wmaprodec: tighter check for num_vec_coeffs

Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit b21ba20cc8)

Conflicts:

	libavcodec/wmaprodec.c

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

.gitignore

@@ -1,17 +1,9 @@
 .config
 .version
-*.a
 *.o
 *.d
-*.def
-*.dll
 *.exe
 *.ho
-*.lib
-*.pc
-*.so
-*.so.*
-*.ver
 *-example
 *-test
 *_g
@@ -30,19 +22,20 @@ ffplay
 ffprobe
 ffserver
 avconv
-doc/avoptions_codec.texi
-doc/avoptions_format.texi
-doc/print_options
-doc/examples/decoding_encoding
-doc/examples/filtering_audio
-doc/examples/filtering_video
-doc/examples/metadata
-doc/examples/muxing
 libavcodec/*_tablegen
 libavcodec/*_tables.c
 libavcodec/*_tables.h
 libavcodec/codec_names.h
+libavcodec/libavcodec*
+libavcore/libavcore*
+libavdevice/libavdevice*
+libavfilter/libavfilter*
+libavformat/libavformat*
 libavutil/avconfig.h
+libavutil/libavutil*
+libpostproc/libpostproc*
+libswresample/libswresample*
+libswscale/libswscale*
 tests/audiogen
 tests/base64
 tests/data
@@ -53,7 +46,6 @@ tests/vsynth1
 tests/vsynth2
 tools/aviocat
 tools/cws2fws
-tools/ffeval
 tools/graph2dot
 tools/ismindex
 tools/lavfi-showfiltfmts

COPYING.LGPLv2.1

@@ -500,3 +500,5 @@ necessary.  Here is a sample; alter the names:
   Ty Coon, President of Vice
 
 That's all there is to it!
+
+

Changelog

@@ -3,47 +3,133 @@ releases are sorted from youngest to oldest.
 
 version next:
 
