0, 0, INT_MAX, vp->height, vp);
+    default_width  = rect.w;
+    default_height = rect.h;
+}
+
 static int video_open(VideoState *is, int force_set_video_mode, VideoPicture *vp)
 {
     int flags = SDL_HWSURFACE | SDL_ASYNCBLIT | SDL_HWACCEL;
     int w,h;
-    SDL_Rect rect;
 
     if (is_full_screen) flags |= SDL_FULLSCREEN;
     else                flags |= SDL_RESIZABLE;
 
-    if (vp && vp->width) {
-        calculate_display_rect(&rect, 0, 0, INT_MAX, vp->height, vp);
-        default_width  = rect.w;
-        default_height = rect.h;
-    }
+    if (vp && vp->width)
+        set_default_window_size(vp);
 
     if (is_full_screen && fs_screen_width) {
         w = fs_screen_width;
@@ -1292,6 +1294,18 @@ static double compute_target_delay(double delay, VideoState *is)
     return delay;
 }
 
+static double vp_duration(VideoState *is, VideoPicture *vp, VideoPicture *nextvp) {
+    if (vp->serial == nextvp->serial) {
+        double duration = nextvp->pts - vp->pts;
+        if (isnan(duration) || duration <= 0 || duration > is->max_frame_duration)
+            return vp->duration;
+        else
+            return duration;
+    } else {
+        return 0.0;
+    }
+}
+
 static void pictq_next_picture(VideoState *is) {
     /* update queue size and signal for next picture */
     if (++is->pictq_rindex == VIDEO_PICTURE_QUEUE_SIZE)
@@ -1327,14 +1341,12 @@ static void update_video_pts(VideoState *is, double pts, int64_t pos, int serial
     set_clock(&is->vidclk, pts, serial);
     sync_clock_to_slave(&is->extclk, &is->vidclk);
     is->video_current_pos = pos;
-    is->frame_last_pts = pts;
 }
 
 /* called to display each frame */
 static void video_refresh(void *opaque, double *remaining_time)
 {
     VideoState *is = opaque;
-    VideoPicture *vp;
     double time;
 
     SubPicture *sp, *sp2;
@@ -1357,37 +1369,34 @@ static void video_refresh(void *opaque, double *remaining_time)
             redisplay = pictq_prev_picture(is);
 retry:
         if (is->pictq_size == 0) {
-            SDL_LockMutex(is->pictq_mutex);
-            if (is->frame_last_dropped_pts != AV_NOPTS_VALUE && is->frame_last_dropped_pts > is->frame_last_pts) {
-                update_video_pts(is, is->frame_last_dropped_pts, is->frame_last_dropped_pos, is->frame_last_dropped_serial);
-                is->frame_last_dropped_pts = AV_NOPTS_VALUE;
-            }
-            SDL_UnlockMutex(is->pictq_mutex);
             // nothing to do, no picture to display in the queue
         } else {
             double last_duration, duration, delay;
+            VideoPicture *vp, *lastvp;
+
             /* dequeue the picture */
             vp = &is->pictq[is->pictq_rindex];
+            lastvp = &is->pictq[(is->pictq_rindex + VIDEO_PICTURE_QUEUE_SIZE - 1) % VIDEO_PICTURE_QUEUE_SIZE];
 
             if (vp->serial != is->videoq.serial) {
                 pictq_next_picture(is);
+                is->video_current_pos = -1;
                 redisplay = 0;
                 goto retry;
             }
 
+            if (lastvp->serial != vp->serial && !redisplay)
+                is->frame_timer = av_gettime() / 1000000.0;
+
             if (is->paused)
                 goto display;
 
             /* compute nominal last_duration */
-            last_duration = vp->pts - is->frame_last_pts;
-            if (!isnan(last_duration) && last_duration > 0 && last_duration < is->max_frame_duration) {
-                /* if duration of the last frame was sane, update last_duration in video state */
-                is->frame_last_duration = last_duration;
-            }
+            last_duration = vp_duration(is, lastvp, vp);
             if (redisplay)
                 delay = 0.0;
             else
-                delay = compute_target_delay(is->frame_last_duration, is);
+                delay = compute_target_delay(last_duration, is);
 
             time= av_gettime()/1000000.0;
             if (time < is->frame_timer + delay && !redisplay) {
@@ -1406,7 +1415,7 @@ retry:
 
             if (is->pictq_size > 1) {
                 VideoPicture *nextvp = &is->pictq[(is->pictq_rindex + 1) % VIDEO_PICTURE_QUEUE_SIZE];
-                duration = nextvp->pts - vp->pts;
+                duration = vp_duration(is, vp, nextvp);
                 if(!is->step && (redisplay || framedrop>0 || (framedrop && get_master_sync_type(is) != AV_SYNC_VIDEO_MASTER)) && time > is->frame_timer + duration){
                     if (!redisplay)
                         is->frame_drops_late++;
@@ -1549,7 +1558,7 @@ static void duplicate_right_border_pixels(SDL_Overlay *bmp) {
     }
 }
 
-static int queue_picture(VideoState *is, AVFrame *src_frame, double pts, int64_t pos, int serial)
+static int queue_picture(VideoState *is, AVFrame *src_frame, double pts, double duration, int64_t pos, int serial)
 {
     VideoPicture *vp;
 
@@ -1599,7 +1608,7 @@ static int queue_picture(VideoState *is, AVFrame *src_frame, double pts, int64_t
         }
         /* if the queue is aborted, we have to pop the pending ALLOC event or wait for the allocation to complete */
         if (is->videoq.abort_request && SDL_PeepEvents(&event, 1, SDL_GETEVENT, SDL_EVENTMASK(FF_ALLOC_EVENT)) != 1) {
-            while (!vp->allocated) {
+            while (!vp->allocated && !is->abort_request) {
                 SDL_CondWait(is->pictq_cond, is->pictq_mutex);
             }
         }
@@ -1646,6 +1655,7 @@ static int queue_picture(VideoState *is, AVFrame *src_frame, double pts, int64_t
         SDL_UnlockYUVOverlay(vp->bmp);
 
         vp->pts = pts;
+        vp->duration = duration;
         vp->pos = pos;
         vp->serial = serial;
 
@@ -1668,18 +1678,6 @@ static int get_video_frame(VideoState *is, AVFrame *frame, AVPacket *pkt, int *s
 
     if (pkt->data == flush_pkt.data) {
         avcodec_flush_buffers(is->video_st->codec);
-
-        SDL_LockMutex(is->pictq_mutex);
-        // Make sure there are no long delay timers (ideally we should just flush the queue but that's harder)
-        while (is->pictq_size && !is->videoq.abort_request) {
-            SDL_CondWait(is->pictq_cond, is->pictq_mutex);
-        }
-        is->video_current_pos = -1;
-        is->frame_last_pts = AV_NOPTS_VALUE;
-        is->frame_last_duration = 0;
-        is->frame_timer = (double)av_gettime() / 1000000.0;
-        is->frame_last_dropped_pts = AV_NOPTS_VALUE;
-        SDL_UnlockMutex(is->pictq_mutex);
         return 0;
     }
 
@@ -1707,23 +1705,17 @@ static int get_video_frame(VideoState *is, AVFrame *frame, AVPacket *pkt, int *s
         frame->sample_aspect_ratio = av_guess_sample_aspect_ratio(is->ic, is->video_st, frame);
 
         if (framedrop>0 || (framedrop && get_master_sync_type(is) != AV_SYNC_VIDEO_MASTER)) {
-            SDL_LockMutex(is->pictq_mutex);
-            if (is->frame_last_pts != AV_NOPTS_VALUE && frame->pts != AV_NOPTS_VALUE) {
-                double clockdiff = get_clock(&is->vidclk) - get_master_clock(is);
-                double ptsdiff = dpts - is->frame_last_pts;
-                if (!isnan(clockdiff) && fabs(clockdiff) < AV_NOSYNC_THRESHOLD &&
-                    !isnan(ptsdiff) && ptsdiff > 0 && ptsdiff < AV_NOSYNC_THRESHOLD &&
-                    clockdiff + ptsdiff - is->frame_last_filter_delay < 0 &&
+            if (frame->pts != AV_NOPTS_VALUE) {
+                double diff = dpts - get_master_clock(is);
+                if (!isnan(diff) && fabs(diff) < AV_NOSYNC_THRESHOLD &&
+                    diff - is->frame_last_filter_delay < 0 &&
+                    *serial == is->vidclk.serial &&
                     is->videoq.nb_packets) {
-                    is->frame_last_dropped_pos = av_frame_get_pkt_pos(frame);
-                    is->frame_last_dropped_pts = dpts;
-                    is->frame_last_dropped_serial = *serial;
                     is->frame_drops_early++;
                     av_frame_unref(frame);
                     ret = 0;
                 }
             }
-            SDL_UnlockMutex(is->pictq_mutex);
         }
 
         return ret;
@@ -1735,7 +1727,8 @@ static int get_video_frame(VideoState *is, AVFrame *frame, AVPacket *pkt, int *s
 static int configure_filtergraph(AVFilterGraph *graph, const char *filtergraph,
                                  AVFilterContext *source_ctx, AVFilterContext *sink_ctx)
 {
-    int ret;
+    int ret, i;
+    int nb_filters = graph->nb_filters;
     AVFilterInOut *outputs = NULL, *inputs = NULL;
 
     if (filtergraph) {
@@ -1763,6 +1756,10 @@ static int configure_filtergraph(AVFilterGraph *graph, const char *filtergraph,
             goto fail;
     }
 
+    /* Reorder the filters to ensure that inputs of the custom filters are merged first */
+    for (i = 0; i < graph->nb_filters - nb_filters; i++)
+        FFSWAP(AVFilterContext*, graph->filters[i], graph->filters[i + nb_filters]);
+
     ret = avfilter_graph_config(graph, NULL);
 fail:
     avfilter_inout_free(&outputs);
@@ -1909,8 +1906,11 @@ static int video_thread(void *arg)
     VideoState *is = arg;
     AVFrame *frame = av_frame_alloc();
     double pts;
+    double duration;
     int ret;
     int serial = 0;
+    AVRational tb = is->video_st->time_base;
+    AVRational frame_rate = av_guess_frame_rate(is->ic, is->video_st, NULL);
 
 #if CONFIG_AVFILTER
     AVFilterGraph *graph = avfilter_graph_alloc();
@@ -1925,7 +1925,6 @@ static int video_thread(void *arg)
         while (is->paused && !is->videoq.abort_request)
             SDL_Delay(10);
 
-        avcodec_get_frame_defaults(frame);
         av_free_packet(&pkt);
 
         ret = get_video_frame(is, frame, &pkt, &serial);
@@ -1952,7 +1951,6 @@ static int video_thread(void *arg)
                 event.type = FF_QUIT_EVENT;
                 event.user.data1 = is;
                 SDL_PushEvent(&event);
-                av_free_packet(&pkt);
                 goto the_end;
             }
             filt_in  = is->in_video_filter;
@@ -1961,14 +1959,12 @@ static int video_thread(void *arg)
             last_h = frame->height;
             last_format = frame->format;
             last_serial = serial;
+            frame_rate = filt_out->inputs[0]->frame_rate;
         }
 
         ret = av_buffersrc_add_frame(filt_in, frame);
         if (ret < 0)
             goto the_end;
-        av_frame_unref(frame);
-        avcodec_get_frame_defaults(frame);
-        av_free_packet(&pkt);
 
         while (ret >= 0) {
             is->frame_last_returned_time = av_gettime() / 1000000.0;
@@ -1984,22 +1980,20 @@ static int video_thread(void *arg)
             is->frame_last_filter_delay = av_gettime() / 1000000.0 - is->frame_last_returned_time;
             if (fabs(is->frame_last_filter_delay) > AV_NOSYNC_THRESHOLD / 10.0)
                 is->frame_last_filter_delay = 0;
-
-            pts = (frame->pts == AV_NOPTS_VALUE) ? NAN : frame->pts * av_q2d(filt_out->inputs[0]->time_base);
-            ret = queue_picture(is, frame, pts, av_frame_get_pkt_pos(frame), serial);
+            tb = filt_out->inputs[0]->time_base;
+#endif
+            duration = (frame_rate.num && frame_rate.den ? av_q2d((AVRational){frame_rate.den, frame_rate.num}) : 0);
+            pts = (frame->pts == AV_NOPTS_VALUE) ? NAN : frame->pts * av_q2d(tb);
+            ret = queue_picture(is, frame, pts, duration, av_frame_get_pkt_pos(frame), serial);
             av_frame_unref(frame);
+#if CONFIG_AVFILTER
         }
-#else
-        pts = (frame->pts == AV_NOPTS_VALUE) ? NAN : frame->pts * av_q2d(is->video_st->time_base);
-        ret = queue_picture(is, frame, pts, av_frame_get_pkt_pos(frame), serial);
-        av_frame_unref(frame);
 #endif
 
         if (ret < 0)
             goto the_end;
     }
  the_end:
-    avcodec_flush_buffers(is->video_st->codec);
 #if CONFIG_AVFILTER
     avfilter_graph_free(&graph);
 #endif
@@ -2169,11 +2163,10 @@ static int audio_decode_frame(VideoState *is)
         /* NOTE: the audio packet can contain several frames */
         while (pkt_temp->stream_index != -1 || is->audio_buf_frames_pending) {
             if (!is->frame) {
-                if (!(is->frame = avcodec_alloc_frame()))
+                if (!(is->frame = av_frame_alloc()))
                     return AVERROR(ENOMEM);
             } else {
                 av_frame_unref(is->frame);
-                avcodec_get_frame_defaults(is->frame);
             }
 
             if (is->audioq.serial != is->audio_pkt_temp_serial)
@@ -2248,7 +2241,6 @@ static int audio_decode_frame(VideoState *is)
 
                 if ((ret = av_buffersrc_add_frame(is->in_audio_filter, is->frame)) < 0)
                     return ret;
-                av_frame_unref(is->frame);
 #endif
             }
 #if CONFIG_AVFILTER
@@ -2385,8 +2377,6 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
 {
     VideoState *is = opaque;
     int audio_size, len1;
-    int bytes_per_sec;
-    int frame_size = av_samples_get_buffer_size(NULL, is->audio_tgt.channels, 1, is->audio_tgt.fmt, 1);
 
     audio_callback_time = av_gettime();
 
@@ -2396,7 +2386,7 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
            if (audio_size < 0) {
                 /* if error, just output silence */
                is->audio_buf      = is->silence_buf;
-               is->audio_buf_size = sizeof(is->silence_buf) / frame_size * frame_size;
+               is->audio_buf_size = sizeof(is->silence_buf) / is->audio_tgt.frame_size * is->audio_tgt.frame_size;
            } else {
                if (is->show_mode != SHOW_MODE_VIDEO)
                    update_sample_display(is, (int16_t *)is->audio_buf, audio_size);
@@ -2412,11 +2402,10 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
         stream += len1;
         is->audio_buf_index += len1;
     }
-    bytes_per_sec = is->audio_tgt.freq * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt);
     is->audio_write_buf_size = is->audio_buf_size - is->audio_buf_index;
     /* Let's assume the audio driver that is used by SDL has two periods. */
     if (!isnan(is->audio_clock)) {
-        set_clock_at(&is->audclk, is->audio_clock - (double)(2 * is->audio_hw_buf_size + is->audio_write_buf_size) / bytes_per_sec, is->audio_clock_serial, audio_callback_time / 1000000.0);
+        set_clock_at(&is->audclk, is->audio_clock - (double)(2 * is->audio_hw_buf_size + is->audio_write_buf_size) / is->audio_tgt.bytes_per_sec, is->audio_clock_serial, audio_callback_time / 1000000.0);
         sync_clock_to_slave(&is->extclk, &is->audclk);
     }
 }
@@ -2475,6 +2464,12 @@ static int audio_open(void *opaque, int64_t wanted_channel_layout, int wanted_nb
     audio_hw_params->freq = spec.freq;
     audio_hw_params->channel_layout = wanted_channel_layout;
     audio_hw_params->channels =  spec.channels;
+    audio_hw_params->frame_size = av_samples_get_buffer_size(NULL, audio_hw_params->channels, 1, audio_hw_params->fmt, 1);
+    audio_hw_params->bytes_per_sec = av_samples_get_buffer_size(NULL, audio_hw_params->channels, audio_hw_params->freq, audio_hw_params->fmt, 1);
+    if (audio_hw_params->bytes_per_sec <= 0 || audio_hw_params->frame_size <= 0) {
+        av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size failed\n");
+        return -1;
+    }
     return spec.size;
 }
 
@@ -2579,7 +2574,7 @@ static int stream_component_open(VideoState *is, int stream_index)
         is->audio_diff_avg_count = 0;
         /* since we do not have a precise anough audio fifo fullness,
            we correct audio sync only if larger than this threshold */
-        is->audio_diff_threshold = 2.0 * is->audio_hw_buf_size / av_samples_get_buffer_size(NULL, is->audio_tgt.channels, is->audio_tgt.freq, is->audio_tgt.fmt, 1);
+        is->audio_diff_threshold = 2.0 * is->audio_hw_buf_size / is->audio_tgt.bytes_per_sec;
 
         memset(&is->audio_pkt, 0, sizeof(is->audio_pkt));
         memset(&is->audio_pkt_temp, 0, sizeof(is->audio_pkt_temp));
@@ -2824,6 +2819,13 @@ static int read_thread(void *arg)
     }
 
     is->show_mode = show_mode;
+    if (st_index[AVMEDIA_TYPE_VIDEO] >= 0) {
+        AVStream *st = ic->streams[st_index[AVMEDIA_TYPE_VIDEO]];
+        AVCodecContext *avctx = st->codec;
+        VideoPicture vp = {.width = avctx->width, .height = avctx->height, .sar = av_guess_sample_aspect_ratio(ic, st, NULL)};
+        if (vp.width)
+            set_default_window_size(&vp);
+    }
 
     /* open the streams */
     if (st_index[AVMEDIA_TYPE_AUDIO] >= 0) {
@@ -2946,6 +2948,8 @@ static int read_thread(void *arg)
                 packet_queue_put_nullpacket(&is->videoq, is->video_stream);
             if (is->audio_stream >= 0)
                 packet_queue_put_nullpacket(&is->audioq, is->audio_stream);
+            if (is->subtitle_stream >= 0)
+                packet_queue_put_nullpacket(&is->subtitleq, is->subtitle_stream);
             SDL_Delay(10);
             eof=0;
             continue;
@@ -3166,6 +3170,33 @@ static void refresh_loop_wait_event(VideoState *is, SDL_Event *event) {
     }
 }
 
+static void seek_chapter(VideoState *is, int incr)
+{
+    int64_t pos = get_master_clock(is) * AV_TIME_BASE;
+    int i;
+
+    if (!is->ic->nb_chapters)
+        return;
+
+    /* find the current chapter */
+    for (i = 0; i < is->ic->nb_chapters; i++) {
+        AVChapter *ch = is->ic->chapters[i];
+        if (av_compare_ts(pos, AV_TIME_BASE_Q, ch->start, ch->time_base) < 0) {
+            i--;
+            break;
+        }
+    }
+
+    i += incr;
+    i = FFMAX(i, 0);
+    if (i >= is->ic->nb_chapters)
+        return;
+
+    av_log(NULL, AV_LOG_VERBOSE, "Seeking to chapter %d.\n", i);
+    stream_seek(is, av_rescale_q(is->ic->chapters[i]->start, is->ic->chapters[i]->time_base,
+                                 AV_TIME_BASE_Q), 0, 0);
+}
+
 /* handle an event sent by the GUI */
 static void event_loop(VideoState *cur_stream)
 {
@@ -3215,11 +3246,19 @@ static void event_loop(VideoState *cur_stream)
                 toggle_audio_display(cur_stream);
                 break;
             case SDLK_PAGEUP:
-                incr = 600.0;
-                goto do_seek;
+                if (cur_stream->ic->nb_chapters <= 1) {
+                    incr = 600.0;
+                    goto do_seek;
+                }
+                seek_chapter(cur_stream, 1);
+                break;
             case SDLK_PAGEDOWN:
-                incr = -600.0;
-                goto do_seek;
+                if (cur_stream->ic->nb_chapters <= 1) {
+                    incr = -600.0;
+                    goto do_seek;
+                }
+                seek_chapter(cur_stream, -1);
+                break;
             case SDLK_LEFT:
                 incr = -10.0;
                 goto do_seek;
@@ -3552,7 +3591,6 @@ int main(int argc, char **argv)
     parse_loglevel(argc, argv, options);
 
     /* register all codecs, demux and protocols */
-    avcodec_register_all();
 #if CONFIG_AVDEVICE
     avdevice_register_all();
 #endif

ffprobe.c

@@ -24,7 +24,7 @@
  */
 
 #include "config.h"
-#include "version.h"
+#include "libavutil/ffversion.h"
 
 #include <string.h>
 
@@ -66,6 +66,12 @@ static int do_show_data    = 0;
 static int do_show_program_version  = 0;
 static int do_show_library_versions = 0;
 