-version 0.11:
-Fixes:CVE-2012-2772, CVE-2012-2774, CVE-2012-2775, CVE-2012-2776, CVE-2012-2777,
-      CVE-2012-2779, CVE-2012-2782, CVE-2012-2783, CVE-2012-2784, CVE-2012-2785,
-      CVE-2012-2786, CVE-2012-2787, CVE-2012-2788, CVE-2012-2789, CVE-2012-2790,
-      CVE-2012-2791, CVE-2012-2792, CVE-2012-2793, CVE-2012-2794, CVE-2012-2795,
-      CVE-2012-2796, CVE-2012-2797, CVE-2012-2798, CVE-2012-2799, CVE-2012-2800,
-      CVE-2012-2801, CVE-2012-2802, CVE-2012-2803, CVE-2012-2804,
-- v408 Quicktime and Microsoft AYUV Uncompressed 4:4:4:4 encoder and decoder
-- setfield filter
-- CDXL demuxer and decoder
-- Apple ProRes encoder
-- ffprobe -count_packets and -count_frames options
-- Sun Rasterfile Encoder
-- ID3v2 attached pictures reading and writing
-- WMA Lossless decoder
-- bluray protocol
-- blackdetect filter
-- libutvideo encoder wrapper (--enable-libutvideo)
+
+version 0.10.8
+- kmvc: Clip pixel position to valid range
+- kmvc: use fixed sized arrays in the context
+- indeo: use a typedef for the mc function pointer
+- lavc: check for overflow in init_get_bits
+- mjpegdec: properly report unsupported disabled features
+- jpegls: return meaningful errors
+- jpegls: factorize return paths
+- jpegls: check the scan offset
+- wavpack: validate samples size parsed in wavpack_decode_block
+- ljpeg: use the correct number of components in yuv
+- mjpeg: Validate sampling factors
+- mjpegdec: validate parameters in mjpeg_decode_scan_progressive_ac
+- wavpack: check packet size early
+- wavpack: return meaningful errors
+- apetag: use int64_t for filesize
+- tiff: do not overread the source buffer
+- Prepare for 0.8.8 Release
+- smacker: fix an off by one in huff.length computation
+- smacker: check the return value of smacker_decode_tree
+- smacker: pad the extradata allocation
+- smacker: check frame size validity
+- vmdav: convert to bytestream2
+- 4xm: don't rely on get_buffer() initializing the frame.
+- 4xm: check the return value of read_huffman_tables().
+- 4xm: use the correct logging context
+- 4xm: reject frames not compatible with the declared version
+- 4xm: check bitstream_size boundary before using it
+- 4xm: do not overread the source buffer in decode_p_block
+- avfiltergraph: check for sws opts being non-NULL before using them
+- bmv: check for len being valid in bmv_decode_frame()
+- dfa: check for invalid access in decode_wdlt()
+- indeo3: check motion vectors
+- indeo3: fix data size check
+- indeo3: switch parsing the header to bytestream2
+- lavf: make sure stream probe data gets freed.
+- oggdec: fix faulty cleanup prototype
+- oma: Validate sample rates
+- qdm2: check that the FFT size is a power of 2
+- rv10: check that extradata is large enough
+- xmv: check audio track parameters validity
+- xmv: do not leak memory in the error paths in xmv_read_header()
+- aac: check the maximum number of channels
+- indeo3: fix off by one in MV validity check, Bug #503
+- id3v2: check for end of file while unescaping tags
+- wav: Always seek to an even offset, Bug #500, LP: #1174737
+- proresdec: support mixed interlaced/non-interlaced content
+
+
+version 0.10.6:
+
+- many bug fixes that where found with Coverity
+
+- The following CVE fixes where backported:
+  CVE-2012-2796, CVE-2012-2775, CVE-2012-2772, CVE-2012-2776,
+  CVE-2012-2779, CVE-2012-2787, CVE-2012-2794, CVE-2012-2800,
+  CVE-2012-2802, CVE-2012-2801, CVE-2012-2786, CVE-2012-2798,
+  CVE-2012-2793, CVE-2012-2789, CVE-2012-2788, CVE-2012-2790,
+  CVE-2012-2777, CVE-2012-2784
+
+- hundreads of other bug fixes, some possibly security relevant,
+  see the git log for details.
+
+
+version 0.10.5:
+
+- Several bugs and crashes have been fixed as well as build problems
+  with recent mingw64
+
+
+version 0.10.4:
+
+- Several bugs and crashes have been fixed
+  Note, CVE-2012-0851 and CVE-2011-3937 have been fixed in previous releases
+
+version 0.10.3:
+
+- Security fixes in the 4xm demuxer, avi demuxer, cook decoder,
+  mm demuxer, mpegvideo decoder, vqavideo decoder (CVE-2012-0947) and
+  xmv demuxer.
+
+- Several bugs and crashes have been fixed in the following codecs: AAC,
+  APE, H.263, H.264, Indeo 4, Mimic, MJPEG, Motion Pixels Video, RAW,
+  TTA, VC1, VQA, WMA Voice, vqavideo.
+
+- Several bugs and crashes have been fixed in the following formats:
+  ASF, ID3v2, MOV, xWMA
+
+- This release additionally updates the following codecs to the
+  bytestream2 API, and therefore benefit from additional overflow
+  checks: truemotion2, utvideo, vqavideo
+
+
+version 0.10.1
+- Several security fixes, many bugfixes affecting many formats and
+  codecs, the list below is not complete.
+
 - swapuv filter
-- bbox filter
-- XBM encoder and decoder
-- RealAudio Lossless decoder
-- ZeroCodec decoder
-- tile video filter
-- Metal Gear Solid: The Twin Snakes demuxer
-- OpenEXR image decoder
-- removelogo filter
-- drop support for ffmpeg without libavfilter
-- drawtext video filter: fontconfig support
-- ffmpeg -benchmark_all option
-- super2xsai filter ported from libmpcodecs
-- add libavresample audio conversion library for compatibility
-- MicroDVD decoder
-- Avid Meridien (AVUI) encoder and decoder
-- accept + prefix to -pix_fmt option to disable automatic conversions.
-- complete audio filtering in libavfilter and ffmpeg
-- add fps filter
-- audio split filter
-- vorbis parser
-- png parser
-- audio mix filter
+
+- Several bugs and crashes have been fixed in the following codecs: AAC,
+  AC-3, ADPCM, AMR (both NB and WB), ATRAC3, CAVC, Cook, camstudio, DCA,
+  DPCM, DSI CIN, DV, EA TGQ, FLAC, fraps, G.722 (both encoder and
+  decoder), H.264, huvffyuv, BB JV decoder, Indeo 3, KGV1, LCL, the
+  libx264 wrapper, MJPEG, mp3on4, Musepack, MPEG1/2, PNG, QDM2, Qt RLE,
+  ROQ, RV10, RV30/RV34/RV40, shorten, smacker, subrip, SVQ3, TIFF,
+  Truemotion2, TTA, VC1, VMware Screen codec, Vorbis, VP5, VP6, WMA,
+  Westwood SNDx, XXAN.
+
+- This release additionally updates the following codecs to the
+  bytestream2 API, and therefore benefit from additional overflow
+  checks: XXAN, ALG MM, TQG, SMC, Qt SMC, ROQ, PNG
+
+- Several bugs and crashes have been fixed in the following formats:
+  AIFF, ASF, DV, Matroska, NSV, MOV, MPEG-TS, Smacker, Sony OpenMG, RM,
+  SWF.
+
+- Libswscale has an potential overflow for large image size fixed.
+
+- The following APIs have been added:
+
+  avcodec_is_open()
+  avformat_get_riff_video_tags()
+  avformat_get_riff_audio_tags()
+
+  Please see the file doc/APIchanges and the Doxygen documentation for
+  further information.
 
 
 version 0.10:
@@ -787,7 +873,7 @@ version 0.4.5:
 - MPEG-4 vol header fixes (Jonathan Marsden <snmjbm at pacbell.net>)
 - ARM optimizations (Lionel Ulmer <lionel.ulmer at free.fr>).
 - Windows porting of file converter
-- added MJPEG raw format (input/output)
+- added MJPEG raw format (input/ouput)
 - added JPEG image format support (input/output)
 
 

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         = 0.11.1
+PROJECT_NUMBER         = 0.10.8
 
 # With the PROJECT_LOGO tag one can specify an logo or icon that is included
 # in the documentation. The maximum height of the logo should not exceed 55

LICENSE

@@ -20,12 +20,8 @@ Specifically, the GPL parts of FFmpeg are
 
 There are a handful of files under other licensing terms, namely:
 
-* The files libavcodec/jfdctfst.c, libavcodec/jfdctint_template.c and
-  libavcodec/jrevdct.c are taken from libjpeg, see the top of the files for
-  licensing details. Specifically note that you must credit the IJG in the
-  documentation accompanying your program if you only distribute executables.
-  You must also indicate any changes including additions and deletions to
-  those three files in the documentation.
+* The files libavcodec/jfdctfst.c, libavcodec/jfdctint.c, libavcodec/jrevdct.c
+  are taken from libjpeg, see the top of the files for licensing details.
 
 Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
 the configure parameter --enable-version3 will activate this licensing option

MAINTAINERS

@@ -4,7 +4,7 @@ FFmpeg maintainers
 Below is a list of the people maintaining different parts of the
 FFmpeg code.
 
-Please try to keep entries where you are the maintainer up to date!
+Please try to keep entries where you are the maintainer upto date!
 