+static int do_show_chapter_tags = 0;
+static int do_show_format_tags = 0;
+static int do_show_frame_tags = 0;
+static int do_show_program_tags = 0;
+static int do_show_stream_tags = 0;
+
 static int show_value_unit              = 0;
 static int use_value_prefix             = 0;
 static int use_byte_value_binary_prefix = 0;
@@ -135,6 +141,7 @@ typedef enum {
     SECTION_ID_STREAM_DISPOSITION,
     SECTION_ID_STREAMS,
     SECTION_ID_STREAM_TAGS,
+    SECTION_ID_SUBTITLE,
 } SectionID;
 
 static struct section sections[] = {
@@ -144,7 +151,7 @@ static struct section sections[] = {
     [SECTION_ID_ERROR] =              { SECTION_ID_ERROR, "error", 0, { -1 } },
     [SECTION_ID_FORMAT] =             { SECTION_ID_FORMAT, "format", 0, { SECTION_ID_FORMAT_TAGS, -1 } },
     [SECTION_ID_FORMAT_TAGS] =        { SECTION_ID_FORMAT_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "format_tags" },
-    [SECTION_ID_FRAMES] =             { SECTION_ID_FRAMES, "frames", SECTION_FLAG_IS_ARRAY, { SECTION_ID_FRAME, -1 } },
+    [SECTION_ID_FRAMES] =             { SECTION_ID_FRAMES, "frames", SECTION_FLAG_IS_ARRAY, { SECTION_ID_FRAME, SECTION_ID_SUBTITLE, -1 } },
     [SECTION_ID_FRAME] =              { SECTION_ID_FRAME, "frame", 0, { SECTION_ID_FRAME_TAGS, -1 } },
     [SECTION_ID_FRAME_TAGS] =         { SECTION_ID_FRAME_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "frame_tags" },
     [SECTION_ID_LIBRARY_VERSIONS] =   { SECTION_ID_LIBRARY_VERSIONS, "library_versions", SECTION_FLAG_IS_ARRAY, { SECTION_ID_LIBRARY_VERSION, -1 } },
@@ -167,6 +174,7 @@ static struct section sections[] = {
     [SECTION_ID_STREAM] =             { SECTION_ID_STREAM, "stream", 0, { SECTION_ID_STREAM_DISPOSITION, SECTION_ID_STREAM_TAGS, -1 } },
     [SECTION_ID_STREAM_DISPOSITION] = { SECTION_ID_STREAM_DISPOSITION, "disposition", 0, { -1 }, .unique_name = "stream_disposition" },
     [SECTION_ID_STREAM_TAGS] =        { SECTION_ID_STREAM_TAGS, "tags", SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "stream_tags" },
+    [SECTION_ID_SUBTITLE] =           { SECTION_ID_SUBTITLE, "subtitle", 0, { -1 } },
 };
 
 static const OptionDef *options;
@@ -183,6 +191,7 @@ static const char unit_hertz_str[]          = "Hz"   ;
 static const char unit_byte_str[]           = "byte" ;
 static const char unit_bit_per_second_str[] = "bit/s";
 
+static int nb_streams;
 static uint64_t *nb_streams_packets;
 static uint64_t *nb_streams_frames;
 static int *selected_streams;
@@ -238,6 +247,7 @@ static char *value_string(char *buf, int buf_size, struct unit_value uv)
                 vald /= pow(10, index * 3);
                 prefix_string = decimal_unit_prefixes[index];
             }
+            vali = vald;
         }
 
         if (show_float || (use_value_prefix && vald != (long long int)vald))
@@ -258,6 +268,13 @@ typedef struct WriterContext WriterContext;
 #define WRITER_FLAG_DISPLAY_OPTIONAL_FIELDS 1
 #define WRITER_FLAG_PUT_PACKETS_AND_FRAMES_IN_SAME_CHAPTER 2
 
+typedef enum {
+    WRITER_STRING_VALIDATION_FAIL,
+    WRITER_STRING_VALIDATION_REPLACE,
+    WRITER_STRING_VALIDATION_IGNORE,
+    WRITER_STRING_VALIDATION_NB
+} StringValidation;
+
 typedef struct Writer {
     const AVClass *priv_class;      ///< private class of the writer, if any
     int priv_size;                  ///< private size for the writer context
@@ -298,6 +315,10 @@ struct WriterContext {
     unsigned int nb_section_packet; ///< number of the packet section in case we are in "packets_and_frames" section
     unsigned int nb_section_frame;  ///< number of the frame  section in case we are in "packets_and_frames" section
     unsigned int nb_section_packet_frame; ///< nb_section_packet or nb_section_frame according if is_packets_and_frames
+
+    StringValidation string_validation;
+    char *string_validation_replacement;
+    unsigned int string_validation_utf8_flags;
 };
 
 static const char *writer_get_name(void *p)
@@ -306,11 +327,35 @@ static const char *writer_get_name(void *p)
     return wctx->writer->name;
 }
 
+#define OFFSET(x) offsetof(WriterContext, x)
+
+static const AVOption writer_options[] = {
+    { "string_validation", "set string validation mode",
+      OFFSET(string_validation), AV_OPT_TYPE_INT, {.i64=WRITER_STRING_VALIDATION_REPLACE}, 0, WRITER_STRING_VALIDATION_NB-1, .unit = "sv" },
+    { "sv", "set string validation mode",
+      OFFSET(string_validation), AV_OPT_TYPE_INT, {.i64=WRITER_STRING_VALIDATION_REPLACE}, 0, WRITER_STRING_VALIDATION_NB-1, .unit = "sv" },
+    { "ignore",  NULL, 0, AV_OPT_TYPE_CONST, {.i64 = WRITER_STRING_VALIDATION_IGNORE},  .unit = "sv" },
+    { "replace", NULL, 0, AV_OPT_TYPE_CONST, {.i64 = WRITER_STRING_VALIDATION_REPLACE}, .unit = "sv" },
+    { "fail",    NULL, 0, AV_OPT_TYPE_CONST, {.i64 = WRITER_STRING_VALIDATION_FAIL},    .unit = "sv" },
+    { "string_validation_replacement", "set string validation replacement string", OFFSET(string_validation_replacement), AV_OPT_TYPE_STRING, {.str=""}},
+    { "svr", "set string validation replacement string", OFFSET(string_validation_replacement), AV_OPT_TYPE_STRING, {.str="\xEF\xBF\xBD"}},
+    { NULL }
+};
+
+static void *writer_child_next(void *obj, void *prev)
+{
+    WriterContext *ctx = obj;
+    if (!prev && ctx->writer && ctx->writer->priv_class && ctx->priv)
+        return ctx->priv;
+    return NULL;
+}
+
 static const AVClass writer_class = {
-    "Writer",
-    writer_get_name,
-    NULL,
-    LIBAVUTIL_VERSION_INT,
+    .class_name = "Writer",
+    .item_name  = writer_get_name,
+    .option     = writer_options,
+    .version    = LIBAVUTIL_VERSION_INT,
+    .child_next = writer_child_next,
 };
 
 static void writer_close(WriterContext **wctx)
@@ -327,9 +372,19 @@ static void writer_close(WriterContext **wctx)
     if ((*wctx)->writer->priv_class)
         av_opt_free((*wctx)->priv);
     av_freep(&((*wctx)->priv));
+    av_opt_free(*wctx);
     av_freep(wctx);
 }
 
+static void bprint_bytes(AVBPrint *bp, const uint8_t *ubuf, size_t ubuf_size)
+{
+    int i;
+    av_bprintf(bp, "0X");
+    for (i = 0; i < ubuf_size; i++)
+        av_bprintf(bp, "%02X", ubuf[i]);
+}
+
+
 static int writer_open(WriterContext **wctx, const Writer *writer, const char *args,
                        const struct section *sections, int nb_sections)
 {
@@ -351,14 +406,55 @@ static int writer_open(WriterContext **wctx, const Writer *writer, const char *a
     (*wctx)->sections = sections;
     (*wctx)->nb_sections = nb_sections;
 
+    av_opt_set_defaults(*wctx);
+
     if (writer->priv_class) {
         void *priv_ctx = (*wctx)->priv;
         *((const AVClass **)priv_ctx) = writer->priv_class;
         av_opt_set_defaults(priv_ctx);
+    }
 
-        if (args &&
-            (ret = av_set_options_string(priv_ctx, args, "=", ":")) < 0)
+    /* convert options to dictionary */
+    if (args) {
+        AVDictionary *opts = NULL;
+        AVDictionaryEntry *opt = NULL;
+
+        if ((ret = av_dict_parse_string(&opts, args, "=", ":", 0)) < 0) {
+            av_log(*wctx, AV_LOG_ERROR, "Failed to parse option string '%s' provided to writer context\n", args);
+            av_dict_free(&opts);
             goto fail;
+        }
+
+        while ((opt = av_dict_get(opts, "", opt, AV_DICT_IGNORE_SUFFIX))) {
+            if ((ret = av_opt_set(*wctx, opt->key, opt->value, AV_OPT_SEARCH_CHILDREN)) < 0) {
+                av_log(*wctx, AV_LOG_ERROR, "Failed to set option '%s' with value '%s' provided to writer context\n",
+                       opt->key, opt->value);
+                av_dict_free(&opts);
+                goto fail;
+            }
+        }
+
+        av_dict_free(&opts);
+    }
+
+    /* validate replace string */
+    {
+        const uint8_t *p = (*wctx)->string_validation_replacement;
+        const uint8_t *endp = p + strlen(p);
+        while (*p) {
+            const uint8_t *p0 = p;
+            int32_t code;
+            ret = av_utf8_decode(&code, &p, endp, (*wctx)->string_validation_utf8_flags);
+            if (ret < 0) {
+                AVBPrint bp;
+                av_bprint_init(&bp, 0, AV_BPRINT_SIZE_AUTOMATIC);
+                bprint_bytes(&bp, p0, p-p0),
+                    av_log(wctx, AV_LOG_ERROR,
+                           "Invalid UTF8 sequence %s found in string validation replace '%s'\n",
+                           bp.str, (*wctx)->string_validation_replacement);
+                return ret;
+            }
+        }
     }
 
     for (i = 0; i < SECTION_MAX_NB_LEVELS; i++)
@@ -428,18 +524,98 @@ static inline void writer_print_integer(WriterContext *wctx,
     }
 }
 
-static inline void writer_print_string(WriterContext *wctx,
-                                       const char *key, const char *val, int opt)
+static inline int validate_string(WriterContext *wctx, char **dstp, const char *src)
+{
+    const uint8_t *p, *endp;
+    AVBPrint dstbuf;
+    int invalid_chars_nb = 0, ret = 0;
+
+    av_bprint_init(&dstbuf, 0, AV_BPRINT_SIZE_UNLIMITED);
+
+    endp = src + strlen(src);
+    for (p = (uint8_t *)src; *p;) {
+        uint32_t code;
+        int invalid = 0;
+        const uint8_t *p0 = p;
+
+        if (av_utf8_decode(&code, &p, endp, wctx->string_validation_utf8_flags) < 0) {
+            AVBPrint bp;
+            av_bprint_init(&bp, 0, AV_BPRINT_SIZE_AUTOMATIC);
+            bprint_bytes(&bp, p0, p-p0);
+            av_log(wctx, AV_LOG_DEBUG,
+                   "Invalid UTF-8 sequence %s found in string '%s'\n", bp.str, src);
+            invalid = 1;
+        }
+
+        if (invalid) {
+            invalid_chars_nb++;
+
+            switch (wctx->string_validation) {
+            case WRITER_STRING_VALIDATION_FAIL:
+                av_log(wctx, AV_LOG_ERROR,
+                       "Invalid UTF-8 sequence found in string '%s'\n", src);
+                ret = AVERROR_INVALIDDATA;
+                goto end;
+                break;
+
+            case WRITER_STRING_VALIDATION_REPLACE:
+                av_bprintf(&dstbuf, "%s", wctx->string_validation_replacement);
+                break;
+            }
+        }
+
+        if (!invalid || wctx->string_validation == WRITER_STRING_VALIDATION_IGNORE)
+            av_bprint_append_data(&dstbuf, p0, p-p0);
+    }
+
+    if (invalid_chars_nb && wctx->string_validation == WRITER_STRING_VALIDATION_REPLACE) {
+        av_log(wctx, AV_LOG_WARNING,
+               "%d invalid UTF-8 sequence(s) found in string '%s', replaced with '%s'\n",
+               invalid_chars_nb, src, wctx->string_validation_replacement);
+    }
+
+end:
+    av_bprint_finalize(&dstbuf, dstp);
+    return ret;
+}
+
+#define PRINT_STRING_OPT      1
+#define PRINT_STRING_VALIDATE 2
+
+static inline int writer_print_string(WriterContext *wctx,
+                                      const char *key, const char *val, int flags)
 {
     const struct section *section = wctx->section[wctx->level];
+    int ret = 0;
 
-    if (opt && !(wctx->writer->flags & WRITER_FLAG_DISPLAY_OPTIONAL_FIELDS))
-        return;
+    if ((flags & PRINT_STRING_OPT)
+        && !(wctx->writer->flags & WRITER_FLAG_DISPLAY_OPTIONAL_FIELDS))
+        return 0;
 
     if (section->show_all_entries || av_dict_get(section->entries_to_show, key, NULL, 0)) {
-        wctx->writer->print_string(wctx, key, val);
+        if (flags & PRINT_STRING_VALIDATE) {
+            char *key1 = NULL, *val1 = NULL;
+            ret = validate_string(wctx, &key1, key);
+            if (ret < 0) goto end;
+            ret = validate_string(wctx, &val1, val);
+            if (ret < 0) goto end;
+            wctx->writer->print_string(wctx, key1, val1);
+        end:
+            if (ret < 0) {
+                av_log(wctx, AV_LOG_ERROR,
+                       "Invalid key=value string combination %s=%s in section %s\n",
+                       key, val, section->unique_name);
+            }
+            av_free(key1);
+            av_free(val1);
+        } else {
+            wctx->writer->print_string(wctx, key, val);
+        }
+
         wctx->nb_item[wctx->level]++;
     }
+
+    return ret;
 }
 
 static inline void writer_print_rational(WriterContext *wctx,
@@ -457,7 +633,7 @@ static void writer_print_time(WriterContext *wctx, const char *key,
     char buf[128];
 
     if ((!is_duration && ts == AV_NOPTS_VALUE) || (is_duration && ts == 0)) {
-        writer_print_string(wctx, key, "N/A", 1);
+        writer_print_string(wctx, key, "N/A", PRINT_STRING_OPT);
     } else {
         double d = ts * av_q2d(*time_base);
         struct unit_value uv;
@@ -471,7 +647,7 @@ static void writer_print_time(WriterContext *wctx, const char *key,
 static void writer_print_ts(WriterContext *wctx, const char *key, int64_t ts, int is_duration)
 {
     if ((!is_duration && ts == AV_NOPTS_VALUE) || (is_duration && ts == 0)) {
-        writer_print_string(wctx, key, "N/A", 1);
+        writer_print_string(wctx, key, "N/A", PRINT_STRING_OPT);
     } else {
         writer_print_integer(wctx, key, ts);
     }
@@ -540,9 +716,9 @@ static const char *name##_get_name(void *ctx)       \
     return #name ;                                  \
 }                                                   \
 static const AVClass name##_class = {               \
-    #name,                                          \
-    name##_get_name,                                \
-    name##_options                                  \
+    .class_name = #name,                            \
+    .item_name  = name##_get_name,                  \
+    .option     = name##_options                    \
 }
 
 /* Default output */
@@ -554,6 +730,7 @@ typedef struct DefaultContext {
     int nested_section[SECTION_MAX_NB_LEVELS];
 } DefaultContext;
 
+#undef OFFSET
 #define OFFSET(x) offsetof(DefaultContext, x)
 
 static const AVOption default_options[] = {
@@ -1440,7 +1617,8 @@ static void writer_register_all(void)
 #define print_int(k, v)         writer_print_integer(w, k, v)
 #define print_q(k, v, s)        writer_print_rational(w, k, v, s)
 #define print_str(k, v)         writer_print_string(w, k, v, 0)
-#define print_str_opt(k, v)     writer_print_string(w, k, v, 1)
+#define print_str_opt(k, v)     writer_print_string(w, k, v, PRINT_STRING_OPT)
+#define print_str_validate(k, v) writer_print_string(w, k, v, PRINT_STRING_VALIDATE)
 #define print_time(k, v, tb)    writer_print_time(w, k, v, tb, 0)
 #define print_ts(k, v)          writer_print_ts(w, k, v, 0)
 #define print_duration_time(k, v, tb) writer_print_time(w, k, v, tb, 1)
@@ -1455,16 +1633,30 @@ static void writer_register_all(void)
 #define print_section_header(s) writer_print_section_header(w, s)
 #define print_section_footer(s) writer_print_section_footer(w, s)
 
-static inline void show_tags(WriterContext *wctx, AVDictionary *tags, int section_id)
+#define REALLOCZ_ARRAY_STREAM(ptr, cur_n, new_n)                        \
+{                                                                       \
+    ret = av_reallocp_array(&(ptr), (new_n), sizeof(*(ptr)));           \
+    if (ret < 0)                                                        \
+        goto end;                                                       \
+    memset( (ptr) + (cur_n), 0, ((new_n) - (cur_n)) * sizeof(*(ptr)) ); \
+}
+
+static inline int show_tags(WriterContext *w, AVDictionary *tags, int section_id)
 {
     AVDictionaryEntry *tag = NULL;
+    int ret = 0;
 
     if (!tags)
-        return;
-    writer_print_section_header(wctx, section_id);
-    while ((tag = av_dict_get(tags, "", tag, AV_DICT_IGNORE_SUFFIX)))
-        writer_print_string(wctx, tag->key, tag->value, 0);
-    writer_print_section_footer(wctx);
+        return 0;
+    writer_print_section_header(w, section_id);
+
+    while ((tag = av_dict_get(tags, "", tag, AV_DICT_IGNORE_SUFFIX))) {
+        if ((ret = print_str_validate(tag->key, tag->value)) < 0)
+            break;
+    }
+    writer_print_section_footer(w);
+
+    return ret;
 }
 
 static void show_packet(WriterContext *w, AVFormatContext *fmt_ctx, AVPacket *pkt, int packet_idx)
@@ -1502,6 +1694,29 @@ static void show_packet(WriterContext *w, AVFormatContext *fmt_ctx, AVPacket *pk
     fflush(stdout);
 }
 
+static void show_subtitle(WriterContext *w, AVSubtitle *sub, AVStream *stream,
+                          AVFormatContext *fmt_ctx)
+{
+    AVBPrint pbuf;
+
+    av_bprint_init(&pbuf, 1, AV_BPRINT_SIZE_UNLIMITED);
+
+    writer_print_section_header(w, SECTION_ID_SUBTITLE);
+
+    print_str ("media_type",         "subtitle");
+    print_ts  ("pts",                 sub->pts);
+    print_time("pts_time",            sub->pts, &AV_TIME_BASE_Q);
+    print_int ("format",              sub->format);
+    print_int ("start_display_time",  sub->start_display_time);
+    print_int ("end_display_time",    sub->end_display_time);
+    print_int ("num_rects",           sub->num_rects);
+
+    writer_print_section_footer(w);
+
+    av_bprint_finalize(&pbuf, NULL);
+    fflush(stdout);
+}
+
 static void show_frame(WriterContext *w, AVFrame *frame, AVStream *stream,
                        AVFormatContext *fmt_ctx)
 {
@@ -1520,6 +1735,8 @@ static void show_frame(WriterContext *w, AVFrame *frame, AVStream *stream,
     print_time("pkt_pts_time",          frame->pkt_pts, &stream->time_base);
     print_ts  ("pkt_dts",               frame->pkt_dts);
     print_time("pkt_dts_time",          frame->pkt_dts, &stream->time_base);
+    print_ts  ("best_effort_timestamp", av_frame_get_best_effort_timestamp(frame));
+    print_time("best_effort_timestamp_time", av_frame_get_best_effort_timestamp(frame), &stream->time_base);
     print_duration_ts  ("pkt_duration",      av_frame_get_pkt_duration(frame));
     print_duration_time("pkt_duration_time", av_frame_get_pkt_duration(frame), &stream->time_base);
     if (av_frame_get_pkt_pos (frame) != -1) print_fmt    ("pkt_pos", "%"PRId64, av_frame_get_pkt_pos(frame));
@@ -1565,7 +1782,8 @@ static void show_frame(WriterContext *w, AVFrame *frame, AVStream *stream,
             print_str_opt("channel_layout", "unknown");
         break;
     }
-    show_tags(w, av_frame_get_metadata(frame), SECTION_ID_FRAME_TAGS);
+    if (do_show_frame_tags)
+        show_tags(w, av_frame_get_metadata(frame), SECTION_ID_FRAME_TAGS);
 
     writer_print_section_footer(w);
 
@@ -1578,9 +1796,9 @@ static av_always_inline int process_frame(WriterContext *w,
                                           AVFrame *frame, AVPacket *pkt)
 {
     AVCodecContext *dec_ctx = fmt_ctx->streams[pkt->stream_index]->codec;
+    AVSubtitle sub;
     int ret = 0, got_frame = 0;
 
-    avcodec_get_frame_defaults(frame);
     if (dec_ctx->codec) {
         switch (dec_ctx->codec_type) {
         case AVMEDIA_TYPE_VIDEO:
@@ -1590,6 +1808,10 @@ static av_always_inline int process_frame(WriterContext *w,
         case AVMEDIA_TYPE_AUDIO:
             ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, pkt);
             break;
+
+        case AVMEDIA_TYPE_SUBTITLE:
+            ret = avcodec_decode_subtitle2(dec_ctx, &sub, &got_frame, pkt);
+            break;
         }
     }
 
@@ -1599,9 +1821,15 @@ static av_always_inline int process_frame(WriterContext *w,
     pkt->data += ret;
     pkt->size -= ret;
     if (got_frame) {
+        int is_sub = (dec_ctx->codec_type == AVMEDIA_TYPE_SUBTITLE);
         nb_streams_frames[pkt->stream_index]++;
         if (do_show_frames)
-            show_frame(w, frame, fmt_ctx->streams[pkt->stream_index], fmt_ctx);
+            if (is_sub)
+                show_subtitle(w, &sub, fmt_ctx->streams[pkt->stream_index], fmt_ctx);
+            else
+                show_frame(w, frame, fmt_ctx->streams[pkt->stream_index], fmt_ctx);
+        if (is_sub)
+            avsubtitle_free(&sub);
     }
     return got_frame;
 }
@@ -1634,7 +1862,7 @@ static int read_interval_packets(WriterContext *w, AVFormatContext *fmt_ctx,
                                  const ReadInterval *interval, int64_t *cur_ts)
 {
     AVPacket pkt, pkt1;
-    AVFrame frame;
+    AVFrame *frame = NULL;
     int ret = 0, i = 0, frame_count = 0;
     int64_t start = -INT64_MAX, end = interval->end;
     int has_start = 0, has_end = interval->has_end && !interval->end_is_offset;
@@ -1668,7 +1896,18 @@ static int read_interval_packets(WriterContext *w, AVFormatContext *fmt_ctx,
         }
     }
 
+    frame = av_frame_alloc();
+    if (!frame) {
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
     while (!av_read_frame(fmt_ctx, &pkt)) {
+        if (fmt_ctx->nb_streams > nb_streams) {
+            REALLOCZ_ARRAY_STREAM(nb_streams_frames,  nb_streams, fmt_ctx->nb_streams);
+            REALLOCZ_ARRAY_STREAM(nb_streams_packets, nb_streams, fmt_ctx->nb_streams);
+            REALLOCZ_ARRAY_STREAM(selected_streams,   nb_streams, fmt_ctx->nb_streams);
+            nb_streams = fmt_ctx->nb_streams;
+        }
         if (selected_streams[pkt.stream_index]) {
             AVRational tb = fmt_ctx->streams[pkt.stream_index]->time_base;
 
@@ -1700,7 +1939,7 @@ static int read_interval_packets(WriterContext *w, AVFormatContext *fmt_ctx,
             }
             if (do_read_frames) {
                 pkt1 = pkt;
-                while (pkt1.size && process_frame(w, fmt_ctx, &frame, &pkt1) > 0);
+                while (pkt1.size && process_frame(w, fmt_ctx, frame, &pkt1) > 0);
             }
         }
         av_free_packet(&pkt);
@@ -1712,10 +1951,11 @@ static int read_interval_packets(WriterContext *w, AVFormatContext *fmt_ctx,
     for (i = 0; i < fmt_ctx->nb_streams; i++) {
         pkt.stream_index = i;
         if (do_read_frames)
-            while (process_frame(w, fmt_ctx, &frame, &pkt) > 0);
+            while (process_frame(w, fmt_ctx, frame, &pkt) > 0);
     }
 
 end:
+    av_frame_free(&frame);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Could not read packets in interval ");
         log_read_interval(interval, NULL, AV_LOG_ERROR);
@@ -1723,7 +1963,7 @@ end:
     return ret;
 }
 
-static void read_packets(WriterContext *w, AVFormatContext *fmt_ctx)
+static int read_packets(WriterContext *w, AVFormatContext *fmt_ctx)
 {
     int i, ret = 0;
     int64_t cur_ts = fmt_ctx->start_time;
@@ -1738,9 +1978,11 @@ static void read_packets(WriterContext *w, AVFormatContext *fmt_ctx)
                 break;
         }
     }
+
+    return ret;
 }
 
-static void show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_idx, int in_program)
+static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_idx, int in_program)
 {
     AVStream *stream = fmt_ctx->streams[stream_idx];
     AVCodecContext *dec_ctx;
@@ -1749,6 +1991,7 @@ static void show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_i
     const char *s;
     AVRational sar, dar;
     AVBPrint pbuf;
+    int ret = 0;
 
     av_bprint_init(&pbuf, 1, AV_BPRINT_SIZE_UNLIMITED);
 
@@ -1903,26 +2146,35 @@ static void show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_i
     writer_print_section_footer(w);
     }
 
-    show_tags(w, stream->metadata, in_program ? SECTION_ID_PROGRAM_STREAM_TAGS : SECTION_ID_STREAM_TAGS);
+    if (do_show_stream_tags)
+        ret = show_tags(w, stream->metadata, in_program ? SECTION_ID_PROGRAM_STREAM_TAGS : SECTION_ID_STREAM_TAGS);
 
     writer_print_section_footer(w);
     av_bprint_finalize(&pbuf, NULL);
     fflush(stdout);
+
+    return ret;
 }
 
-static void show_streams(WriterContext *w, AVFormatContext *fmt_ctx)
+static int show_streams(WriterContext *w, AVFormatContext *fmt_ctx)
 {
-    int i;
+    int i, ret = 0;
+
     writer_print_section_header(w, SECTION_ID_STREAMS);
     for (i = 0; i < fmt_ctx->nb_streams; i++)
-        if (selected_streams[i])
-            show_stream(w, fmt_ctx, i, 0);
+        if (selected_streams[i]) {
+            ret = show_stream(w, fmt_ctx, i, 0);
+            if (ret < 0)
+                break;
+        }
     writer_print_section_footer(w);
+
+    return ret;
 }
 
-static void show_program(WriterContext *w, AVFormatContext *fmt_ctx, AVProgram *program)
+static int show_program(WriterContext *w, AVFormatContext *fmt_ctx, AVProgram *program)
 {
-    int i;
+    int i, ret = 0;
 
     writer_print_section_header(w, SECTION_ID_PROGRAM);
     print_int("program_id", program->id);
@@ -1934,35 +2186,46 @@ static void show_program(WriterContext *w, AVFormatContext *fmt_ctx, AVProgram *
     print_time("start_time", program->start_time, &AV_TIME_BASE_Q);
     print_ts("end_pts", program->end_time);
     print_time("end_time", program->end_time, &AV_TIME_BASE_Q);
-    show_tags(w, program->metadata, SECTION_ID_PROGRAM_TAGS);
+    if (do_show_program_tags)
+        ret = show_tags(w, program->metadata, SECTION_ID_PROGRAM_TAGS);
+    if (ret < 0)
+        goto end;
 
     writer_print_section_header(w, SECTION_ID_PROGRAM_STREAMS);
     for (i = 0; i < program->nb_stream_indexes; i++) {
-        if (selected_streams[program->stream_index[i]])
-            show_stream(w, fmt_ctx, program->stream_index[i], 1);
+        if (selected_streams[program->stream_index[i]]) {
+            ret = show_stream(w, fmt_ctx, program->stream_index[i], 1);
+            if (ret < 0)
+                break;
+        }
     }
     writer_print_section_footer(w);
 
+end:
     writer_print_section_footer(w);
+    return ret;
 }
 
-static void show_programs(WriterContext *w, AVFormatContext *fmt_ctx)
+static int show_programs(WriterContext *w, AVFormatContext *fmt_ctx)
 {
-    int i;
+    int i, ret = 0;
 
     writer_print_section_header(w, SECTION_ID_PROGRAMS);
     for (i = 0; i < fmt_ctx->nb_programs; i++) {
         AVProgram *program = fmt_ctx->programs[i];
         if (!program)
             continue;
-        show_program(w, fmt_ctx, program);
+        ret = show_program(w, fmt_ctx, program);
+        if (ret < 0)
+            break;
     }
     writer_print_section_footer(w);
+    return ret;
 }
 
-static void show_chapters(WriterContext *w, AVFormatContext *fmt_ctx)
+static int show_chapters(WriterContext *w, AVFormatContext *fmt_ctx)
 {
-    int i;
+    int i, ret = 0;
 
     writer_print_section_header(w, SECTION_ID_CHAPTERS);
     for (i = 0; i < fmt_ctx->nb_chapters; i++) {
@@ -1975,19 +2238,23 @@ static void show_chapters(WriterContext *w, AVFormatContext *fmt_ctx)
         print_time("start_time", chapter->start, &chapter->time_base);
         print_int("end", chapter->end);
         print_time("end_time", chapter->end, &chapter->time_base);
-        show_tags(w, chapter->metadata, SECTION_ID_CHAPTER_TAGS);
+        if (do_show_chapter_tags)
+            ret = show_tags(w, chapter->metadata, SECTION_ID_CHAPTER_TAGS);
         writer_print_section_footer(w);
     }
     writer_print_section_footer(w);
+
+    return ret;
 }
 
-static void show_format(WriterContext *w, AVFormatContext *fmt_ctx)
+static int show_format(WriterContext *w, AVFormatContext *fmt_ctx)
 {
     char val_str[128];
     int64_t size = fmt_ctx->pb ? avio_size(fmt_ctx->pb) : -1;
+    int ret = 0;
 
     writer_print_section_header(w, SECTION_ID_FORMAT);
-    print_str("filename",         fmt_ctx->filename);
+    print_str_validate("filename", fmt_ctx->filename);
     print_int("nb_streams",       fmt_ctx->nb_streams);
     print_int("nb_programs",      fmt_ctx->nb_programs);
     print_str("format_name",      fmt_ctx->iformat->name);
@@ -2002,10 +2269,12 @@ static void show_format(WriterContext *w, AVFormatContext *fmt_ctx)
     if (fmt_ctx->bit_rate > 0) print_val    ("bit_rate", fmt_ctx->bit_rate, unit_bit_per_second_str);
     else                       print_str_opt("bit_rate", "N/A");
     print_int("probe_score", av_format_get_probe_score(fmt_ctx));
-    show_tags(w, fmt_ctx->metadata, SECTION_ID_FORMAT_TAGS);
+    if (do_show_format_tags)
+        ret = show_tags(w, fmt_ctx->metadata, SECTION_ID_FORMAT_TAGS);
 
     writer_print_section_footer(w);
     fflush(stdout);
+    return ret;
 }
 
 static void show_error(WriterContext *w, int err)
@@ -2111,17 +2380,19 @@ static int probe_file(WriterContext *wctx, const char *filename)
     if (ret < 0)
         return ret;
 
-    nb_streams_frames  = av_calloc(fmt_ctx->nb_streams, sizeof(*nb_streams_frames));
-    nb_streams_packets = av_calloc(fmt_ctx->nb_streams, sizeof(*nb_streams_packets));
-    selected_streams   = av_calloc(fmt_ctx->nb_streams, sizeof(*selected_streams));
+#define CHECK_END if (ret < 0) goto end
+
+    nb_streams = fmt_ctx->nb_streams;
+    REALLOCZ_ARRAY_STREAM(nb_streams_frames,0,fmt_ctx->nb_streams);
+    REALLOCZ_ARRAY_STREAM(nb_streams_packets,0,fmt_ctx->nb_streams);
+    REALLOCZ_ARRAY_STREAM(selected_streams,0,fmt_ctx->nb_streams);
 
     for (i = 0; i < fmt_ctx->nb_streams; i++) {
         if (stream_specifier) {
             ret = avformat_match_stream_specifier(fmt_ctx,
                                                   fmt_ctx->streams[i],
                                                   stream_specifier);
-            if (ret < 0)
-                goto end;
+            CHECK_END;
             else
                 selected_streams[i] = ret;
             ret = 0;
@@ -2140,18 +2411,29 @@ static int probe_file(WriterContext *wctx, const char *filename)
             section_id = SECTION_ID_FRAMES;
         if (do_show_frames || do_show_packets)
             writer_print_section_header(wctx, section_id);
-        read_packets(wctx, fmt_ctx);
+        ret = read_packets(wctx, fmt_ctx);
         if (do_show_frames || do_show_packets)
             writer_print_section_footer(wctx);
+        CHECK_END;
+    }
+
+    if (do_show_programs) {
+        ret = show_programs(wctx, fmt_ctx);
+        CHECK_END;
+    }
+
+    if (do_show_streams) {
+        ret = show_streams(wctx, fmt_ctx);
+        CHECK_END;
+    }
+    if (do_show_chapters) {
+        ret = show_chapters(wctx, fmt_ctx);
+        CHECK_END;
+    }
+    if (do_show_format) {
+        ret = show_format(wctx, fmt_ctx);
+        CHECK_END;
     }
-    if (do_show_programs)
-        show_programs(wctx, fmt_ctx);
-    if (do_show_streams)
-        show_streams(wctx, fmt_ctx);
-    if (do_show_chapters)
-        show_chapters(wctx, fmt_ctx);
-    if (do_show_format)
-        show_format(wctx, fmt_ctx);
 
 end:
     close_input_file(&fmt_ctx);
@@ -2177,7 +2459,7 @@ static void ffprobe_show_program_version(WriterContext *w)
     writer_print_section_header(w, SECTION_ID_PROGRAM_VERSION);
     print_str("version", FFMPEG_VERSION);
     print_fmt("copyright", "Copyright (c) %d-%d the FFmpeg developers",
-              program_birth_year, this_year);
+              program_birth_year, CONFIG_THIS_YEAR);
     print_str("build_date", __DATE__);
     print_str("build_time", __TIME__);
     print_str("compiler_ident", CC_IDENT);
@@ -2648,6 +2930,12 @@ int main(int argc, char **argv)
     SET_DO_SHOW(STREAM_DISPOSITION, stream_disposition);
     SET_DO_SHOW(PROGRAM_STREAM_DISPOSITION, stream_disposition);
 
+    SET_DO_SHOW(CHAPTER_TAGS, chapter_tags);
+    SET_DO_SHOW(FORMAT_TAGS, format_tags);
+    SET_DO_SHOW(FRAME_TAGS, frame_tags);
+    SET_DO_SHOW(PROGRAM_TAGS, program_tags);
+    SET_DO_SHOW(STREAM_TAGS, stream_tags);
+
     if (do_bitexact && (do_show_program_version || do_show_library_versions)) {
         av_log(NULL, AV_LOG_ERROR,
                "-bitexact and -show_program_version or -show_library_versions "
@@ -2676,6 +2964,9 @@ int main(int argc, char **argv)
 
     if ((ret = writer_open(&wctx, w, w_args,
                            sections, FF_ARRAY_ELEMS(sections))) >= 0) {
+        if (w == &xml_writer)
+            wctx->string_validation_utf8_flags |= AV_UTF8_FLAG_EXCLUDE_XML_INVALID_CONTROL_CODES;
+
         writer_print_section_header(wctx, SECTION_ID_ROOT);
 
         if (do_show_program_version)

ffserver.c

@@ -216,6 +216,7 @@ typedef struct FFStream {
     struct FFStream *feed;   /* feed we are using (can be null if
                                 coming from file) */
     AVDictionary *in_opts;   /* input parameters */
+    AVDictionary *metadata;  /* metadata to set on the stream */
     AVInputFormat *ifmt;       /* if non NULL, force input format */
     AVOutputFormat *fmt;
     IPAddressACL *acl;
@@ -228,10 +229,6 @@ typedef struct FFStream {
     int feed_streams[MAX_STREAMS]; /* index of streams in the feed */
     char feed_filename[1024]; /* file name of the feed storage, or
                                  input file name for a stream */
-    char author[512];
-    char title[512];
-    char copyright[512];
-    char comment[512];
     pid_t pid;  /* Of ffmpeg process */
     time_t pid_start;  /* Of ffmpeg process */
     char **child_argv;
@@ -290,8 +287,7 @@ static void rtsp_cmd_describe(HTTPContext *c, const char *url);
 static void rtsp_cmd_options(HTTPContext *c, const char *url);
 static void rtsp_cmd_setup(HTTPContext *c, const char *url, RTSPMessageHeader *h);
 static void rtsp_cmd_play(HTTPContext *c, const char *url, RTSPMessageHeader *h);
-static void rtsp_cmd_pause(HTTPContext *c, const char *url, RTSPMessageHeader *h);
-static void rtsp_cmd_teardown(HTTPContext *c, const char *url, RTSPMessageHeader *h);
+static void rtsp_cmd_interrupt(HTTPContext *c, const char *url, RTSPMessageHeader *h, int pause_only);
 
 /* SDP handling */
 static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer,
@@ -509,8 +505,9 @@ static void start_children(FFStream *feed)
                 char *slash;
                 int i;
 
+                /* replace "ffserver" with "ffmpeg" in the path of current
+                 * program. Ignore user provided path */
                 av_strlcpy(pathname, my_program_name, sizeof(pathname));
-
                 slash = strrchr(pathname, '/');
                 if (!slash)
                     slash = pathname;
@@ -630,17 +627,16 @@ static void start_multicast(void)
                 }
             }
 
-            /* change state to send data */
             rtp_c->state = HTTPSTATE_SEND_DATA;
         }
     }
 }
 
-/* main loop of the http server */
+/* main loop of the HTTP server */
 static int http_server(void)
 {
     int server_fd = 0, rtsp_server_fd = 0;
-    int ret, delay, delay1;
+    int ret, delay;
     struct pollfd *poll_table, *poll_entry;
     HTTPContext *c, *c_next;
 
@@ -704,18 +700,19 @@ static int http_server(void)
             case HTTPSTATE_SEND_DATA:
             case HTTPSTATE_SEND_DATA_TRAILER:
                 if (!c->is_packetized) {
-                    /* for TCP, we output as much as we can (may need to put a limit) */
+                    /* for TCP, we output as much as we can
+                     * (may need to put a limit) */
                     c->poll_entry = poll_entry;
                     poll_entry->fd = fd;
                     poll_entry->events = POLLOUT;
                     poll_entry++;
                 } else {
                     /* when ffserver is doing the timing, we work by
-                       looking at which packet need to be sent every
+                       looking at which packet needs to be sent every
                        10 ms */
-                    delay1 = 10; /* one tick wait XXX: 10 ms assumed */
-                    if (delay1 < delay)
-                        delay = delay1;
+                    /* one tick wait XXX: 10 ms assumed */
+                    if (delay > 10)
+                        delay = 10;
                 }
                 break;
             case HTTPSTATE_WAIT_REQUEST:
@@ -755,8 +752,8 @@ static int http_server(void)
         for(c = first_http_ctx; c != NULL; c = c_next) {
             c_next = c->next;
             if (handle_connection(c) < 0) {
-                /* close and free the connection */
                 log_connection(c);
+                /* close and free the connection */
                 close_connection(c);
             }
         }
@@ -1005,9 +1002,7 @@ static int handle_connection(HTTPContext *c)
         if (len < 0) {
             if (ff_neterrno() != AVERROR(EAGAIN) &&
                 ff_neterrno() != AVERROR(EINTR)) {
-                /* error : close connection */
-                av_freep(&c->pb_buffer);
-                return -1;
+                goto close_connection;
             }
         } else {
             c->buffer_ptr += len;
@@ -1019,7 +1014,8 @@ static int handle_connection(HTTPContext *c)
                 /* if error, exit */
                 if (c->http_error)
                     return -1;
-                /* all the buffer was sent : synchronize to the incoming stream */
+                /* all the buffer was sent : synchronize to the incoming
+                 * stream */
                 c->state = HTTPSTATE_SEND_DATA_HEADER;
                 c->buffer_ptr = c->buffer_end = c->buffer;
             }
@@ -1030,7 +1026,7 @@ static int handle_connection(HTTPContext *c)
     case HTTPSTATE_SEND_DATA_HEADER:
     case HTTPSTATE_SEND_DATA_TRAILER:
         /* for packetized output, we consider we can always write (the
-           input streams sets the speed). It may be better to verify
+           input streams set the speed). It may be better to verify
            that we do not rely too much on the kernel queues */
         if (!c->is_packetized) {
             if (c->poll_entry->revents & (POLLERR | POLLHUP))
@@ -1064,10 +1060,8 @@ static int handle_connection(HTTPContext *c)
         break;
 
     case RTSPSTATE_SEND_REPLY:
-        if (c->poll_entry->revents & (POLLERR | POLLHUP)) {
-            av_freep(&c->pb_buffer);
-            return -1;
-        }
+        if (c->poll_entry->revents & (POLLERR | POLLHUP))
+            goto close_connection;
         /* no need to write if no events */
         if (!(c->poll_entry->revents & POLLOUT))
             return 0;
@@ -1075,9 +1069,7 @@ static int handle_connection(HTTPContext *c)
         if (len < 0) {
             if (ff_neterrno() != AVERROR(EAGAIN) &&
                 ff_neterrno() != AVERROR(EINTR)) {
-                /* error : close connection */
-                av_freep(&c->pb_buffer);
-                return -1;
+                goto close_connection;
             }
         } else {
             c->buffer_ptr += len;
@@ -1122,6 +1114,10 @@ static int handle_connection(HTTPContext *c)
         return -1;
     }
     return 0;
+
+close_connection:
+    av_freep(&c->pb_buffer);
+    return -1;
 }
 
 static int extract_rates(char *rates, int ratelen, const char *request)
@@ -1460,7 +1456,7 @@ static int validate_acl(FFStream *stream, HTTPContext *c)
 }
 