 Names in () mean that the maintainer currently has no time to maintain the code.
 A CC after the name means that the maintainer prefers to be CC-ed on patches
@@ -159,7 +159,6 @@ Codecs:
   indeo5*                               Kostya Shishkov
   interplayvideo.c                      Mike Melanson
   ivi*                                  Kostya Shishkov
-  jacosub*                              Clément Bœsch
   jpeg_ls.c                             Kostya Shishkov
   jvdec.c                               Peter Ross
   kmvc.c                                Kostya Shishkov
@@ -172,7 +171,6 @@ Codecs:
   libschroedinger*                      David Conrad
   libspeexdec.c                         Justin Ruggles
   libtheoraenc.c                        David Conrad
-  libutvideo*                           Derek Buitenhuis
   libvorbis.c                           David Conrad
   libxavs.c                             Stefan Gehrer
   libx264.c                             Mans Rullgard, Jason Garrett-Glaser
@@ -245,7 +243,6 @@ Codecs:
   xan.c                                 Mike Melanson
   xl.c                                  Kostya Shishkov
   xvmc.c                                Ivan Kalvachev
-  zerocodec.c                           Derek Buitenhuis
   zmbv*                                 Kostya Shishkov
 
 Hardware acceleration:
@@ -316,7 +313,6 @@ Muxers/Demuxers:
   ipmovie.c                             Mike Melanson
   img2.c                                Michael Niedermayer
   iss.c                                 Stefan Gehrer
-  jacosub*                              Clément Bœsch
   jvdec.c                               Peter Ross
   libmodplug.c                          Clément Bœsch
   libnut.c                              Oded Shimon
@@ -374,7 +370,6 @@ Muxers/Demuxers:
   wv.c                                  Kostya Shishkov
 
 Protocols:
-  bluray.c                              Petri Hintukainen
   http.c                                Ronald S. Bultje
   mms*.c                                Ronald S. Bultje
   udp.c                                 Luca Abeni
@@ -400,8 +395,7 @@ x86                                     Michael Niedermayer
 Releases
 ========
 
-0.11                                    Michael Niedermayer
-0.10                                    Michael Niedermayer
+0.9                                     Michael Niedermayer
 
 
 

Makefile

@@ -19,7 +19,7 @@ PROGS      := $(PROGS-yes:%=%$(EXESUF))
 INSTPROGS   = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
 OBJS        = $(PROGS-yes:%=%.o) cmdutils.o
 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr base64
-HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
+HOSTPROGS  := $(TESTTOOLS:%=tests/%)
 TOOLS       = qt-faststart trasher
 TOOLS-$(CONFIG_ZLIB) += cws2fws
 
@@ -31,7 +31,6 @@ ALLMANPAGES = $(BASENAMES:%=%.1)
 FFLIBS-$(CONFIG_AVDEVICE) += avdevice
 FFLIBS-$(CONFIG_AVFILTER) += avfilter
 FFLIBS-$(CONFIG_AVFORMAT) += avformat
-FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
 FFLIBS-$(CONFIG_AVCODEC)  += avcodec
 FFLIBS-$(CONFIG_POSTPROC) += postproc
 FFLIBS-$(CONFIG_SWRESAMPLE)+= swresample
@@ -40,7 +39,6 @@ FFLIBS-$(CONFIG_SWSCALE)  += swscale
 FFLIBS := avutil
 
 DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd
-EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile
 
 SKIPHEADERS = cmdutils_common_opts.h
 
@@ -66,11 +64,9 @@ config.h: .config
 	@-printf '\nWARNING: $(?F) newer than config.h, rerun configure\n\n'
 	@-tput sgr0 2>/dev/null
 
-SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS      \
-               ARCH_HEADERS BUILT_HEADERS SKIPHEADERS                    \
-               ALTIVEC-OBJS ARMV5TE-OBJS ARMV6-OBJS ARMVFP-OBJS MMI-OBJS \
-               MMX-OBJS NEON-OBJS VIS-OBJS YASM-OBJS                     \
-               OBJS TESTOBJS
+SUBDIR_VARS := OBJS FFLIBS CLEANFILES DIRS TESTPROGS EXAMPLES SKIPHEADERS \
+               ALTIVEC-OBJS MMX-OBJS NEON-OBJS X86-OBJS YASM-OBJS-FFT YASM-OBJS \
+               HOSTPROGS BUILT_HEADERS TESTOBJS ARCH_HEADERS ARMV6-OBJS TOOLS
 
 define RESET
 $(1) :=
@@ -126,10 +122,9 @@ install-progs: install-progs-yes $(PROGS)
 	$(Q)mkdir -p "$(BINDIR)"
 	$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"
 