 /* compute the real filename of a file by matching it without its
-   extensions to all the stream filenames */
+   extensions to all the stream's filenames */
 static void compute_real_filename(char *filename, int max_size)
 {
     char file1[1024];
@@ -1494,7 +1490,7 @@ enum RedirType {
     REDIR_SDP,
 };
 
-/* parse http request and prepare header */
+/* parse HTTP request and prepare header */
 static int http_parse_request(HTTPContext *c)
 {
     const char *p;
@@ -1865,7 +1861,7 @@ static int http_parse_request(HTTPContext *c)
         goto send_error;
     }
 
-    /* prepare http header */
+    /* prepare HTTP header */
     c->buffer[0] = 0;
     av_strlcatf(c->buffer, c->buffer_size, "HTTP/1.0 200 OK\r\n");
     mime_type = c->stream->fmt->mime_type;
@@ -1944,7 +1940,7 @@ static void compute_status(HTTPContext *c)
     }
 
     avio_printf(pb, "HTTP/1.0 200 OK\r\n");
-    avio_printf(pb, "Content-type: %s\r\n", "text/html");
+    avio_printf(pb, "Content-type: text/html\r\n");
     avio_printf(pb, "Pragma: no-cache\r\n");
     avio_printf(pb, "\r\n");
 
@@ -2183,8 +2179,10 @@ static int open_input_stream(HTTPContext *c, const char *info)
         buf_size = FFM_PACKET_SIZE;
         /* compute position (absolute time) */
         if (av_find_info_tag(buf, sizeof(buf), "date", info)) {
-            if ((ret = av_parse_time(&stream_pos, buf, 0)) < 0)
+            if ((ret = av_parse_time(&stream_pos, buf, 0)) < 0) {
+                http_log("Invalid date specification '%s' for stream\n", buf);
                 return ret;
+            }
         } else if (av_find_info_tag(buf, sizeof(buf), "buffer", info)) {
             int prebuffer = strtol(buf, 0, 10);
             stream_pos = av_gettime() - prebuffer * (int64_t)1000000;
@@ -2195,18 +2193,22 @@ static int open_input_stream(HTTPContext *c, const char *info)
         buf_size = 0;
         /* compute position (relative time) */
         if (av_find_info_tag(buf, sizeof(buf), "date", info)) {
-            if ((ret = av_parse_time(&stream_pos, buf, 1)) < 0)
+            if ((ret = av_parse_time(&stream_pos, buf, 1)) < 0) {
+                http_log("Invalid date specification '%s' for stream\n", buf);
                 return ret;
+            }
         } else
             stream_pos = 0;
     }
-    if (input_filename[0] == '\0')
-        return -1;
+    if (!input_filename[0]) {
+        http_log("No filename was specified for stream\n");
+        return AVERROR(EINVAL);
+    }
 
     /* open stream */
     if ((ret = avformat_open_input(&s, input_filename, c->stream->ifmt, &c->stream->in_opts)) < 0) {
-        http_log("could not open %s: %d\n", input_filename, ret);
-        return -1;
+        http_log("Could not open input '%s': %s\n", input_filename, av_err2str(ret));
+        return ret;
     }
 
     /* set buffer size */
@@ -2214,14 +2216,15 @@ static int open_input_stream(HTTPContext *c, const char *info)
 
     s->flags |= AVFMT_FLAG_GENPTS;
     c->fmt_in = s;
-    if (strcmp(s->iformat->name, "ffm") && avformat_find_stream_info(c->fmt_in, NULL) < 0) {
-        http_log("Could not find stream info '%s'\n", input_filename);
+    if (strcmp(s->iformat->name, "ffm") &&
+        (ret = avformat_find_stream_info(c->fmt_in, NULL)) < 0) {
+        http_log("Could not find stream info for input '%s'\n", input_filename);
         avformat_close_input(&s);
-        return -1;
+        return ret;
     }
 
-    /* choose stream as clock source (we favorize video stream if
-       present) for packet sending */
+    /* choose stream as clock source (we favor the video stream if
+     * present) for packet sending */
     c->pts_stream_index = 0;
     for(i=0;i<c->stream->nb_streams;i++) {
         if (c->pts_stream_index == 0 &&
@@ -2270,12 +2273,10 @@ static int http_prepare_data(HTTPContext *c)
     av_freep(&c->pb_buffer);
     switch(c->state) {
     case HTTPSTATE_SEND_DATA_HEADER:
-        memset(&c->fmt_ctx, 0, sizeof(c->fmt_ctx));
-        av_dict_set(&c->fmt_ctx.metadata, "author"   , c->stream->author   , 0);
-        av_dict_set(&c->fmt_ctx.metadata, "comment"  , c->stream->comment  , 0);
-        av_dict_set(&c->fmt_ctx.metadata, "copyright", c->stream->copyright, 0);
-        av_dict_set(&c->fmt_ctx.metadata, "title"    , c->stream->title    , 0);
-
+        ctx = avformat_alloc_context();
+        c->fmt_ctx = *ctx;
+        av_freep(&ctx);
+        av_dict_copy(&(c->fmt_ctx.metadata), c->stream->metadata, 0);
         c->fmt_ctx.streams = av_mallocz(sizeof(AVStream *) * c->stream->nb_streams);
 
         for(i=0;i<c->stream->nb_streams;i++) {
@@ -2290,8 +2291,8 @@ static int http_prepare_data(HTTPContext *c)
 
             *(c->fmt_ctx.streams[i]) = *src;
             c->fmt_ctx.streams[i]->priv_data = 0;
-            c->fmt_ctx.streams[i]->codec->frame_number = 0; /* XXX: should be done in
-                                           AVStream, not in codec */
+            /* XXX: should be done in AVStream, not in codec */
+            c->fmt_ctx.streams[i]->codec->frame_number = 0;
         }
         /* set output format parameters */
         c->fmt_ctx.oformat = c->stream->fmt;
@@ -2309,13 +2310,14 @@ static int http_prepare_data(HTTPContext *c)
         /*
          * HACK to avoid mpeg ps muxer to spit many underflow errors
          * Default value from FFmpeg
-         * Try to set it use configuration option
+         * Try to set it using configuration option
          */
         c->fmt_ctx.max_delay = (int)(0.7*AV_TIME_BASE);
 
-        if (avformat_write_header(&c->fmt_ctx, NULL) < 0) {
-            http_log("Error writing output header\n");
-            return -1;
+        if ((ret = avformat_write_header(&c->fmt_ctx, NULL)) < 0) {
+            http_log("Error writing output header for stream '%s': %s\n",
+                     c->stream->filename, av_err2str(ret));
+            return ret;
         }
         av_dict_free(&c->fmt_ctx.metadata);
 
@@ -2359,7 +2361,7 @@ static int http_prepare_data(HTTPContext *c)
                         goto redo;
                     } else {
                     no_loop:
-                        /* must send trailer now because eof or error */
+                        /* must send trailer now because EOF or error */
                         c->state = HTTPSTATE_SEND_DATA_TRAILER;
                     }
                 }
@@ -2401,8 +2403,8 @@ static int http_prepare_data(HTTPContext *c)
                 send_it:
                     ist = c->fmt_in->streams[source_index];
                     /* specific handling for RTP: we use several
-                       output stream (one for each RTP
-                       connection). XXX: need more abstract handling */
+                     * output streams (one for each RTP connection).
+                     * XXX: need more abstract handling */
                     if (c->is_packetized) {
                         /* compute send time and duration */
                         c->cur_pts = av_rescale_q(pkt.dts, ist->time_base, AV_TIME_BASE_Q);
@@ -2446,8 +2448,9 @@ static int http_prepare_data(HTTPContext *c)
                     if (pkt.pts != AV_NOPTS_VALUE)
                         pkt.pts = av_rescale_q(pkt.pts, ist->time_base, ost->time_base);
                     pkt.duration = av_rescale_q(pkt.duration, ist->time_base, ost->time_base);
-                    if (av_write_frame(ctx, &pkt) < 0) {
-                        http_log("Error writing frame to output\n");
+                    if ((ret = av_write_frame(ctx, &pkt)) < 0) {
+                        http_log("Error writing frame to output for stream '%s': %s\n",
+                                 c->stream->filename, av_err2str(ret));
                         c->state = HTTPSTATE_SEND_DATA_TRAILER;
                     }
 
@@ -2491,7 +2494,7 @@ static int http_prepare_data(HTTPContext *c)
 
 /* should convert the format at the same time */
 /* send data starting at c->buffer_ptr to the output connection
-   (either UDP or TCP connection) */
+ * (either UDP or TCP) */
 static int http_send_data(HTTPContext *c)
 {
     int len, ret;
@@ -2614,19 +2617,26 @@ static int http_send_data(HTTPContext *c)
 static int http_start_receive_data(HTTPContext *c)
 {
     int fd;
+    int ret;
 
-    if (c->stream->feed_opened)
-        return -1;
+    if (c->stream->feed_opened) {
+        http_log("Stream feed '%s' was not opened\n", c->stream->feed_filename);
+        return AVERROR(EINVAL);
+    }
 
     /* Don't permit writing to this one */
-    if (c->stream->readonly)
-        return -1;
+    if (c->stream->readonly) {
+        http_log("Cannot write to read-only file '%s'\n", c->stream->feed_filename);
+        return AVERROR(EINVAL);
+    }
 
     /* open feed */
     fd = open(c->stream->feed_filename, O_RDWR);
     if (fd < 0) {
-        http_log("Error opening feeder file: %s\n", strerror(errno));
-        return -1;
+        ret = AVERROR(errno);
+        http_log("Could not open feed file '%s': %s\n",
+                 c->stream->feed_filename, strerror(errno));
+        return ret;
     }
     c->feed_fd = fd;
 
@@ -2635,13 +2645,19 @@ static int http_start_receive_data(HTTPContext *c)
         ffm_write_write_index(c->feed_fd, FFM_PACKET_SIZE);
         http_log("Truncating feed file '%s'\n", c->stream->feed_filename);
         if (ftruncate(c->feed_fd, FFM_PACKET_SIZE) < 0) {
-            http_log("Error truncating feed file: %s\n", strerror(errno));
-            return -1;
+            ret = AVERROR(errno);
+            http_log("Error truncating feed file '%s': %s\n",
+                     c->stream->feed_filename, strerror(errno));
+            return ret;
         }
     } else {
-        if ((c->stream->feed_write_index = ffm_read_write_index(fd)) < 0) {
-            http_log("Error reading write index from feed file: %s\n", strerror(errno));
-            return -1;
+        ret = ffm_read_write_index(fd);
+        if (ret < 0) {
+            http_log("Error reading write index from feed file '%s': %s\n",
+                     c->stream->feed_filename, strerror(errno));
+            return ret;
+        } else {
+            c->stream->feed_write_index = ret;
         }
     }
 
@@ -2945,9 +2961,9 @@ static int rtsp_parse_request(HTTPContext *c)
     else if (!strcmp(cmd, "PLAY"))
         rtsp_cmd_play(c, url, header);
     else if (!strcmp(cmd, "PAUSE"))
-        rtsp_cmd_pause(c, url, header);
+        rtsp_cmd_interrupt(c, url, header, 1);
     else if (!strcmp(cmd, "TEARDOWN"))
-        rtsp_cmd_teardown(c, url, header);
+        rtsp_cmd_interrupt(c, url, header, 0);
     else
         rtsp_reply_error(c, RTSP_STATUS_METHOD);
 
@@ -2970,6 +2986,7 @@ static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer,
     AVFormatContext *avc;
     AVStream *avs = NULL;
     AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);
+    AVDictionaryEntry *entry = av_dict_get(stream->metadata, "title", NULL, 0);
     int i;
 
     avc =  avformat_alloc_context();
@@ -2978,7 +2995,7 @@ static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer,
     }
     avc->oformat = rtp_format;
     av_dict_set(&avc->metadata, "title",
-               stream->title[0] ? stream->title : "No Title", 0);
+                entry ? entry->value : "No Title", 0);
     avc->nb_streams = stream->nb_streams;
     if (stream->is_multicast) {
         snprintf(avc->filename, 1024, "rtp://%s:%d?multicast=1?ttl=%d",
@@ -3300,7 +3317,7 @@ static void rtsp_cmd_play(HTTPContext *c, const char *url, RTSPMessageHeader *h)
     avio_printf(c->pb, "\r\n");
 }
 
-static void rtsp_cmd_pause(HTTPContext *c, const char *url, RTSPMessageHeader *h)
+static void rtsp_cmd_interrupt(HTTPContext *c, const char *url, RTSPMessageHeader *h, int pause_only)
 {
     HTTPContext *rtp_c;
 
@@ -3310,29 +3327,14 @@ static void rtsp_cmd_pause(HTTPContext *c, const char *url, RTSPMessageHeader *h
         return;
     }
 
-    if (rtp_c->state != HTTPSTATE_SEND_DATA &&
-        rtp_c->state != HTTPSTATE_WAIT_FEED) {
-        rtsp_reply_error(c, RTSP_STATUS_STATE);
-        return;
-    }
-
-    rtp_c->state = HTTPSTATE_READY;
-    rtp_c->first_pts = AV_NOPTS_VALUE;
-    /* now everything is OK, so we can send the connection parameters */
-    rtsp_reply_header(c, RTSP_STATUS_OK);
-    /* session ID */
-    avio_printf(c->pb, "Session: %s\r\n", rtp_c->session_id);
-    avio_printf(c->pb, "\r\n");
-}
-
-static void rtsp_cmd_teardown(HTTPContext *c, const char *url, RTSPMessageHeader *h)
-{
-    HTTPContext *rtp_c;
-
-    rtp_c = find_rtp_session_with_url(url, h->session_id);
-    if (!rtp_c) {
-        rtsp_reply_error(c, RTSP_STATUS_SESSION);
-        return;
+    if (pause_only) {
+        if (rtp_c->state != HTTPSTATE_SEND_DATA &&
+            rtp_c->state != HTTPSTATE_WAIT_FEED) {
+            rtsp_reply_error(c, RTSP_STATUS_STATE);
+            return;
+        }
+        rtp_c->state = HTTPSTATE_READY;
+        rtp_c->first_pts = AV_NOPTS_VALUE;
     }
 
     /* now everything is OK, so we can send the connection parameters */
@@ -3341,11 +3343,10 @@ static void rtsp_cmd_teardown(HTTPContext *c, const char *url, RTSPMessageHeader
     avio_printf(c->pb, "Session: %s\r\n", rtp_c->session_id);
     avio_printf(c->pb, "\r\n");
 
-    /* abort the session */
-    close_connection(rtp_c);
+    if (!pause_only)
+        close_connection(rtp_c);
 }
 
-
 /********************************************************************/
 /* RTP handling */
 
@@ -3490,7 +3491,8 @@ static int rtp_new_av_stream(HTTPContext *c,
              ipaddr, ntohs(dest_addr->sin_port),
              c->stream->filename, stream_index, c->protocol);
 
-    /* normally, no packets should be output here, but the packet size may be checked */
+    /* normally, no packets should be output here, but the packet size may
+     * be checked */
     if (ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size) < 0) {
         /* XXX: close stream */
         goto fail;
@@ -3532,7 +3534,7 @@ static AVStream *add_av_stream1(FFStream *stream, AVCodecContext *codec, int cop
         }
     } else {
         /* live streams must use the actual feed's codec since it may be
-         * updated later to carry extradata needed by the streams.
+         * updated later to carry extradata needed by them.
          */
         fst->codec = codec;
     }
@@ -3669,9 +3671,14 @@ static void build_file_streams(void)
                 av_dict_set(&stream->in_opts, "mpeg2ts_compute_pcr", "1", 0);
             }
 
-            http_log("Opening file '%s'\n", stream->feed_filename);
+            if (!stream->feed_filename[0]) {
+                http_log("Unspecified feed file for stream '%s'\n", stream->filename);
+                goto fail;
+            }
+
+            http_log("Opening feed file '%s' for stream '%s'\n", stream->feed_filename, stream->filename);
             if ((ret = avformat_open_input(&infile, stream->feed_filename, stream->ifmt, &stream->in_opts)) < 0) {
-                http_log("Could not open '%s': %d\n", stream->feed_filename, ret);
+                http_log("Could not open '%s': %s\n", stream->feed_filename, av_err2str(ret));
                 /* remove stream (no need to spend more time on it) */
             fail:
                 remove_stream(stream);
@@ -3796,7 +3803,7 @@ static void build_feed_streams(void)
             }
         }
         if (avio_check(feed->feed_filename, AVIO_FLAG_WRITE) <= 0) {
-            AVFormatContext s1 = {0}, *s = &s1;
+            AVFormatContext *s = avformat_alloc_context();
 
             if (feed->readonly) {
                 http_log("Unable to create feed file '%s' as it is marked readonly\n",
@@ -3820,6 +3827,9 @@ static void build_feed_streams(void)
             /* XXX: need better api */
             av_freep(&s->priv_data);
             avio_close(s->pb);
+            s->streams = NULL;
+            s->nb_streams = 0;
+            avformat_free_context(s);
         }
         /* get feed size and write index */
         fd = open(feed->feed_filename, O_RDONLY);
@@ -3942,24 +3952,13 @@ static void add_codec(FFStream *stream, AVCodecContext *av)
     memcpy(st->codec, av, sizeof(AVCodecContext));
 }
 
-static enum AVCodecID opt_audio_codec(const char *arg)
+static enum AVCodecID opt_codec(const char *name, enum AVMediaType type)
 {
-    AVCodec *p= avcodec_find_encoder_by_name(arg);
+    AVCodec *codec = avcodec_find_encoder_by_name(name);
 
-    if (p == NULL || p->type != AVMEDIA_TYPE_AUDIO)
+    if (!codec || codec->type != type)
         return AV_CODEC_ID_NONE;
-
-    return p->id;
-}
-
-static enum AVCodecID opt_video_codec(const char *arg)
-{
-    AVCodec *p= avcodec_find_encoder_by_name(arg);
-
-    if (p == NULL || p->type != AVMEDIA_TYPE_VIDEO)
-        return AV_CODEC_ID_NONE;
-
-    return p->id;
+    return codec->id;
 }
 
 static int ffserver_opt_default(const char *opt, const char *arg,
@@ -3998,9 +3997,9 @@ static int ffserver_opt_preset(const char *arg,
             break;
         }
         if(!strcmp(tmp, "acodec")){
-            *audio_id = opt_audio_codec(tmp2);
+            *audio_id = opt_codec(tmp2, AVMEDIA_TYPE_AUDIO);
         }else if(!strcmp(tmp, "vcodec")){
-            *video_id = opt_video_codec(tmp2);
+            *video_id = opt_codec(tmp2, AVMEDIA_TYPE_VIDEO);
         }else if(!strcmp(tmp, "scodec")){
             /* opt_subtitle_codec(tmp2); */
         }else if(ffserver_opt_default(tmp, tmp2, avctx, type) < 0){
@@ -4015,8 +4014,7 @@ static int ffserver_opt_preset(const char *arg,
     return ret;
 }
 
-static AVOutputFormat *ffserver_guess_format(const char *short_name, const char *filename,
-                                             const char *mime_type)
+static AVOutputFormat *ffserver_guess_format(const char *short_name, const char *filename, const char *mime_type)
 {
     AVOutputFormat *fmt = av_guess_format(short_name, filename, mime_type);
 
@@ -4034,12 +4032,12 @@ static AVOutputFormat *ffserver_guess_format(const char *short_name, const char
     return fmt;
 }
 
-static void report_config_error(const char *filename, int line_num, int *errors, const char *fmt, ...)
+static void report_config_error(const char *filename, int line_num, int log_level, int *errors, const char *fmt, ...)
 {
     va_list vl;
     va_start(vl, fmt);
-    fprintf(stderr, "%s:%d: ", filename, line_num);
-    vfprintf(stderr, fmt, vl);
+    av_log(NULL, log_level, "%s:%d: ", filename, line_num);
+    av_vlog(NULL, log_level, fmt, vl);
     va_end(vl);
 
     (*errors)++;
@@ -4050,21 +4048,23 @@ static int parse_ffconfig(const char *filename)
     FILE *f;
     char line[1024];
     char cmd[64];
-    char arg[1024];
+    char arg[1024], arg2[1024];
     const char *p;
-    int val, errors, line_num;
+    int val, errors, warnings, line_num;
     FFStream **last_stream, *stream, *redirect;
     FFStream **last_feed, *feed, *s;
     AVCodecContext audio_enc, video_enc;
     enum AVCodecID audio_id, video_id;
+    int ret = 0;
 
     f = fopen(filename, "r");
     if (!f) {
-        perror(filename);
-        return -1;
+        ret = AVERROR(errno);
+        av_log(NULL, AV_LOG_ERROR, "Could not open the configuration file '%s'\n", filename);
+        return ret;
     }
 
-    errors = 0;
+    errors = warnings = 0;
     line_num = 0;
     first_stream = NULL;
     last_stream = &first_stream;
@@ -4075,8 +4075,9 @@ static int parse_ffconfig(const char *filename)
     redirect = NULL;
     audio_id = AV_CODEC_ID_NONE;
     video_id = AV_CODEC_ID_NONE;
+#define ERROR(...)   report_config_error(filename, line_num, AV_LOG_ERROR,   &errors,   __VA_ARGS__)
+#define WARNING(...) report_config_error(filename, line_num, AV_LOG_WARNING, &warnings, __VA_ARGS__)
 
-#define ERROR(...) report_config_error(filename, line_num, &errors, __VA_ARGS__)
     for(;;) {
         if (fgets(line, sizeof(line), f) == NULL)
             break;
@@ -4102,7 +4103,7 @@ static int parse_ffconfig(const char *filename)
                 ERROR("%s:%d: Invalid host/IP address: %s\n", arg);
             }
         } else if (!av_strcasecmp(cmd, "NoDaemon")) {
-            // do nothing here, its the default now
+            WARNING("NoDaemon option has no effect, you should remove it\n");
         } else if (!av_strcasecmp(cmd, "RTSPPort")) {
             get_arg(arg, sizeof(arg), &p);
             val = atoi(arg);
@@ -4149,6 +4150,10 @@ static int parse_ffconfig(const char *filename)
                 ERROR("Already in a tag\n");
             } else {
                 feed = av_mallocz(sizeof(FFStream));
+                if (!feed) {
+                    ret = AVERROR(ENOMEM);
+                    goto end;
+                }
                 get_arg(feed->filename, sizeof(feed->filename), &p);
                 q = strrchr(feed->filename, '>');
                 if (*q)
@@ -4161,7 +4166,7 @@ static int parse_ffconfig(const char *filename)
                 }
 
                 feed->fmt = av_guess_format("ffm", NULL, NULL);
-                /* defaut feed file */
+                /* default feed file */
                 snprintf(feed->feed_filename, sizeof(feed->feed_filename),
                          "/tmp/%s.ffm", feed->filename);
                 feed->feed_max_size = 5 * 1024 * 1024;
@@ -4180,36 +4185,50 @@ static int parse_ffconfig(const char *filename)
                 int i;
 
                 feed->child_argv = av_mallocz(64 * sizeof(char *));
-
+                if (!feed->child_argv) {
+                    ret = AVERROR(ENOMEM);
+                    goto end;
+                }
                 for (i = 0; i < 62; i++) {
                     get_arg(arg, sizeof(arg), &p);
                     if (!arg[0])
                         break;
 
                     feed->child_argv[i] = av_strdup(arg);
+                    if (!feed->child_argv[i]) {
+                        ret = AVERROR(ENOMEM);
+                        goto end;
+                    }
                 }
 
-                feed->child_argv[i] = av_asprintf("http://%s:%d/%s",
-                        (my_http_addr.sin_addr.s_addr == INADDR_ANY) ? "127.0.0.1" :
-                    inet_ntoa(my_http_addr.sin_addr),
-                    ntohs(my_http_addr.sin_port), feed->filename);
+                feed->child_argv[i] =
+                    av_asprintf("http://%s:%d/%s",
+                                (my_http_addr.sin_addr.s_addr == INADDR_ANY) ? "127.0.0.1" :
+                                inet_ntoa(my_http_addr.sin_addr), ntohs(my_http_addr.sin_port),
+                                feed->filename);
+                if (!feed->child_argv[i]) {
+                    ret = AVERROR(ENOMEM);
+                    goto end;
+                }
             }
-        } else if (!av_strcasecmp(cmd, "ReadOnlyFile")) {
+        } else if (!av_strcasecmp(cmd, "File") || !av_strcasecmp(cmd, "ReadOnlyFile")) {
             if (feed) {
                 get_arg(feed->feed_filename, sizeof(feed->feed_filename), &p);
-                feed->readonly = 1;
+                feed->readonly = !av_strcasecmp(cmd, "ReadOnlyFile");
             } else if (stream) {
                 get_arg(stream->feed_filename, sizeof(stream->feed_filename), &p);
             }
-        } else if (!av_strcasecmp(cmd, "File")) {
-            if (feed) {
-                get_arg(feed->feed_filename, sizeof(feed->feed_filename), &p);
-            } else if (stream)
-                get_arg(stream->feed_filename, sizeof(stream->feed_filename), &p);
         } else if (!av_strcasecmp(cmd, "Truncate")) {
             if (feed) {
                 get_arg(arg, sizeof(arg), &p);
-                feed->truncate = strtod(arg, NULL);
+                /* assume Truncate is true in case no argument is specified */
+                if (!arg[0]) {
+                    feed->truncate = 1;
+                } else {
+                    WARNING("Truncate N syntax in configuration file is deprecated, "
+                            "use Truncate alone with no arguments\n");
+                    feed->truncate = strtod(arg, NULL);
+                }
             }
         } else if (!av_strcasecmp(cmd, "FileMaxSize")) {
             if (feed) {
@@ -4249,6 +4268,10 @@ static int parse_ffconfig(const char *filename)
             } else {
                 FFStream *s;
                 stream = av_mallocz(sizeof(FFStream));
+                if (!stream) {
+                    ret = AVERROR(ENOMEM);
+                    goto end;
+                }
                 get_arg(stream->filename, sizeof(stream->filename), &p);
                 q = strrchr(stream->filename, '>');
                 if (q)
@@ -4286,7 +4309,7 @@ static int parse_ffconfig(const char *filename)
                     sfeed = sfeed->next_feed;
                 }
                 if (!sfeed)
-                    ERROR("feed '%s' not defined\n", arg);
+                    ERROR("Feed with name '%s' for stream '%s' is not defined\n", arg, stream->filename);
                 else
                     stream->feed = sfeed;
             }
@@ -4325,18 +4348,36 @@ static int parse_ffconfig(const char *filename)
             } else {
                 ERROR("FaviconURL only permitted for status streams\n");
             }
-        } else if (!av_strcasecmp(cmd, "Author")) {
-            if (stream)
-                get_arg(stream->author, sizeof(stream->author), &p);
-        } else if (!av_strcasecmp(cmd, "Comment")) {
-            if (stream)
-                get_arg(stream->comment, sizeof(stream->comment), &p);
-        } else if (!av_strcasecmp(cmd, "Copyright")) {
-            if (stream)
-                get_arg(stream->copyright, sizeof(stream->copyright), &p);
-        } else if (!av_strcasecmp(cmd, "Title")) {
-            if (stream)
-                get_arg(stream->title, sizeof(stream->title), &p);
+        } else if (!av_strcasecmp(cmd, "Author")    ||
+                   !av_strcasecmp(cmd, "Comment")   ||
+                   !av_strcasecmp(cmd, "Copyright") ||
+                   !av_strcasecmp(cmd, "Title")) {
+            get_arg(arg, sizeof(arg), &p);
+
+            if (stream) {
+                char key[32];
+                int i, ret;
+
+                for (i = 0; i < strlen(cmd); i++)
+                    key[i] = av_tolower(cmd[i]);
+                key[i] = 0;
+                WARNING("'%s' option in configuration file is deprecated, "
+                        "use 'Metadata %s VALUE' instead\n", cmd, key);
+                if ((ret = av_dict_set(&stream->metadata, key, arg, 0)) < 0) {
+                    ERROR("Could not set metadata '%s' to value '%s': %s\n",
+                          key, arg, av_err2str(ret));
+                }
+            }
+        } else if (!av_strcasecmp(cmd, "Metadata")) {
+            get_arg(arg, sizeof(arg), &p);
+            get_arg(arg2, sizeof(arg2), &p);
+            if (stream) {
+                int ret;
+                if ((ret = av_dict_set(&stream->metadata, arg, arg2, 0)) < 0) {
+                    ERROR("Could not set metadata '%s' to value '%s': %s\n",
+                          arg, arg2, av_err2str(ret));
+                }
+            }
         } else if (!av_strcasecmp(cmd, "Preroll")) {
             get_arg(arg, sizeof(arg), &p);
             if (stream)
@@ -4346,13 +4387,13 @@ static int parse_ffconfig(const char *filename)
                 stream->send_on_key = 1;
         } else if (!av_strcasecmp(cmd, "AudioCodec")) {
             get_arg(arg, sizeof(arg), &p);
-            audio_id = opt_audio_codec(arg);
+            audio_id = opt_codec(arg, AVMEDIA_TYPE_AUDIO);
             if (audio_id == AV_CODEC_ID_NONE) {
                 ERROR("Unknown AudioCodec: %s\n", arg);
             }
         } else if (!av_strcasecmp(cmd, "VideoCodec")) {
             get_arg(arg, sizeof(arg), &p);
-            video_id = opt_video_codec(arg);
+            video_id = opt_codec(arg, AVMEDIA_TYPE_VIDEO);
             if (video_id == AV_CODEC_ID_NONE) {
                 ERROR("Unknown VideoCodec: %s\n", arg);
             }
@@ -4372,11 +4413,6 @@ static int parse_ffconfig(const char *filename)
             get_arg(arg, sizeof(arg), &p);
             if (stream)
                 audio_enc.sample_rate = atoi(arg);
-        } else if (!av_strcasecmp(cmd, "AudioQuality")) {
-            get_arg(arg, sizeof(arg), &p);
-            if (stream) {
-//                audio_enc.quality = atof(arg) * 1000;
-            }
         } else if (!av_strcasecmp(cmd, "VideoBitRateRange")) {
             if (stream) {
                 int minrate, maxrate;
@@ -4418,10 +4454,14 @@ static int parse_ffconfig(const char *filename)
         } else if (!av_strcasecmp(cmd, "VideoSize")) {
             get_arg(arg, sizeof(arg), &p);
             if (stream) {
-                av_parse_video_size(&video_enc.width, &video_enc.height, arg);
-                if ((video_enc.width % 16) != 0 ||
-                    (video_enc.height % 16) != 0) {
-                    ERROR("Image size must be a multiple of 16\n");
+                ret = av_parse_video_size(&video_enc.width, &video_enc.height, arg);
+                if (ret < 0) {
+                    ERROR("Invalid video size '%s'\n", arg);
+                } else {
+                    if ((video_enc.width % 16) != 0 ||
+                        (video_enc.height % 16) != 0) {
+                        ERROR("Image size must be a multiple of 16\n");
+                    }
                 }
             }
         } else if (!av_strcasecmp(cmd, "VideoFrameRate")) {
@@ -4460,7 +4500,6 @@ static int parse_ffconfig(const char *filename)
             }
         } else if (!av_strcasecmp(cmd, "AVOptionVideo") ||
                    !av_strcasecmp(cmd, "AVOptionAudio")) {
-            char arg2[1024];
             AVCodecContext *avctx;
             int type;
             get_arg(arg, sizeof(arg), &p);
@@ -4473,7 +4512,7 @@ static int parse_ffconfig(const char *filename)
                 type = AV_OPT_FLAG_AUDIO_PARAM;
             }
             if (ffserver_opt_default(arg, arg2, avctx, type|AV_OPT_FLAG_ENCODING_PARAM)) {
-                ERROR("AVOption error: %s %s\n", arg, arg2);
+                ERROR("Error setting %s option to %s %s\n", cmd, arg, arg2);
             }
         } else if (!av_strcasecmp(cmd, "AVPresetVideo") ||
                    !av_strcasecmp(cmd, "AVPresetAudio")) {
@@ -4604,6 +4643,10 @@ static int parse_ffconfig(const char *filename)
                 ERROR("Already in a tag\n");
             } else {
                 redirect = av_mallocz(sizeof(FFStream));
+                if (!redirect) {
+                    ret = AVERROR(ENOMEM);
+                    goto end;
+                }
                 *last_stream = redirect;
                 last_stream = &redirect->next;
 
@@ -4633,9 +4676,12 @@ static int parse_ffconfig(const char *filename)
     }
 #undef ERROR
 
+end:
     fclose(f);
+    if (ret < 0)
+        return ret;
     if (errors)
-        return -1;
+        return AVERROR(EINVAL);
     else
         return 0;
 }
@@ -4690,6 +4736,7 @@ static const OptionDef options[] = {
 int main(int argc, char **argv)
 {
     struct sigaction sigact = { { 0 } };
+    int ret = 0;
 
     config_filename = av_strdup("/etc/ffserver.conf");
 
@@ -4711,8 +4758,9 @@ int main(int argc, char **argv)
     sigact.sa_flags = SA_NOCLDSTOP | SA_RESTART;
     sigaction(SIGCHLD, &sigact, 0);
 
-    if (parse_ffconfig(config_filename) < 0) {
-        fprintf(stderr, "Incorrect config file - exiting.\n");
+    if ((ret = parse_ffconfig(config_filename)) < 0) {
+        fprintf(stderr, "Error reading configuration file '%s': %s\n",
+                config_filename, av_err2str(ret));
         exit(1);
     }
     av_freep(&config_filename);

libavcodec/4xm.c

@@ -26,6 +26,7 @@
 
 #include "libavutil/avassert.h"
 #include "libavutil/frame.h"
+#include "libavutil/imgutils.h"
 #include "libavutil/intreadwrite.h"
 #include "avcodec.h"
 #include "bytestream.h"
@@ -131,7 +132,8 @@ typedef struct CFrameBuffer {
 typedef struct FourXContext {
     AVCodecContext *avctx;
     DSPContext dsp;
-    AVFrame *current_picture, *last_picture;
+    uint16_t *frame_buffer;
+    uint16_t *last_frame_buffer;
     GetBitContext pre_gb;          ///< ac/dc prefix
     GetBitContext gb;
     GetByteContext g;
@@ -341,7 +343,7 @@ static int decode_p_block(FourXContext *f, uint16_t *dst, uint16_t *src,
     int code        = get_vlc2(&f->gb,
                                block_type_vlc[1 - (f->version > 1)][index].table,
                                BLOCK_TYPE_VLC_BITS, 1);
-    uint16_t *start = (uint16_t *)f->last_picture->data[0];
+    uint16_t *start = f->last_frame_buffer;
     uint16_t *end   = start + stride * (f->avctx->height - h + 1) - (1 << log2w);
     int ret;
     int scale   = 1;
@@ -414,29 +416,18 @@ static int decode_p_block(FourXContext *f, uint16_t *dst, uint16_t *src,
     return 0;
 }
 
-static int decode_p_frame(FourXContext *f, AVFrame *frame,
-                          const uint8_t *buf, int length)
+static int decode_p_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 *)frame->data[0];
-    const int stride =             frame->linesize[0] >> 1;
+    uint16_t *dst    = f->frame_buffer;
     uint16_t *src;
     unsigned int bitstream_size, bytestream_size, wordstream_size, extra,
                  bytestream_offset, wordstream_offset;
     int ret;
 
-    if (!f->last_picture->data[0]) {
-        if ((ret = ff_get_buffer(f->avctx, f->last_picture,
-                                 AV_GET_BUFFER_FLAG_REF)) < 0) {
-            return ret;
-        }
-        for (y=0; y<f->avctx->height; y++)
-            memset(f->last_picture->data[0] + y*f->last_picture->linesize[0], 0, 2*f->avctx->width);
-    }
-
-    src = (uint16_t *)f->last_picture->data[0];
+    src = f->last_frame_buffer;
 
     if (f->version > 1) {
         extra           = 20;
@@ -461,14 +452,12 @@ static int decode_p_frame(FourXContext *f, AVFrame *frame,
         return AVERROR_INVALIDDATA;
     }
 
-    av_fast_malloc(&f->bitstream_buffer, &f->bitstream_buffer_size,
-                   bitstream_size + FF_INPUT_BUFFER_PADDING_SIZE);
+    av_fast_padded_malloc(&f->bitstream_buffer, &f->bitstream_buffer_size,
+                          bitstream_size);
     if (!f->bitstream_buffer)
         return AVERROR(ENOMEM);
     f->dsp.bswap_buf(f->bitstream_buffer, (const uint32_t*)(buf + extra),
                      bitstream_size / 4);
-    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;
@@ -478,14 +467,14 @@ static int decode_p_frame(FourXContext *f, AVFrame *frame,
     bytestream2_init(&f->g, buf + bytestream_offset,
                      length - bytestream_offset);
 
-    init_mv(f, frame->linesize[0]);
+    init_mv(f, width * 2);
 
     for (y = 0; y < height; y += 8) {
         for (x = 0; x < width; x += 8)
-            if ((ret = decode_p_block(f, dst + x, src + x, 3, 3, stride)) < 0)
+            if ((ret = decode_p_block(f, dst + x, src + x, 3, 3, width)) < 0)
                 return ret;
-        src += 8 * stride;
-        dst += 8 * stride;
+        src += 8 * width;
+        dst += 8 * width;
     }
 
     return 0;
@@ -550,12 +539,12 @@ static int decode_i_block(FourXContext *f, int16_t *block)
     return 0;
 }
 
-static inline void idct_put(FourXContext *f, AVFrame *frame, int x, int y)
+static inline void idct_put(FourXContext *f, int x, int y)
 {
     int16_t (*block)[64] = f->block;
-    int stride           = frame->linesize[0] >> 1;
+    int stride           = f->avctx->width;
     int i;
-    uint16_t *dst = ((uint16_t*)frame->data[0]) + y * stride + x;
+    uint16_t *dst = f->frame_buffer + y * stride + x;
 
     for (i = 0; i < 4; i++) {
         block[i][0] += 0x80 * 8 * 8;
@@ -715,14 +704,13 @@ static int mix(int c0, int c1)
     return red / 3 * 1024 + green / 3 * 32 + blue / 3;
 }
 
-static int decode_i2_frame(FourXContext *f, AVFrame *frame, const uint8_t *buf, int length)
+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*)frame->data[0];
-    const int stride =            frame->linesize[0]>>1;
+    uint16_t *dst    = f->frame_buffer;
     const uint8_t *buf_end = buf + length;
     GetByteContext g3;
 
@@ -753,18 +741,18 @@ static int decode_i2_frame(FourXContext *f, AVFrame *frame, const uint8_t *buf,
             for (y2 = 0; y2 < 16; y2++) {
                 for (x2 = 0; x2 < 16; x2++) {
                     int index = 2 * (x2 >> 2) + 8 * (y2 >> 2);
-                    dst[y2 * stride + x2] = color[(bits >> index) & 3];
+                    dst[y2 * width + x2] = color[(bits >> index) & 3];
                 }
             }
             dst += 16;
         }
-        dst += 16 * stride - x;
+        dst += 16 * width - x;
     }
 
     return 0;
 }
 
-static int decode_i_frame(FourXContext *f, AVFrame *frame, const uint8_t *buf, int length)
+static int decode_i_frame(FourXContext *f, const uint8_t *buf, int length)
 {
     int x, y, ret;
     const int width  = f->avctx->width;
@@ -803,14 +791,12 @@ static int decode_i_frame(FourXContext *f, AVFrame *frame, const uint8_t *buf, i
 
     prestream_size = length + buf - prestream;
 
-    av_fast_malloc(&f->bitstream_buffer, &f->bitstream_buffer_size,
-                   prestream_size + FF_INPUT_BUFFER_PADDING_SIZE);
+    av_fast_padded_malloc(&f->bitstream_buffer, &f->bitstream_buffer_size,
+                          prestream_size);
     if (!f->bitstream_buffer)
         return AVERROR(ENOMEM);
     f->dsp.bswap_buf(f->bitstream_buffer, (const uint32_t*)prestream,
                      prestream_size / 4);
-    memset((uint8_t*)f->bitstream_buffer + prestream_size,
-           0, FF_INPUT_BUFFER_PADDING_SIZE);
     init_get_bits(&f->pre_gb, f->bitstream_buffer, 8 * prestream_size);
 
     f->last_dc = 0 * 128 * 8 * 8;
@@ -820,7 +806,7 @@ static int decode_i_frame(FourXContext *f, AVFrame *frame, const uint8_t *buf, i
             if ((ret = decode_i_mb(f)) < 0)
                 return ret;
 
-            idct_put(f, frame, x, y);
+            idct_put(f, x, y);
         }
     }
 
@@ -923,29 +909,24 @@ static int decode_frame(AVCodecContext *avctx, void *data,
         frame_size = buf_size - 12;
     }
 