-install-data: $(DATA_FILES) $(EXAMPLES_FILES)
-	$(Q)mkdir -p "$(DATADIR)/examples"
+install-data: $(DATA_FILES)
+	$(Q)mkdir -p "$(DATADIR)"
 	$(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)"
-	$(INSTALL) -m 644 $(EXAMPLES_FILES) "$(DATADIR)/examples"
 
 uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data
 
@@ -142,7 +137,6 @@ uninstall-data:
 clean::
 	$(RM) $(ALLPROGS) $(ALLPROGS_G)
 	$(RM) $(CLEANSUFFIXES)
-	$(RM) $(TOOLS)
 	$(RM) $(CLEANSUFFIXES:%=tools/%)
 	$(RM) coverage.info
 	$(RM) -r coverage-html
@@ -179,5 +173,5 @@ $(sort $(OBJDIRS)):
 # so this saves some time on slow systems.
 .SUFFIXES:
 
-.PHONY: all all-yes alltools check *clean config install*
+.PHONY: all all-yes alltools check *clean config examples install*
 .PHONY: testprogs uninstall*

README

@@ -4,15 +4,9 @@ FFmpeg README
 1) Documentation
 ----------------
 
-* Read the documentation in the doc/ directory in git.
-  You can also view it online at http://ffmpeg.org/documentation.html
+* Read the documentation in the doc/ directory.
 
 2) Licensing
 ------------
 
 * See the LICENSE file.
-
-3) Build and Install
---------------------
-
-* See the INSTALL file.

RELEASE

@@ -1 +1 @@
-0.11.1
+0.10.8

VERSION

@@ -1 +1 @@
-0.11.1
+0.10.8

arch.mak

@@ -1,13 +0,0 @@
-OBJS-$(HAVE_ARMV5TE) += $(ARMV5TE-OBJS) $(ARMV5TE-OBJS-yes)
-OBJS-$(HAVE_ARMV6)   += $(ARMV6-OBJS)   $(ARMV6-OBJS-yes)
-OBJS-$(HAVE_ARMVFP)  += $(ARMVFP-OBJS)  $(ARMVFP-OBJS-yes)
-OBJS-$(HAVE_NEON)    += $(NEON-OBJS)    $(NEON-OBJS-yes)
-
-OBJS-$(HAVE_MMI)     += $(MMI-OBJS)     $(MMI-OBJS-yes)
-
-OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
-
-OBJS-$(HAVE_VIS)     += $(VIS-OBJS)     $(VIS-OBJS-yes)
-
-OBJS-$(HAVE_MMX)     += $(MMX-OBJS)     $(MMX-OBJS-yes)
-OBJS-$(HAVE_YASM)    += $(YASM-OBJS)    $(YASM-OBJS-yes)

cmdutils.c

@@ -32,13 +32,11 @@
 #include "libavformat/avformat.h"
 #include "libavfilter/avfilter.h"
 #include "libavdevice/avdevice.h"
-#include "libavresample/avresample.h"
 #include "libswscale/swscale.h"
 #include "libswresample/swresample.h"
 #if CONFIG_POSTPROC
 #include "libpostproc/postprocess.h"
 #endif
-#include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
 #include "libavutil/mathematics.h"
 #include "libavutil/parseutils.h"
@@ -56,10 +54,9 @@
 #endif
 
 struct SwsContext *sws_opts;
-SwrContext *swr_opts;
 AVDictionary *format_opts, *codec_opts;
 