-    FFSWAP(AVFrame*, f->current_picture, f->last_picture);
-
-    // alternatively we would have to use our own buffer management
-    avctx->flags |= CODEC_FLAG_EMU_EDGE;
-
-    if ((ret = ff_reget_buffer(avctx, f->current_picture)) < 0)
+    if ((ret = ff_get_buffer(avctx, picture, 0)) < 0)
         return ret;
 
     if (frame_4cc == AV_RL32("ifr2")) {
-        f->current_picture->pict_type = AV_PICTURE_TYPE_I;
-        if ((ret = decode_i2_frame(f, f->current_picture, buf - 4, frame_size + 4)) < 0) {
+        picture->pict_type = AV_PICTURE_TYPE_I;
+        if ((ret = decode_i2_frame(f, buf - 4, frame_size + 4)) < 0) {
             av_log(f->avctx, AV_LOG_ERROR, "decode i2 frame failed\n");
             return ret;
         }
     } else if (frame_4cc == AV_RL32("ifrm")) {
-        f->current_picture->pict_type = AV_PICTURE_TYPE_I;
-        if ((ret = decode_i_frame(f, f->current_picture, buf, frame_size)) < 0) {
+        picture->pict_type = AV_PICTURE_TYPE_I;
+        if ((ret = decode_i_frame(f, buf, frame_size)) < 0) {
             av_log(f->avctx, AV_LOG_ERROR, "decode i frame failed\n");
             return ret;
         }
     } else if (frame_4cc == AV_RL32("pfrm") || frame_4cc == AV_RL32("pfr2")) {
-        f->current_picture->pict_type = AV_PICTURE_TYPE_P;
-        if ((ret = decode_p_frame(f, f->current_picture, buf, frame_size)) < 0) {
+        picture->pict_type = AV_PICTURE_TYPE_P;
+        if ((ret = decode_p_frame(f, buf, frame_size)) < 0) {
             av_log(f->avctx, AV_LOG_ERROR, "decode p frame failed\n");
             return ret;
         }
@@ -957,10 +938,13 @@ static int decode_frame(AVCodecContext *avctx, void *data,
                buf_size);
     }
 
-    f->current_picture->key_frame = f->current_picture->pict_type == AV_PICTURE_TYPE_I;
+    picture->key_frame = picture->pict_type == AV_PICTURE_TYPE_I;
+
+    av_image_copy_plane(picture->data[0], picture->linesize[0],
+                        (const uint8_t*)f->frame_buffer,  avctx->width * 2,
+                        avctx->width * 2, avctx->height);
+    FFSWAP(uint16_t *, f->frame_buffer, f->last_frame_buffer);
 
-    if ((ret = av_frame_ref(picture, f->current_picture)) < 0)
-        return ret;
     *got_frame = 1;
 
     emms_c();
@@ -968,9 +952,28 @@ static int decode_frame(AVCodecContext *avctx, void *data,
     return buf_size;
 }
 
+static av_cold int decode_end(AVCodecContext *avctx)
+{
+    FourXContext * const f = avctx->priv_data;
+    int i;
+
+    av_freep(&f->frame_buffer);
+    av_freep(&f->last_frame_buffer);
+    av_freep(&f->bitstream_buffer);
+    f->bitstream_buffer_size = 0;
+    for (i = 0; i < CFRAME_BUFFER_COUNT; i++) {
+        av_freep(&f->cfrm[i].data);
+        f->cfrm[i].allocated_size = 0;
+    }
+    ff_free_vlc(&f->pre_vlc);
+
+    return 0;
+}
+
 static av_cold int decode_init(AVCodecContext *avctx)
 {
     FourXContext * const f = avctx->priv_data;
+    int ret;
 
     if (avctx->extradata_size != 4 || !avctx->extradata) {
         av_log(avctx, AV_LOG_ERROR, "extradata wrong or missing\n");
@@ -981,6 +984,17 @@ static av_cold int decode_init(AVCodecContext *avctx)
         return AVERROR_INVALIDDATA;
     }
 
+    ret = av_image_check_size(avctx->width, avctx->height, 0, avctx);
+    if (ret < 0)
+        return ret;
+
+    f->frame_buffer      = av_mallocz(avctx->width * avctx->height * 2);
+    f->last_frame_buffer = av_mallocz(avctx->width * avctx->height * 2);
+    if (!f->frame_buffer || !f->last_frame_buffer) {
+        decode_end(avctx);
+        return AVERROR(ENOMEM);
+    }
+
     f->version = AV_RL32(avctx->extradata) >> 16;
     ff_dsputil_init(&f->dsp, avctx);
     f->avctx = avctx;
@@ -991,30 +1005,6 @@ static av_cold int decode_init(AVCodecContext *avctx)
     else
         avctx->pix_fmt = AV_PIX_FMT_BGR555;
 
-    f->current_picture = av_frame_alloc();
-    f->last_picture    = av_frame_alloc();
-    if (!f->current_picture  || !f->last_picture)
-        return AVERROR(ENOMEM);
-
-    return 0;
-}
-
-
-static av_cold int decode_end(AVCodecContext *avctx)
-{
-    FourXContext * const f = avctx->priv_data;
-    int i;
-
-    av_freep(&f->bitstream_buffer);
-    f->bitstream_buffer_size = 0;
-    for (i = 0; i < CFRAME_BUFFER_COUNT; i++) {
-        av_freep(&f->cfrm[i].data);
-        f->cfrm[i].allocated_size = 0;
-    }
-    ff_free_vlc(&f->pre_vlc);
-    av_frame_free(&f->current_picture);
-    av_frame_free(&f->last_picture);
-
     return 0;
 }
 

libavcodec/Makefile

@@ -26,13 +26,10 @@ OBJS = allcodecs.o                                                      \
        options.o                                                        \
        parser.o                                                         \
        raw.o                                                            \
-       rawdec.o                                                         \
        resample.o                                                       \
        resample2.o                                                      \
        utils.o                                                          \
 
-OBJS-$(HAVE_MSVCRT) += file_open.o
-
 # parts needed for many different codecs
 OBJS-$(CONFIG_AANDCTTABLES)            += aandcttab.o
 OBJS-$(CONFIG_AC3DSP)                  += ac3dsp.o
@@ -50,13 +47,16 @@ OBJS-$(CONFIG_FFT)                     += avfft.o fft_fixed.o fft_float.o \
                                           fft_fixed_32.o fft_init_table.o \
                                           $(FFT-OBJS-yes)
 OBJS-$(CONFIG_GOLOMB)                  += golomb.o
+OBJS-$(CONFIG_H263DSP)                 += h263dsp.o
 OBJS-$(CONFIG_H264CHROMA)              += h264chroma.o
 OBJS-$(CONFIG_H264DSP)                 += h264dsp.o h264idct.o
 OBJS-$(CONFIG_H264PRED)                += h264pred.o
 OBJS-$(CONFIG_H264QPEL)                += h264qpel.o
 OBJS-$(CONFIG_HPELDSP)                 += hpeldsp.o
 OBJS-$(CONFIG_HUFFMAN)                 += huffman.o
+OBJS-$(CONFIG_INTRAX8)                 += intrax8.o intrax8dsp.o
 OBJS-$(CONFIG_LIBXVID)                 += libxvid_rc.o
+OBJS-$(CONFIG_LLVIDDSP)                += lossless_videodsp.o
 OBJS-$(CONFIG_LPC)                     += lpc.o
 OBJS-$(CONFIG_LSP)                     += lsp.o
 OBJS-$(CONFIG_MDCT)                    += mdct_fixed.o mdct_float.o mdct_fixed_32.o
@@ -125,6 +125,8 @@ OBJS-$(CONFIG_ASV2_DECODER)            += asvdec.o asv.o mpeg12data.o
 OBJS-$(CONFIG_ASV2_ENCODER)            += asvenc.o asv.o mpeg12data.o
 OBJS-$(CONFIG_ATRAC1_DECODER)          += atrac1.o atrac.o
 OBJS-$(CONFIG_ATRAC3_DECODER)          += atrac3.o atrac.o
+OBJS-$(CONFIG_ATRAC3P_DECODER)         += atrac3plusdec.o atrac3plus.o \
+                                          atrac3plusdsp.o atrac.o
 OBJS-$(CONFIG_AURA_DECODER)            += cyuv.o
 OBJS-$(CONFIG_AURA2_DECODER)           += aura.o
 OBJS-$(CONFIG_AVRN_DECODER)            += avrndec.o mjpegdec.o mjpeg.o
@@ -152,6 +154,7 @@ OBJS-$(CONFIG_CAVS_DECODER)            += cavs.o cavsdec.o cavsdsp.o \
 OBJS-$(CONFIG_CDGRAPHICS_DECODER)      += cdgraphics.o
 OBJS-$(CONFIG_CDXL_DECODER)            += cdxl.o
 OBJS-$(CONFIG_CINEPAK_DECODER)         += cinepak.o
+OBJS-$(CONFIG_CINEPAK_ENCODER)         += cinepakenc.o elbg.o
 OBJS-$(CONFIG_CLJR_DECODER)            += cljr.o
 OBJS-$(CONFIG_CLJR_ENCODER)            += cljr.o
 OBJS-$(CONFIG_CLLC_DECODER)            += cllc.o
@@ -178,7 +181,7 @@ OBJS-$(CONFIG_DVBSUB_ENCODER)          += dvbsub.o
 OBJS-$(CONFIG_DVDSUB_DECODER)          += dvdsubdec.o
 OBJS-$(CONFIG_DVDSUB_ENCODER)          += dvdsubenc.o
 OBJS-$(CONFIG_DVVIDEO_DECODER)         += dvdec.o dv.o dvdata.o dv_profile.o
-OBJS-$(CONFIG_DVVIDEO_ENCODER)         += dv.o dvdata.o dv_profile.o
+OBJS-$(CONFIG_DVVIDEO_ENCODER)         += dvenc.o dv.o dvdata.o dv_profile.o
 OBJS-$(CONFIG_DXA_DECODER)             += dxa.o
 OBJS-$(CONFIG_DXTORY_DECODER)          += dxtory.o
 OBJS-$(CONFIG_EAC3_DECODER)            += eac3dec.o eac3_data.o
@@ -202,6 +205,7 @@ OBJS-$(CONFIG_FFV1_ENCODER)            += ffv1enc.o ffv1.o
 OBJS-$(CONFIG_FFVHUFF_DECODER)         += huffyuv.o huffyuvdec.o
 OBJS-$(CONFIG_FFVHUFF_ENCODER)         += huffyuv.o huffyuvenc.o
 OBJS-$(CONFIG_FFWAVESYNTH_DECODER)     += ffwavesynth.o
+OBJS-$(CONFIG_FIC_DECODER)             += fic.o
 OBJS-$(CONFIG_FLAC_DECODER)            += flacdec.o flacdata.o flac.o flacdsp.o
 OBJS-$(CONFIG_FLAC_ENCODER)            += flacenc.o flacdata.o flac.o flacdsp.o vorbis_data.o
 OBJS-$(CONFIG_FLASHSV_DECODER)         += flashsv.o
@@ -236,7 +240,7 @@ OBJS-$(CONFIG_H264_VDA_DECODER)        += vda_h264_dec.o
 OBJS-$(CONFIG_HEVC_DECODER)            += hevc.o hevc_mvs.o hevc_ps.o hevc_sei.o \
                                           hevc_cabac.o hevc_refs.o hevcpred.o    \
                                           hevcdsp.o hevc_filter.o cabac.o
-
+OBJS-$(CONFIG_HNM4_VIDEO_DECODER)      += hnm4video.o
 OBJS-$(CONFIG_HUFFYUV_DECODER)         += huffyuv.o huffyuvdec.o
 OBJS-$(CONFIG_HUFFYUV_ENCODER)         += huffyuv.o huffyuvenc.o
 OBJS-$(CONFIG_IAC_DECODER)             += imc.o
@@ -280,40 +284,34 @@ OBJS-$(CONFIG_MMVIDEO_DECODER)         += mmvideo.o
 OBJS-$(CONFIG_MOTIONPIXELS_DECODER)    += motionpixels.o
 OBJS-$(CONFIG_MOVTEXT_DECODER)         += movtextdec.o ass.o
 OBJS-$(CONFIG_MOVTEXT_ENCODER)         += movtextenc.o ass_split.o
-OBJS-$(CONFIG_MP1_DECODER)             += mpegaudiodec.o
+OBJS-$(CONFIG_MP1_DECODER)             += mpegaudiodec_fixed.o
 OBJS-$(CONFIG_MP1FLOAT_DECODER)        += mpegaudiodec_float.o
-OBJS-$(CONFIG_MP2_DECODER)             += mpegaudiodec.o
-OBJS-$(CONFIG_MP2_ENCODER)             += mpegaudioenc.o mpegaudio.o \
+OBJS-$(CONFIG_MP2_DECODER)             += mpegaudiodec_fixed.o
+OBJS-$(CONFIG_MP2_ENCODER)             += mpegaudioenc_float.o mpegaudio.o \
+                                          mpegaudiodata.o mpegaudiodsp_data.o
+OBJS-$(CONFIG_MP2FIXED_ENCODER)        += mpegaudioenc_fixed.o mpegaudio.o \
                                           mpegaudiodata.o mpegaudiodsp_data.o
 OBJS-$(CONFIG_MP2FLOAT_DECODER)        += mpegaudiodec_float.o
-OBJS-$(CONFIG_MP3_DECODER)             += mpegaudiodec.o
-OBJS-$(CONFIG_MP3ADU_DECODER)          += mpegaudiodec.o
+OBJS-$(CONFIG_MP3_DECODER)             += mpegaudiodec_fixed.o
+OBJS-$(CONFIG_MP3ADU_DECODER)          += mpegaudiodec_fixed.o
 OBJS-$(CONFIG_MP3ADUFLOAT_DECODER)     += mpegaudiodec_float.o
 OBJS-$(CONFIG_MP3FLOAT_DECODER)        += mpegaudiodec_float.o
-OBJS-$(CONFIG_MP3ON4_DECODER)          += mpegaudiodec.o mpeg4audio.o
+OBJS-$(CONFIG_MP3ON4_DECODER)          += mpegaudiodec_fixed.o mpeg4audio.o
 OBJS-$(CONFIG_MP3ON4FLOAT_DECODER)     += mpegaudiodec_float.o mpeg4audio.o
 OBJS-$(CONFIG_MPC7_DECODER)            += mpc7.o mpc.o
 OBJS-$(CONFIG_MPC8_DECODER)            += mpc8.o mpc.o
 OBJS-$(CONFIG_MPEGVIDEO_DECODER)       += mpeg12.o mpeg12data.o \
                                           mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MPEG_XVMC_DECODER)       += mpegvideo_xvmc.o
 OBJS-$(CONFIG_MPEG1VIDEO_DECODER)      += mpeg12dec.o mpeg12.o mpeg12data.o
 OBJS-$(CONFIG_MPEG1VIDEO_ENCODER)      += mpeg12enc.o mpeg12.o
 OBJS-$(CONFIG_MPEG2VIDEO_DECODER)      += mpeg12dec.o mpeg12.o mpeg12data.o
-OBJS-$(CONFIG_MPEG2VIDEO_ENCODER)      += mpeg12enc.o mpeg12.o          \
-                                          timecode.o
+OBJS-$(CONFIG_MPEG2VIDEO_ENCODER)      += mpeg12enc.o mpeg12.o
 OBJS-$(CONFIG_MPL2_DECODER)            += mpl2dec.o ass.o
 OBJS-$(CONFIG_MSMPEG4V1_DECODER)       += msmpeg4dec.o msmpeg4.o msmpeg4data.o
-OBJS-$(CONFIG_MSMPEG4V2_DECODER)       += msmpeg4dec.o msmpeg4.o msmpeg4data.o \
-                                          h263dec.o h263.o ituh263dec.o \
-                                          mpeg4videodec.o
-OBJS-$(CONFIG_MSMPEG4V2_ENCODER)       += msmpeg4.o msmpeg4enc.o msmpeg4data.o \
-                                          h263.o
-OBJS-$(CONFIG_MSMPEG4V3_DECODER)       += msmpeg4dec.o msmpeg4.o msmpeg4data.o \
-                                          h263dec.o h263.o ituh263dec.o \
-                                          mpeg4videodec.o
-OBJS-$(CONFIG_MSMPEG4V3_ENCODER)       += msmpeg4.o msmpeg4enc.o msmpeg4data.o \
-                                          h263.o
+OBJS-$(CONFIG_MSMPEG4V2_DECODER)       += msmpeg4dec.o msmpeg4.o msmpeg4data.o
+OBJS-$(CONFIG_MSMPEG4V2_ENCODER)       += msmpeg4enc.o msmpeg4.o msmpeg4data.o
+OBJS-$(CONFIG_MSMPEG4V3_DECODER)       += msmpeg4dec.o msmpeg4.o msmpeg4data.o
+OBJS-$(CONFIG_MSMPEG4V3_ENCODER)       += msmpeg4enc.o msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_MSRLE_DECODER)           += msrle.o msrledec.o
 OBJS-$(CONFIG_MSA1_DECODER)            += mss3.o mss34dsp.o
 OBJS-$(CONFIG_MSS1_DECODER)            += mss1.o mss12.o
@@ -331,27 +329,27 @@ OBJS-$(CONFIG_NUV_DECODER)             += nuv.o rtjpeg.o
 OBJS-$(CONFIG_PAF_VIDEO_DECODER)       += paf.o
 OBJS-$(CONFIG_PAF_AUDIO_DECODER)       += paf.o
 OBJS-$(CONFIG_PAM_DECODER)             += pnmdec.o pnm.o
-OBJS-$(CONFIG_PAM_ENCODER)             += pamenc.o pnm.o
+OBJS-$(CONFIG_PAM_ENCODER)             += pamenc.o
 OBJS-$(CONFIG_PBM_DECODER)             += pnmdec.o pnm.o
-OBJS-$(CONFIG_PBM_ENCODER)             += pnmenc.o pnm.o
+OBJS-$(CONFIG_PBM_ENCODER)             += pnmenc.o
 OBJS-$(CONFIG_PCX_DECODER)             += pcx.o
 OBJS-$(CONFIG_PCX_ENCODER)             += pcxenc.o
 OBJS-$(CONFIG_PGM_DECODER)             += pnmdec.o pnm.o
-OBJS-$(CONFIG_PGM_ENCODER)             += pnmenc.o pnm.o
+OBJS-$(CONFIG_PGM_ENCODER)             += pnmenc.o
 OBJS-$(CONFIG_PGMYUV_DECODER)          += pnmdec.o pnm.o
-OBJS-$(CONFIG_PGMYUV_ENCODER)          += pnmenc.o pnm.o
+OBJS-$(CONFIG_PGMYUV_ENCODER)          += pnmenc.o
 OBJS-$(CONFIG_PGSSUB_DECODER)          += pgssubdec.o
 OBJS-$(CONFIG_PICTOR_DECODER)          += pictordec.o cga_data.o
 OBJS-$(CONFIG_PJS_DECODER)             += textdec.o ass.o
 OBJS-$(CONFIG_PNG_DECODER)             += png.o pngdec.o pngdsp.o
 OBJS-$(CONFIG_PNG_ENCODER)             += png.o pngenc.o
 OBJS-$(CONFIG_PPM_DECODER)             += pnmdec.o pnm.o
-OBJS-$(CONFIG_PPM_ENCODER)             += pnmenc.o pnm.o
+OBJS-$(CONFIG_PPM_ENCODER)             += pnmenc.o
 OBJS-$(CONFIG_PRORES_DECODER)          += proresdec2.o proresdsp.o proresdata.o
 OBJS-$(CONFIG_PRORES_LGPL_DECODER)     += proresdec_lgpl.o proresdsp.o proresdata.o
 OBJS-$(CONFIG_PRORES_ENCODER)          += proresenc_anatoliy.o
 OBJS-$(CONFIG_PRORES_AW_ENCODER)       += proresenc_anatoliy.o
-OBJS-$(CONFIG_PRORES_KS_ENCODER)       += proresenc_kostya.o proresdata.o proresdsp.o
+OBJS-$(CONFIG_PRORES_KS_ENCODER)       += proresenc_kostya.o proresdata.o
 OBJS-$(CONFIG_PTX_DECODER)             += ptx.o
 OBJS-$(CONFIG_QCELP_DECODER)           += qcelpdec.o                     \
                                           celp_filters.o acelp_vectors.o \
@@ -440,7 +438,7 @@ OBJS-$(CONFIG_TRUEMOTION2_DECODER)     += truemotion2.o
 OBJS-$(CONFIG_TRUESPEECH_DECODER)      += truespeech.o
 OBJS-$(CONFIG_TSCC_DECODER)            += tscc.o msrledec.o
 OBJS-$(CONFIG_TSCC2_DECODER)           += tscc2.o
-OBJS-$(CONFIG_TTA_DECODER)             += tta.o ttadata.o
+OBJS-$(CONFIG_TTA_DECODER)             += tta.o ttadata.o ttadsp.o
 OBJS-$(CONFIG_TTA_ENCODER)             += ttaenc.o ttadata.o
 OBJS-$(CONFIG_TWINVQ_DECODER)          += twinvqdec.o twinvq.o
 OBJS-$(CONFIG_TXD_DECODER)             += txd.o s3tc.o
@@ -460,7 +458,7 @@ OBJS-$(CONFIG_VB_DECODER)              += vb.o
 OBJS-$(CONFIG_VBLE_DECODER)            += vble.o
 OBJS-$(CONFIG_VC1_DECODER)             += vc1dec.o vc1.o vc1data.o vc1dsp.o \
                                           msmpeg4dec.o msmpeg4.o msmpeg4data.o \
-                                          intrax8.o intrax8dsp.o wmv2dsp.o
+                                          wmv2dsp.o
 OBJS-$(CONFIG_VCR1_DECODER)            += vcr1.o
 OBJS-$(CONFIG_VMDAUDIO_DECODER)        += vmdav.o
 OBJS-$(CONFIG_VMDVIDEO_DECODER)        += vmdav.o
@@ -481,7 +479,7 @@ OBJS-$(CONFIG_VQA_DECODER)             += vqavideo.o
 OBJS-$(CONFIG_WAVPACK_DECODER)         += wavpack.o
 OBJS-$(CONFIG_WAVPACK_ENCODER)         += wavpackenc.o
 OBJS-$(CONFIG_WEBP_DECODER)            += vp8.o vp8dsp.o vp56rac.o
-OBJS-$(CONFIG_WEBP_DECODER)            += webp.o
+OBJS-$(CONFIG_WEBP_DECODER)            += webp.o exif.o tiff_common.o
 OBJS-$(CONFIG_WEBVTT_DECODER)          += webvttdec.o
 OBJS-$(CONFIG_WMALOSSLESS_DECODER)     += wmalosslessdec.o wma_common.o
 OBJS-$(CONFIG_WMAPRO_DECODER)          += wmaprodec.o wma.o wma_common.o
@@ -494,8 +492,7 @@ OBJS-$(CONFIG_WMAVOICE_DECODER)        += wmavoice.o \
                                           acelp_vectors.o acelp_filters.o
 OBJS-$(CONFIG_WMV1_DECODER)            += msmpeg4dec.o msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_WMV2_DECODER)            += wmv2dec.o wmv2.o wmv2dsp.o \
-                                          msmpeg4dec.o msmpeg4.o msmpeg4data.o \
-                                          intrax8.o intrax8dsp.o
+                                          msmpeg4dec.o msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_WMV2_ENCODER)            += wmv2enc.o wmv2.o wmv2dsp.o \
                                           msmpeg4.o msmpeg4enc.o msmpeg4data.o
 OBJS-$(CONFIG_WNV1_DECODER)            += wnv1.o
@@ -598,6 +595,7 @@ 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-$(CONFIG_ADPCM_G726LE_DECODER)       += g726.o
 OBJS-$(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
@@ -634,9 +632,11 @@ OBJS-$(CONFIG_H264_VAAPI_HWACCEL)         += vaapi_h264.o
 OBJS-$(CONFIG_H264_VDA_HWACCEL)           += vda_h264.o
 OBJS-$(CONFIG_H264_VDPAU_HWACCEL)         += vdpau_h264.o
 OBJS-$(CONFIG_MPEG1_VDPAU_HWACCEL)        += vdpau_mpeg12.o
+OBJS-$(CONFIG_MPEG1_XVMC_HWACCEL)         += mpegvideo_xvmc.o
 OBJS-$(CONFIG_MPEG2_DXVA2_HWACCEL)        += dxva2_mpeg2.o
 OBJS-$(CONFIG_MPEG2_VAAPI_HWACCEL)        += vaapi_mpeg2.o vaapi_mpeg.o
 OBJS-$(CONFIG_MPEG2_VDPAU_HWACCEL)        += vdpau_mpeg12.o
+OBJS-$(CONFIG_MPEG2_XVMC_HWACCEL)         += mpegvideo_xvmc.o
 OBJS-$(CONFIG_MPEG4_VAAPI_HWACCEL)        += vaapi_mpeg4.o vaapi_mpeg.o
 OBJS-$(CONFIG_MPEG4_VDPAU_HWACCEL)        += vdpau_mpeg4.o
 OBJS-$(CONFIG_VC1_DXVA2_HWACCEL)          += dxva2_vc1.o
@@ -649,7 +649,7 @@ OBJS-$(CONFIG_ADX_DEMUXER)             += adx.o
 OBJS-$(CONFIG_CAF_DEMUXER)             += mpeg4audio.o mpegaudiodata.o  \
                                           ac3tab.o
 OBJS-$(CONFIG_DV_DEMUXER)              += dv_profile.o
-OBJS-$(CONFIG_DV_MUXER)                += dv_profile.o timecode.o
+OBJS-$(CONFIG_DV_MUXER)                += dv_profile.o
 OBJS-$(CONFIG_FLAC_DEMUXER)            += flac.o flacdata.o vorbis_data.o \
                                           vorbis_parser.o xiph.o
 OBJS-$(CONFIG_FLAC_MUXER)              += flac.o flacdata.o vorbis_data.o
@@ -665,11 +665,11 @@ OBJS-$(CONFIG_MATROSKA_MUXER)          += mpeg4audio.o mpegaudiodata.o  \
                                           flac.o flacdata.o vorbis_data.o xiph.o
 OBJS-$(CONFIG_MP2_MUXER)               += mpegaudiodata.o mpegaudiodecheader.o
 OBJS-$(CONFIG_MP3_MUXER)               += mpegaudiodata.o mpegaudiodecheader.o
-OBJS-$(CONFIG_MOV_DEMUXER)             += mpeg4audio.o mpegaudiodata.o ac3tab.o timecode.o
+OBJS-$(CONFIG_MOV_DEMUXER)             += mpeg4audio.o mpegaudiodata.o ac3tab.o
 OBJS-$(CONFIG_MOV_MUXER)               += mpeg4audio.o mpegaudiodata.o
 OBJS-$(CONFIG_MPEGTS_MUXER)            += mpeg4audio.o
 OBJS-$(CONFIG_MPEGTS_DEMUXER)          += mpeg4audio.o mpegaudiodata.o
-OBJS-$(CONFIG_MXF_MUXER)               += timecode.o dnxhddata.o
+OBJS-$(CONFIG_MXF_MUXER)               += dnxhddata.o
 OBJS-$(CONFIG_NUT_MUXER)               += mpegaudiodata.o
 OBJS-$(CONFIG_OGG_DEMUXER)             += xiph.o flac.o flacdata.o     \
                                           mpeg12data.o vorbis_parser.o \
@@ -686,6 +686,9 @@ OBJS-$(CONFIG_WEBM_MUXER)              += mpeg4audio.o mpegaudiodata.o  \
                                           vorbis_data.o
 OBJS-$(CONFIG_WTV_DEMUXER)             += mpeg4audio.o mpegaudiodata.o
 
+# libavfilter dependencies
+OBJS-$(CONFIG_ELBG_FILTER)             += elbg.o
+
 # external codec libraries
 OBJS-$(CONFIG_LIBAACPLUS_ENCODER)         += libaacplus.o
 OBJS-$(CONFIG_LIBCELT_DECODER)            += libcelt_dec.o
@@ -727,10 +730,12 @@ OBJS-$(CONFIG_LIBVORBIS_ENCODER)          += libvorbisenc.o \
                                              vorbis_data.o vorbis_parser.o xiph.o
 OBJS-$(CONFIG_LIBVPX_VP8_DECODER)         += libvpxdec.o
 OBJS-$(CONFIG_LIBVPX_VP8_ENCODER)         += libvpxenc.o
-OBJS-$(CONFIG_LIBVPX_VP9_DECODER)         += libvpxdec.o
-OBJS-$(CONFIG_LIBVPX_VP9_ENCODER)         += libvpxenc.o
+OBJS-$(CONFIG_LIBVPX_VP9_DECODER)         += libvpxdec.o libvpx.o
+OBJS-$(CONFIG_LIBVPX_VP9_ENCODER)         += libvpxenc.o libvpx.o
 OBJS-$(CONFIG_LIBWAVPACK_ENCODER)         += libwavpackenc.o
+OBJS-$(CONFIG_LIBWEBP_ENCODER)            += libwebpenc.o
 OBJS-$(CONFIG_LIBX264_ENCODER)            += libx264.o
+OBJS-$(CONFIG_LIBX265_ENCODER)            += libx265.o
 OBJS-$(CONFIG_LIBXAVS_ENCODER)            += libxavs.o
 OBJS-$(CONFIG_LIBXVID_ENCODER)            += libxvid.o
 OBJS-$(CONFIG_LIBZVBI_TELETEXT_DECODER)   += libzvbi-teletextdec.o
@@ -773,6 +778,7 @@ OBJS-$(CONFIG_MPEGAUDIO_PARSER)        += mpegaudio_parser.o \
                                           mpegaudiodecheader.o mpegaudiodata.o
 OBJS-$(CONFIG_MPEGVIDEO_PARSER)        += mpegvideo_parser.o    \
                                           mpeg12.o mpeg12data.o
+OBJS-$(CONFIG_PNG_PARSER)              += png_parser.o
 OBJS-$(CONFIG_PNM_PARSER)              += pnm_parser.o pnm.o
 OBJS-$(CONFIG_RV30_PARSER)             += rv34_parser.o
 OBJS-$(CONFIG_RV40_PARSER)             += rv34_parser.o
@@ -783,6 +789,7 @@ OBJS-$(CONFIG_VC1_PARSER)              += vc1_parser.o vc1.o vc1data.o \
 OBJS-$(CONFIG_VORBIS_PARSER)           += vorbis_parser.o xiph.o
 OBJS-$(CONFIG_VP3_PARSER)              += vp3_parser.o
 OBJS-$(CONFIG_VP8_PARSER)              += vp8_parser.o
+OBJS-$(CONFIG_VP9_PARSER)              += vp9_parser.o
 
 # bitstream filters
 OBJS-$(CONFIG_AAC_ADTSTOASC_BSF)          += aac_adtstoasc_bsf.o aacadtsdec.o \
@@ -794,7 +801,6 @@ OBJS-$(CONFIG_IMX_DUMP_HEADER_BSF)        += imx_dump_header_bsf.o
 OBJS-$(CONFIG_MJPEG2JPEG_BSF)             += mjpeg2jpeg_bsf.o mjpeg.o
 OBJS-$(CONFIG_MJPEGA_DUMP_HEADER_BSF)     += mjpega_dump_header_bsf.o
 OBJS-$(CONFIG_MOV2TEXTSUB_BSF)            += movsub_bsf.o
-OBJS-$(CONFIG_MP3_HEADER_COMPRESS_BSF)    += mp3_header_compress_bsf.o
 OBJS-$(CONFIG_MP3_HEADER_DECOMPRESS_BSF)  += mp3_header_decompress_bsf.o \
                                              mpegaudiodata.o
 OBJS-$(CONFIG_NOISE_BSF)                  += noise_bsf.o
@@ -802,12 +808,14 @@ OBJS-$(CONFIG_REMOVE_EXTRADATA_BSF)       += remove_extradata_bsf.o
 OBJS-$(CONFIG_TEXT2MOVSUB_BSF)            += movsub_bsf.o
 
 # thread libraries
-OBJS-$(HAVE_PTHREADS)                  += pthread.o
-OBJS-$(HAVE_W32THREADS)                += pthread.o
-OBJS-$(HAVE_OS2THREADS)                += pthread.o
+OBJS-$(HAVE_LIBC_MSVCRT)               += file_open.o
+OBJS-$(HAVE_THREADS)                   += pthread.o pthread_slice.o pthread_frame.o
 
 OBJS-$(CONFIG_FRAME_THREAD_ENCODER)    += frame_thread_encoder.o
 
+# Windows resource file
+SLIBOBJS-$(HAVE_GNU_WINDRES)           += avcodecres.o
+
 SKIPHEADERS                            += %_tablegen.h                  \
                                           %_tables.h                    \
                                           aac_tablegen_decl.h           \
@@ -819,7 +827,7 @@ SKIPHEADERS                            += %_tablegen.h                  \
 SKIPHEADERS-$(CONFIG_DXVA2)            += dxva2.h dxva2_internal.h
 SKIPHEADERS-$(CONFIG_LIBSCHROEDINGER)  += libschroedinger.h
 SKIPHEADERS-$(CONFIG_LIBUTVIDEO)       += libutvideo.h
-SKIPHEADERS-$(CONFIG_MPEG_XVMC_DECODER) += xvmc.h
+SKIPHEADERS-$(CONFIG_XVMC)             += xvmc.h
 SKIPHEADERS-$(CONFIG_VAAPI)            += vaapi_internal.h
 SKIPHEADERS-$(CONFIG_VDA)              += vda.h
 SKIPHEADERS-$(CONFIG_VDPAU)            += vdpau.h vdpau_internal.h
@@ -854,6 +862,7 @@ HOSTPROGS = aac_tablegen                                                \
 CLEANFILES = *_tables.c *_tables.h *_tablegen$(HOSTEXESUF)
 
 $(SUBDIR)dct-test$(EXESUF): $(SUBDIR)dctref.o $(SUBDIR)aandcttab.o
+$(SUBDIR)dv_tablegen$(HOSTEXESUF): $(SUBDIR)dvdata_host.o
 
 TRIG_TABLES  = cos cos_fixed sin
 TRIG_TABLES := $(TRIG_TABLES:%=$(SUBDIR)%_tables.c)
@@ -879,9 +888,9 @@ ifdef CONFIG_HARDCODED_TABLES
 $(SUBDIR)aacdec.o: $(SUBDIR)cbrt_tables.h
 $(SUBDIR)aacps.o: $(SUBDIR)aacps_tables.h
 $(SUBDIR)aactab.o: $(SUBDIR)aac_tables.h
-$(SUBDIR)dv.o: $(SUBDIR)dv_tables.h
+$(SUBDIR)dvenc.o: $(SUBDIR)dv_tables.h
 $(SUBDIR)sinewin.o: $(SUBDIR)sinewin_tables.h
-$(SUBDIR)mpegaudiodec.o: $(SUBDIR)mpegaudio_tables.h
+$(SUBDIR)mpegaudiodec_fixed.o: $(SUBDIR)mpegaudio_tables.h
 $(SUBDIR)mpegaudiodec_float.o: $(SUBDIR)mpegaudio_tables.h
 $(SUBDIR)motionpixels.o: $(SUBDIR)motionpixels_tables.h
 $(SUBDIR)pcm.o: $(SUBDIR)pcm_tables.h

libavcodec/a64multienc.c

@@ -40,9 +40,6 @@
 #define C64YRES 200
 
 typedef struct A64Context {
-    /* general variables */
-    AVFrame picture;
-
     /* variables for multicolor modes */
     AVLFG randctx;
     int mc_lifetime;
@@ -189,6 +186,7 @@ static void render_charset(AVCodecContext *avctx, uint8_t *charset,
 static av_cold int a64multi_close_encoder(AVCodecContext *avctx)
 {
     A64Context *c = avctx->priv_data;
+    av_frame_free(&avctx->coded_frame);
     av_free(c->mc_meta_charset);
     av_free(c->mc_best_cb);
     av_free(c->mc_charset);
@@ -197,7 +195,7 @@ static av_cold int a64multi_close_encoder(AVCodecContext *avctx)
     return 0;
 }
 
-static av_cold int a64multi_init_encoder(AVCodecContext *avctx)
+static av_cold int a64multi_encode_init(AVCodecContext *avctx)
 {
     A64Context *c = avctx->priv_data;
     int a;
@@ -240,8 +238,12 @@ static av_cold int a64multi_init_encoder(AVCodecContext *avctx)
     AV_WB32(avctx->extradata, c->mc_lifetime);
     AV_WB32(avctx->extradata + 16, INTERLACED);
 
-    avcodec_get_frame_defaults(&c->picture);
-    avctx->coded_frame            = &c->picture;
+    avctx->coded_frame = av_frame_alloc();
+    if (!avctx->coded_frame) {
+        a64multi_close_encoder(avctx);
+        return AVERROR(ENOMEM);
+    }
+
     avctx->coded_frame->pict_type = AV_PICTURE_TYPE_I;
     avctx->coded_frame->key_frame = 1;
     if (!avctx->codec_tag)
@@ -271,7 +273,7 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
                                  const AVFrame *pict, int *got_packet)
 {
     A64Context *c = avctx->priv_data;
-    AVFrame *const p = &c->picture;
+    AVFrame *const p = avctx->coded_frame;
 
     int frame;
     int x, y;
@@ -338,8 +340,8 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
             buf = pkt->data;
 
             /* calc optimal new charset + charmaps */
-            ff_init_elbg(meta, 32, 1000 * c->mc_lifetime, best_cb, CHARSET_CHARS, 50, charmap, &c->randctx);
-            ff_do_elbg  (meta, 32, 1000 * c->mc_lifetime, best_cb, CHARSET_CHARS, 50, charmap, &c->randctx);
+            avpriv_init_elbg(meta, 32, 1000 * c->mc_lifetime, best_cb, CHARSET_CHARS, 50, charmap, &c->randctx);
+            avpriv_do_elbg  (meta, 32, 1000 * c->mc_lifetime, best_cb, CHARSET_CHARS, 50, charmap, &c->randctx);
 
             /* create colorram map and a c64 readable charset */
             render_charset(avctx, charset, colram);
@@ -400,7 +402,7 @@ AVCodec ff_a64multi_encoder = {
     .type           = AVMEDIA_TYPE_VIDEO,
     .id             = AV_CODEC_ID_A64_MULTI,
     .priv_data_size = sizeof(A64Context),
-    .init           = a64multi_init_encoder,
+    .init           = a64multi_encode_init,
     .encode2        = a64multi_encode_frame,
     .close          = a64multi_close_encoder,
     .pix_fmts       = (const enum AVPixelFormat[]) {AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE},
@@ -414,7 +416,7 @@ AVCodec ff_a64multi5_encoder = {
     .type           = AVMEDIA_TYPE_VIDEO,
     .id             = AV_CODEC_ID_A64_MULTI5,
     .priv_data_size = sizeof(A64Context),
-    .init           = a64multi_init_encoder,
+    .init           = a64multi_encode_init,
     .encode2        = a64multi_encode_frame,
     .close          = a64multi_close_encoder,
     .pix_fmts       = (const enum AVPixelFormat[]) {AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE},

libavcodec/aac.h

@@ -81,7 +81,7 @@ enum BandType {
     INTENSITY_BT   = 15,    ///< Scalefactor data are intensity stereo positions.
 };
 
-#define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10)
+#define IS_CODEBOOK_UNSIGNED(x) (((x) - 1) & 10)
 
 enum ChannelPosition {
     AAC_CHANNEL_OFF   = 0,

libavcodec/aac_ac3_parser.c

@@ -20,6 +20,7 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
+#include "libavutil/channel_layout.h"
 #include "libavutil/common.h"
 #include "parser.h"
 #include "aac_ac3_parser.h"
@@ -82,14 +83,23 @@ get_next:
     if (avctx->codec_id != AV_CODEC_ID_AAC) {
         avctx->sample_rate = s->sample_rate;
 
-        /* allow downmixing to stereo (or mono for AC-3) */
-        if(avctx->request_channels > 0 &&
-                avctx->request_channels < s->channels &&
-                (avctx->request_channels <= 2 ||
-                (avctx->request_channels == 1 &&
-                (avctx->codec_id == AV_CODEC_ID_AC3 ||
-                 avctx->codec_id == AV_CODEC_ID_EAC3)))) {
-            avctx->channels = avctx->request_channels;
+        /* (E-)AC-3: allow downmixing to stereo or mono */
+#if FF_API_REQUEST_CHANNELS
+FF_DISABLE_DEPRECATION_WARNINGS
+        if (avctx->request_channels == 1)
+            avctx->request_channel_layout = AV_CH_LAYOUT_MONO;
+        else if (avctx->request_channels == 2)
+            avctx->request_channel_layout = AV_CH_LAYOUT_STEREO;
+FF_ENABLE_DEPRECATION_WARNINGS
+#endif
+        if (s->channels > 1 &&
+            avctx->request_channel_layout == AV_CH_LAYOUT_MONO) {
+            avctx->channels       = 1;
+            avctx->channel_layout = AV_CH_LAYOUT_MONO;
+        } else if (s->channels > 2 &&
+                   avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
+            avctx->channels       = 2;
+            avctx->channel_layout = AV_CH_LAYOUT_STEREO;
         } else {
             avctx->channels = s->channels;
             avctx->channel_layout = s->channel_layout;

libavcodec/aac_adtstoasc_bsf.c

@@ -87,6 +87,7 @@ static int aac_adtstoasc_filter(AVBitStreamFilterContext *bsfc,
             buf_size -= get_bits_count(&gb)/8;
             buf      += get_bits_count(&gb)/8;
         }
+        av_free(avctx->extradata);
         avctx->extradata_size = 2 + pce_size;
         avctx->extradata = av_mallocz(avctx->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
 

libavcodec/aacdec.c

@@ -74,6 +74,7 @@
  * N                    SinuSoidal Coding (Transient, Sinusoid, Noise)
  * Y                    Parametric Stereo
  * N                    Direct Stream Transfer
+ * Y                    Enhanced AAC Low Delay (ER AAC ELD)
  *
  * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
  *       - HE AAC v2 comprises LC AAC with Spectral Band Replication and
@@ -104,6 +105,7 @@
 #include <assert.h>
 #include <errno.h>
 #include <math.h>
+#include <stdint.h>
 #include <string.h>
 