-const int this_year = 2012;
+const int this_year = 2013;
 
 static FILE *report_file;
 
@@ -69,7 +66,6 @@ void init_opts(void)
     sws_opts = sws_getContext(16, 16, 0, 16, 16, 0, SWS_BICUBIC,
                               NULL, NULL, NULL);
 #endif
-    swr_opts = swr_alloc();
 }
 
 void uninit_opts(void)
@@ -78,7 +74,6 @@ void uninit_opts(void)
     sws_freeContext(sws_opts);
     sws_opts = NULL;
 #endif
-    swr_free(&swr_opts);
     av_dict_free(&format_opts);
     av_dict_free(&codec_opts);
 }
@@ -255,12 +250,14 @@ int parse_option(void *optctx, const char *opt, const char *arg,
     if (!po->name && opt[0] == 'n' && opt[1] == 'o') {
         /* handle 'no' bool option */
         po = find_option(options, opt + 2);
-        if ((po->name && (po->flags & OPT_BOOL)))
-            bool_val = 0;
+        if (!(po->name && (po->flags & OPT_BOOL)))
+            goto unknown_opt;
+        bool_val = 0;
     }
     if (!po->name)
         po = find_option(options, "default");
     if (!po->name) {
+unknown_opt:
         av_log(NULL, AV_LOG_ERROR, "Unrecognized option '%s'\n", opt);
         return AVERROR(EINVAL);
     }
@@ -345,8 +342,11 @@ void parse_options(void *optctx, int argc, char **argv, const OptionDef *options
     }
 }
 
-int locate_option(int argc, char **argv, const OptionDef *options,
-                  const char *optname)
+/*
+ * Return index of option opt in argv or 0 if not found.
+ */
+static int locate_option(int argc, char **argv, const OptionDef *options,
+                         const char *optname)
 {
     const OptionDef *po;
     int i;
@@ -420,10 +420,10 @@ void parse_loglevel(int argc, char **argv, const OptionDef *options)
 #define FLAGS(o) ((o)->type == AV_OPT_TYPE_FLAGS) ? AV_DICT_APPEND : 0
 int opt_default(const char *opt, const char *arg)
 {
-    const AVOption *oc, *of, *os, *oswr = NULL;
+    const AVOption *oc, *of, *os;
     char opt_stripped[128];
     const char *p;
-    const AVClass *cc = avcodec_get_class(), *fc = avformat_get_class(), *sc, *swr_class;
+    const AVClass *cc = avcodec_get_class(), *fc = avformat_get_class(), *sc;
 
     if (!(p = strchr(opt, ':')))
         p = opt + strlen(opt);
@@ -449,17 +449,8 @@ int opt_default(const char *opt, const char *arg)
         }
     }
 #endif
-    swr_class = swr_get_class();
-    if (!oc && !of && !os && (oswr = av_opt_find(&swr_class, opt, NULL, 0,
-                          AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
-        int ret = av_opt_set(swr_opts, opt, arg, 0);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Error setting option %s.\n", opt);
-            return ret;
-        }
-    }
 
-    if (oc || of || os || oswr)
+    if (oc || of || os)
         return 0;
     av_log(NULL, AV_LOG_ERROR, "Unrecognized option '%s'\n", opt);
     return AVERROR_OPTION_NOT_FOUND;
@@ -546,18 +537,6 @@ int opt_max_alloc(const char *opt, const char *arg)
     return 0;
 }
 
-int opt_cpuflags(const char *opt, const char *arg)
-{
-    int ret;
-    unsigned flags = av_get_cpu_flags();
-
-    if ((ret = av_parse_cpu_caps(&flags, arg)) < 0)
-        return ret;
-
-    av_force_cpu_flags(flags);
-    return 0;
-}
-
 int opt_codec_debug(const char *opt, const char *arg)
 {
     av_log_set_level(AV_LOG_DEBUG);
@@ -599,8 +578,7 @@ static int warned_cfg = 0;
         const char *indent = flags & INDENT? "  " : "";                 \
         if (flags & SHOW_VERSION) {                                     \
             unsigned int version = libname##_version();                 \
-            av_log(NULL, level,                                         \
-                   "%slib%-11s %2d.%3d.%3d / %2d.%3d.%3d\n",            \
+            av_log(NULL, level, "%slib%-11s %2d.%3d.%3d / %2d.%3d.%3d\n",\
                    indent, #libname,                                    \
                    LIB##LIBNAME##_VERSION_MAJOR,                        \
                    LIB##LIBNAME##_VERSION_MINOR,                        \
@@ -629,7 +607,6 @@ static void print_all_libs_info(int flags, int level)
     PRINT_LIB_INFO(avformat, AVFORMAT, flags, level);
     PRINT_LIB_INFO(avdevice, AVDEVICE, flags, level);
     PRINT_LIB_INFO(avfilter, AVFILTER, flags, level);
-//    PRINT_LIB_INFO(avresample, AVRESAMPLE, flags, level);
     PRINT_LIB_INFO(swscale,  SWSCALE,  flags, level);
     PRINT_LIB_INFO(swresample,SWRESAMPLE,  flags, level);
 #if CONFIG_POSTPROC
@@ -827,9 +804,9 @@ int opt_codecs(const char *opt, const char *arg)
                 decode = encode = cap = 0;
             }
             if (p2 && strcmp(p->name, p2->name) == 0) {
-                if (av_codec_is_decoder(p))
+                if (p->decode)
                     decode = 1;
-                if (av_codec_is_encoder(p))
+                if (p->encode || p->encode2)
                     encode = 1;
                 cap |= p->capabilities;
             }
@@ -875,16 +852,20 @@ int opt_bsfs(const char *opt, const char *arg)
 
 int opt_protocols(const char *opt, const char *arg)
 {
-    void *opaque = NULL;
-    const char *name;
+    URLProtocol *up=NULL;
 
     printf("Supported file protocols:\n"
-           "Input:\n");
-    while ((name = avio_enum_protocols(&opaque, 0)))
-        printf("%s\n", name);
-    printf("Output:\n");
-    while ((name = avio_enum_protocols(&opaque, 1)))
-        printf("%s\n", name);
+           "I.. = Input  supported\n"
+           ".O. = Output supported\n"
+           "..S = Seek   supported\n"
+           "FLAGS NAME\n"
+           "----- \n");
+    while((up = av_protocol_next(up)))
+        printf("%c%c%c   %s\n",
+               up->url_read  ? 'I' : '.',
+               up->url_write ? 'O' : '.',
+               up->url_seek  ? 'S' : '.',
+               up->name);
     return 0;
 }
 
@@ -1052,7 +1033,7 @@ FILE *get_preset_file(char *filename, size_t filename_size,
             if (!f && codec_name) {
                 snprintf(filename, filename_size,
                          "%s%s/%s-%s.ffpreset",
-                         base[i], i != 1 ? "" : "/.ffmpeg", codec_name,
+                         base[i],  i != 1 ? "" : "/.ffmpeg", codec_name,
                          preset_name);
                 f = fopen(filename, "r");
             }
@@ -1076,7 +1057,7 @@ int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec)
         case 's': type = AVMEDIA_TYPE_SUBTITLE;   break;
         case 'd': type = AVMEDIA_TYPE_DATA;       break;
         case 't': type = AVMEDIA_TYPE_ATTACHMENT; break;
-        default:  av_assert0(0);
+        default: abort(); // never reached, silence warning
         }
         if (type != st->codec->codec_type)
             return 0;
@@ -1109,12 +1090,6 @@ int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec)
                     return 1;
         }
         return 0;
-    } else if (*spec == '#') {
-        int sid;
-        char *endptr;
-        sid = strtol(spec + 1, &endptr, 0);
-        if (!*endptr)
-            return st->id == sid;
     } else if (!*spec) /* empty specifier, matches everything */
         return 1;
 

cmdutils.h

@@ -51,7 +51,6 @@ extern const int this_year;
 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
 extern struct SwsContext *sws_opts;
-extern struct SwrContext *swr_opts;
 extern AVDictionary *format_opts, *codec_opts;
 