 #if ARCH_ARM
@@ -194,6 +196,9 @@ static int frame_configure_elements(AVCodecContext *avctx)
 
     /* get output buffer */
     av_frame_unref(ac->frame);
+    if (!avctx->channels)
+        return 1;
+
     ac->frame->nb_samples = 2048;
     if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0)
         return ret;
@@ -532,6 +537,25 @@ static int set_default_channel_config(AVCodecContext *avctx,
     *tags = tags_per_config[channel_config];
     memcpy(layout_map, aac_channel_layout_map[channel_config - 1],
            *tags * sizeof(*layout_map));
+
+    /*
+     * AAC specification has 7.1(wide) as a default layout for 8-channel streams.
+     * However, at least Nero AAC encoder encodes 7.1 streams using the default
+     * channel config 7, mapping the side channels of the original audio stream
+     * to the second AAC_CHANNEL_FRONT pair in the AAC stream. Similarly, e.g. FAAD
+     * decodes the second AAC_CHANNEL_FRONT pair as side channels, therefore decoding
+     * the incorrect streams as if they were correct (and as the encoder intended).
+     *
+     * As actual intended 7.1(wide) streams are very rare, default to assuming a
+     * 7.1 layout was intended.
+     */
+    if (channel_config == 7 && avctx->strict_std_compliance < FF_COMPLIANCE_STRICT) {
+        av_log(avctx, AV_LOG_INFO, "Assuming an incorrectly encoded 7.1 channel layout"
+               " instead of a spec-compliant 7.1(wide) layout, use -strict %d to decode"
+               " according to the specification instead.\n", FF_COMPLIANCE_STRICT);
+        layout_map[2][2] = AAC_CHANNEL_SIDE;
+    }
+
     return 0;
 }
 
@@ -1112,7 +1136,6 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
     ff_mdct_init(&ac->mdct_ltp,   11, 0, -2.0 * 32768.0);
     // window initialization
     ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
-    ff_kbd_window_init(ff_aac_kbd_long_512,  4.0, 512);
     ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
     ff_init_ff_sine_windows(10);
     ff_init_ff_sine_windows( 9);
@@ -1225,13 +1248,14 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
         if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
             ics->swb_offset        =     ff_swb_offset_512[ac->oc[1].m4ac.sampling_index];
             ics->num_swb           =    ff_aac_num_swb_512[ac->oc[1].m4ac.sampling_index];
+            ics->tns_max_bands     =  ff_tns_max_bands_512[ac->oc[1].m4ac.sampling_index];
             if (!ics->num_swb || !ics->swb_offset)
                 return AVERROR_BUG;
         } else {
             ics->swb_offset        =    ff_swb_offset_1024[ac->oc[1].m4ac.sampling_index];
             ics->num_swb           =   ff_aac_num_swb_1024[ac->oc[1].m4ac.sampling_index];
+            ics->tns_max_bands     = ff_tns_max_bands_1024[ac->oc[1].m4ac.sampling_index];
         }
-        ics->tns_max_bands         = ff_tns_max_bands_1024[ac->oc[1].m4ac.sampling_index];
         if (aot != AOT_ER_AAC_ELD) {
             ics->predictor_present     = get_bits1(gb);
             ics->predictor_reset_group = 0;
@@ -1402,12 +1426,12 @@ static int decode_pulses(Pulse *pulse, GetBitContext *gb,
         return -1;
     pulse->pos[0]    = swb_offset[pulse_swb];
     pulse->pos[0]   += get_bits(gb, 5);
-    if (pulse->pos[0] > 1023)
+    if (pulse->pos[0] >= swb_offset[num_swb])
         return -1;
     pulse->amp[0]    = get_bits(gb, 4);
     for (i = 1; i < pulse->num_pulse; i++) {
         pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
-        if (pulse->pos[i] > 1023)
+        if (pulse->pos[i] >= swb_offset[num_swb])
             return -1;
         pulse->amp[i] = get_bits(gb, 4);
     }
@@ -2262,10 +2286,12 @@ static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
         } else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->channels == 1) {
             ac->oc[1].m4ac.sbr = 1;
             ac->oc[1].m4ac.ps = 1;
+            ac->avctx->profile = FF_PROFILE_AAC_HE_V2;
             output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
                              ac->oc[1].status, 1);
         } else {
             ac->oc[1].m4ac.sbr = 1;
+            ac->avctx->profile = FF_PROFILE_AAC_HE;
         }
         res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
         break;
@@ -2505,14 +2531,20 @@ static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
     float *in    = sce->coeffs;
     float *out   = sce->ret;
     float *saved = sce->saved;
-    const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_512 : ff_sine_512;
     float *buf  = ac->buf_mdct;
 
     // imdct
     ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
 
     // window overlapping
-    ac->fdsp.vector_fmul_window(out, saved, buf, lwindow_prev, 256);
+    if (ics->use_kb_window[1]) {
+        // AAC LD uses a low overlap sine window instead of a KBD window
+        memcpy(out, saved, 192 * sizeof(float));
+        ac->fdsp.vector_fmul_window(out + 192, saved + 192, buf, ff_sine_128, 64);
+        memcpy(                     out + 320, buf + 64, 192 * sizeof(float));
+    } else {
+        ac->fdsp.vector_fmul_window(out, saved, buf, ff_sine_512, 256);
+    }
 
     // buffer update
     memcpy(saved, buf + 256, 256 * sizeof(float));
@@ -2533,7 +2565,7 @@ static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
     // Inverse transform, mapped to the conventional IMDCT by
     // Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
     // "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks,"
-    // Audio, Language and Image Processing, 2008. ICALIP 2008. International Conference on
+    // International Conference on Audio, Language and Image Processing, ICALIP 2008.
     // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
     for (i = 0; i < n2; i+=2) {
         float temp;
@@ -2800,6 +2832,10 @@ static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
     if ((err = frame_configure_elements(avctx)) < 0)
         return err;
 
+    // The FF_PROFILE_AAC_* defines are all object_type - 1
+    // This may lead to an undefined profile being signaled
+    ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
+
     ac->tags_mapped = 0;
 
     if (chan_config < 0 || chan_config >= 8) {
@@ -2836,6 +2872,7 @@ static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
     spectral_to_sample(ac);
 
     ac->frame->nb_samples = samples;
+    ac->frame->sample_rate = avctx->sample_rate;
     *got_frame_ptr = 1;
 
     skip_bits_long(gb, get_bits_left(gb));
@@ -2869,6 +2906,10 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
     if ((err = frame_configure_elements(avctx)) < 0)
         goto fail;
 
+    // The FF_PROFILE_AAC_* defines are all object_type - 1
+    // This may lead to an undefined profile being signaled
+    ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
+
     ac->tags_mapped = 0;
     // parse
     while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
@@ -2966,22 +3007,6 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
 
     multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
     samples <<= multiplier;
-    /* for dual-mono audio (SCE + SCE) */
-    is_dmono = ac->dmono_mode && sce_count == 2 &&
-               ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
-
-    if (samples)
-        ac->frame->nb_samples = samples;
-    else
-        av_frame_unref(ac->frame);
-    *got_frame_ptr = !!samples;
-
-    if (is_dmono) {
-        if (ac->dmono_mode == 1)
-            ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
-        else if (ac->dmono_mode == 2)
-            ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
-    }
 
     if (ac->oc[1].status && audio_found) {
         avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
@@ -2995,6 +3020,25 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
         if (side && side_size>=4)
             AV_WL32(side, 2*AV_RL32(side));
     }
+
+    *got_frame_ptr = !!samples;
+    if (samples) {
+        ac->frame->nb_samples = samples;
+        ac->frame->sample_rate = avctx->sample_rate;
+    } else
+        av_frame_unref(ac->frame);
+    *got_frame_ptr = !!samples;
+
+    /* for dual-mono audio (SCE + SCE) */
+    is_dmono = ac->dmono_mode && sce_count == 2 &&
+               ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
+    if (is_dmono) {
+        if (ac->dmono_mode == 1)
+            ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
+        else if (ac->dmono_mode == 2)
+            ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
+    }
+
     return 0;
 fail:
     pop_output_configuration(ac);

libavcodec/aacps.c

@@ -322,14 +322,15 @@ static void hybrid2_re(float (*in)[2], float (*out)[32][2], const float filter[8
 }
 
 /** Split one subband into 6 subsubbands with a complex filter */
-static void hybrid6_cx(PSDSPContext *dsp, float (*in)[2], float (*out)[32][2], const float (*filter)[8][2], int len)
+static void hybrid6_cx(PSDSPContext *dsp, float (*in)[2], float (*out)[32][2],
+                       TABLE_CONST float (*filter)[8][2], int len)
 {
     int i;
     int N = 8;
     LOCAL_ALIGNED_16(float, temp, [8], [2]);
 
     for (i = 0; i < len; i++, in++) {
-        dsp->hybrid_analysis(temp, in, filter, 1, N);
+        dsp->hybrid_analysis(temp, in, (const float (*)[8][2]) filter, 1, N);
         out[0][i][0] = temp[6][0];
         out[0][i][1] = temp[6][1];
         out[1][i][0] = temp[7][0];
@@ -345,12 +346,14 @@ static void hybrid6_cx(PSDSPContext *dsp, float (*in)[2], float (*out)[32][2], c
     }
 }
 
-static void hybrid4_8_12_cx(PSDSPContext *dsp, float (*in)[2], float (*out)[32][2], const float (*filter)[8][2], int N, int len)
+static void hybrid4_8_12_cx(PSDSPContext *dsp,
+                            float (*in)[2], float (*out)[32][2],
+                            TABLE_CONST float (*filter)[8][2], int N, int len)
 {
     int i;
 
     for (i = 0; i < len; i++, in++) {
-        dsp->hybrid_analysis(out[0] + i, in, filter, 32, N);
+        dsp->hybrid_analysis(out[0] + i, in, (const float (*)[8][2]) filter, 32, N);
     }
 }
 
@@ -429,6 +432,7 @@ static void hybrid_synthesis(PSDSPContext *dsp, float out[2][38][64],
 #define DECAY_SLOPE      0.05f
 /// Number of frequency bands that can be addressed by the parameter index, b(k)
 static const int   NR_PAR_BANDS[]      = { 20, 34 };
+static const int   NR_IPDOPD_BANDS[]   = { 11, 17 };
 /// Number of frequency bands that can be addressed by the sub subband index, k
 static const int   NR_BANDS[]          = { 71, 91 };
 /// Start frequency band for the all-pass filter decay slope
@@ -685,7 +689,8 @@ static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[3
             memcpy(ap_delay[k][m],   ap_delay[k][m]+numQMFSlots,           5*sizeof(ap_delay[k][m][0]));
         }
         ps->dsp.decorrelate(out[k], delay[k] + PS_MAX_DELAY - 2, ap_delay[k],
-                            phi_fract[is34][k], Q_fract_allpass[is34][k],
+                            phi_fract[is34][k],
+                            (const float (*)[2]) Q_fract_allpass[is34][k],
                             transient_gain[b], g_decay_slope, nL - n0);
     }
     for (; k < SHORT_DELAY_BAND[is34]; k++) {
@@ -763,7 +768,7 @@ static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2
     int8_t (*ipd_mapped)[PS_MAX_NR_IIDICC] = ipd_mapped_buf;
     int8_t (*opd_mapped)[PS_MAX_NR_IIDICC] = opd_mapped_buf;
     const int8_t *k_to_i = is34 ? k_to_i_34 : k_to_i_20;
-    const float (*H_LUT)[8][4] = (PS_BASELINE || ps->icc_mode < 3) ? HA : HB;
+    TABLE_CONST float (*H_LUT)[8][4] = (PS_BASELINE || ps->icc_mode < 3) ? HA : HB;
 
     //Remapping
     if (ps->num_env_old) {
@@ -824,7 +829,7 @@ static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2
             h21 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][2];
             h22 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][3];
 
-            if (!PS_BASELINE && ps->enable_ipdopd && 2*b <= NR_PAR_BANDS[is34]) {
+            if (!PS_BASELINE && ps->enable_ipdopd && b < NR_IPDOPD_BANDS[is34]) {
                 //The spec say says to only run this smoother when enable_ipdopd
                 //is set but the reference decoder appears to run it constantly
                 float h11i, h12i, h21i, h22i;
@@ -914,7 +919,7 @@ int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float
         memset(ps->ap_delay + top, 0, (NR_ALLPASS_BANDS[is34] - top)*sizeof(ps->ap_delay[0]));
 
     hybrid_analysis(&ps->dsp, Lbuf, ps->in_buf, L, is34, len);
-    decorrelation(ps, Rbuf, Lbuf, is34);
+    decorrelation(ps, Rbuf, (const float (*)[32][2]) Lbuf, is34);
     stereo_processing(ps, Lbuf, Rbuf, is34);
     hybrid_synthesis(&ps->dsp, L, Lbuf, is34, len);
     hybrid_synthesis(&ps->dsp, R, Rbuf, is34, len);

libavcodec/aacps_tablegen.c

@@ -82,7 +82,7 @@ int main(void)
     write_float_3d_array(f34_2_4, 4, 8, 2);
     printf("};\n");
 
-    printf("static const DECLARE_ALIGNED(16, float, Q_fract_allpass)[2][50][3][2] = {\n");
+    printf("static TABLE_CONST DECLARE_ALIGNED(16, float, Q_fract_allpass)[2][50][3][2] = {\n");
     write_float_4d_array(Q_fract_allpass, 2, 50, 3, 2);
     printf("};\n");
     printf("static const DECLARE_ALIGNED(16, float, phi_fract)[2][50][2] = {\n");

libavcodec/aacps_tablegen.h

@@ -28,6 +28,7 @@
 
 #if CONFIG_HARDCODED_TABLES
 #define ps_tableinit()
+#define TABLE_CONST const
 #include "libavcodec/aacps_tables.h"
 #else
 #include "libavutil/common.h"
@@ -37,6 +38,7 @@
 #define NR_ALLPASS_BANDS20 30
 #define NR_ALLPASS_BANDS34 50
 #define PS_AP_LINKS 3
+#define TABLE_CONST
 static float pd_re_smooth[8*8*8];
 static float pd_im_smooth[8*8*8];
 static float HA[46][8][4];
@@ -45,7 +47,7 @@ static DECLARE_ALIGNED(16, float, f20_0_8) [ 8][8][2];
 static DECLARE_ALIGNED(16, float, f34_0_12)[12][8][2];
 static DECLARE_ALIGNED(16, float, f34_1_8) [ 8][8][2];
 static DECLARE_ALIGNED(16, float, f34_2_4) [ 4][8][2];
-static DECLARE_ALIGNED(16, float, Q_fract_allpass)[2][50][3][2];
+static TABLE_CONST DECLARE_ALIGNED(16, float, Q_fract_allpass)[2][50][3][2];
 static DECLARE_ALIGNED(16, float, phi_fract)[2][50][2];
 
 static const float g0_Q8[] = {

libavcodec/aacpsdsp.c

@@ -93,7 +93,7 @@ static void ps_hybrid_synthesis_deint_c(float out[2][38][64],
 
 static void ps_decorrelate_c(float (*out)[2], float (*delay)[2],
                              float (*ap_delay)[PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2],
-                             const float phi_fract[2], float (*Q_fract)[2],
+                             const float phi_fract[2], const float (*Q_fract)[2],
                              const float *transient_gain,
                              float g_decay_slope,
                              int len)

libavcodec/aacpsdsp.h

@@ -38,7 +38,7 @@ typedef struct PSDSPContext {
                                    int i, int len);
     void (*decorrelate)(float (*out)[2], float (*delay)[2],
                         float (*ap_delay)[PS_QMF_TIME_SLOTS+PS_MAX_AP_DELAY][2],
-                        const float phi_fract[2], float (*Q_fract)[2],
+                        const float phi_fract[2], const float (*Q_fract)[2],
                         const float *transient_gain,
                         float g_decay_slope,
                         int len);

libavcodec/aacsbr.c

@@ -932,6 +932,7 @@ static void read_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
         } else {
 #if 1
             *num_bits_left -= ff_ps_read_data(ac->avctx, gb, &sbr->ps, *num_bits_left);
+            ac->avctx->profile = FF_PROFILE_AAC_HE_V2;
 #else
             avpriv_report_missing_feature(ac->avctx, "Parametric Stereo");
             skip_bits_long(gb, *num_bits_left); // bs_fill_bits
@@ -1701,12 +1702,18 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
         sbr_qmf_analysis(&ac->fdsp, &sbr->mdct_ana, &sbr->dsp, ch ? R : L, sbr->data[ch].analysis_filterbank_samples,
                          (float*)sbr->qmf_filter_scratch,
                          sbr->data[ch].W, sbr->data[ch].Ypos);
-        sbr->c.sbr_lf_gen(ac, sbr, sbr->X_low, sbr->data[ch].W, sbr->data[ch].Ypos);
+        sbr->c.sbr_lf_gen(ac, sbr, sbr->X_low,
+                          (const float (*)[32][32][2]) sbr->data[ch].W,
+                          sbr->data[ch].Ypos);
         sbr->data[ch].Ypos ^= 1;
         if (sbr->start) {
-            sbr->c.sbr_hf_inverse_filter(&sbr->dsp, sbr->alpha0, sbr->alpha1, sbr->X_low, sbr->k[0]);
+            sbr->c.sbr_hf_inverse_filter(&sbr->dsp, sbr->alpha0, sbr->alpha1,
+                                         (const float (*)[40][2]) sbr->X_low, sbr->k[0]);
             sbr_chirp(sbr, &sbr->data[ch]);
-            sbr_hf_gen(ac, sbr, sbr->X_high, sbr->X_low, sbr->alpha0, sbr->alpha1,
+            sbr_hf_gen(ac, sbr, sbr->X_high,
+                       (const float (*)[40][2]) sbr->X_low,
+                       (const float (*)[2]) sbr->alpha0,
+                       (const float (*)[2]) sbr->alpha1,
                        sbr->data[ch].bw_array, sbr->data[ch].t_env,
                        sbr->data[ch].bs_num_env);
 
@@ -1716,16 +1723,17 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
                 sbr_env_estimate(sbr->e_curr, sbr->X_high, sbr, &sbr->data[ch]);
                 sbr_gain_calc(ac, sbr, &sbr->data[ch], sbr->data[ch].e_a);
                 sbr->c.sbr_hf_assemble(sbr->data[ch].Y[sbr->data[ch].Ypos],
-                                sbr->X_high, sbr, &sbr->data[ch],
+                                (const float (*)[40][2]) sbr->X_high,
+                                sbr, &sbr->data[ch],
                                 sbr->data[ch].e_a);
             }
         }
 
         /* synthesis */
         sbr->c.sbr_x_gen(sbr, sbr->X[ch],
-                  sbr->data[ch].Y[1-sbr->data[ch].Ypos],
-                  sbr->data[ch].Y[  sbr->data[ch].Ypos],
-                  sbr->X_low, ch);
+                  (const float (*)[64][2]) sbr->data[ch].Y[1-sbr->data[ch].Ypos],
+                  (const float (*)[64][2]) sbr->data[ch].Y[  sbr->data[ch].Ypos],
+                  (const float (*)[40][2]) sbr->X_low, ch);
     }
 
     if (ac->oc[1].m4ac.ps == 1) {

libavcodec/aactab.c

@@ -34,7 +34,6 @@
 #include <stdint.h>
 
 DECLARE_ALIGNED(32, float,  ff_aac_kbd_long_1024)[1024];
-DECLARE_ALIGNED(32, float,  ff_aac_kbd_long_512 )[512];
 DECLARE_ALIGNED(32, float,  ff_aac_kbd_short_128)[128];
 
 const uint8_t ff_aac_num_swb_1024[] = {
@@ -1237,6 +1236,10 @@ const uint8_t ff_tns_max_bands_1024[] = {
     31, 31, 34, 40, 42, 51, 46, 46, 42, 42, 42, 39, 39
 };
 
+const uint8_t ff_tns_max_bands_512[] = {
+    0, 0, 0, 31, 32, 37, 31, 31, 0, 0, 0, 0, 0
+};
+
 const uint8_t ff_tns_max_bands_128[] = {
     9, 9, 10, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14
 };

libavcodec/aactab.h

@@ -45,7 +45,6 @@
  * @{
  */
 DECLARE_ALIGNED(32, extern float,  ff_aac_kbd_long_1024)[1024];
-DECLARE_ALIGNED(32, extern float,  ff_aac_kbd_long_512 )[512];
 DECLARE_ALIGNED(32, extern float,  ff_aac_kbd_short_128)[128];
 const DECLARE_ALIGNED(32, extern float, ff_aac_eld_window)[1920];
 // @}
@@ -76,6 +75,7 @@ extern const uint16_t * const ff_swb_offset_512 [13];
 extern const uint16_t * const ff_swb_offset_128 [13];
 
 extern const uint8_t ff_tns_max_bands_1024[13];
+extern const uint8_t ff_tns_max_bands_512 [13];
 extern const uint8_t ff_tns_max_bands_128 [13];
 
 #endif /* AVCODEC_AACTAB_H */