 /**
@@ -86,8 +85,6 @@ int opt_report(const char *opt);
 
 int opt_max_alloc(const char *opt, const char *arg);
 
-int opt_cpuflags(const char *opt, const char *arg);
-
 int opt_codec_debug(const char *opt, const char *arg);
 
 /**
@@ -207,12 +204,6 @@ int parse_option(void *optctx, const char *opt, const char *arg,
  */
 void parse_loglevel(int argc, char **argv, const OptionDef *options);
 
-/**
- * Return index of option opt in argv or 0 if not found.
- */
-int locate_option(int argc, char **argv, const OptionDef *options,
-                  const char *optname);
-
 /**
  * Check if the given stream matches a stream specifier.
  *
@@ -358,7 +349,7 @@ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
  * at configuration time or in a "ffpresets" folder along the executable
  * on win32, in that order. If no such file is found and
  * codec_name is defined, then search for a file named
- * codec_name-preset_name.avpreset in the above-mentioned directories.
+ * codec_name-preset_name.ffpreset in the above-mentioned directories.
  *
  * @param filename buffer where the name of the found filename is written
  * @param filename_size size in bytes of the filename buffer

cmdutils_common_opts.h

@@ -14,7 +14,5 @@
     { "loglevel", HAS_ARG, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },
     { "v", HAS_ARG, {(void*)opt_loglevel}, "set libav* logging level", "loglevel" },
     { "debug", HAS_ARG, {(void*)opt_codec_debug}, "set debug flags", "flags" },
-    { "fdebug", HAS_ARG, {(void*)opt_codec_debug}, "set debug flags", "flags" },
     { "report", 0, {(void*)opt_report}, "generate a report" },
     { "max_alloc", HAS_ARG, {(void*)opt_max_alloc}, "set maximum size of a single allocated block", "bytes" },
-    { "cpuflags", HAS_ARG | OPT_EXPERT, {(void*)opt_cpuflags}, "force specific cpu flags", "flags" },

common.mak

@@ -20,7 +20,7 @@ $(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
 $(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
 endif
 
-ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample
+ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale swresample
 
 # NASM requires -I path terminated with /
 IFLAGS     := -I. -I$(SRC_PATH)/
@@ -73,7 +73,7 @@ COMPILE_S = $(call COMPILE,AS)
 $(OBJS):
 endif
 
-include $(SRC_PATH)/arch.mak
+OBJS-$(HAVE_MMX) +=  $(MMX-OBJS-yes)
 
 OBJS      += $(OBJS-yes)
 FFLIBS    := $(FFLIBS-yes) $(FFLIBS)
@@ -115,6 +115,15 @@ OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOSTOBJS) $(TESTOBJS))
 
 CLEANSUFFIXES     = *.d *.o *~ *.ho *.map *.ver *.gcno *.gcda
 DISTCLEANSUFFIXES = *.pc
-LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
+LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a *.exp
+
+define RULES
+clean::
+	$(RM) $(OBJS) $(OBJS:.o=.d)
+	$(RM) $(HOSTPROGS)
+	$(RM) $(TOOLS)
+endef
+
+$(eval $(RULES))
 
 -include $(wildcard $(OBJS:.o=.d) $(TESTOBJS:.o=.d))

doc/Makefile

@@ -8,7 +8,6 @@ HTMLPAGES   = $(PROGS-yes:%=doc/%.html)                                 \
               doc/git-howto.html                                        \
               doc/libavfilter.html                                      \
               doc/platform.html                                         \
-              doc/syntax.html                                           \
 
 TXTPAGES    = doc/fate.txt                                              \
 
@@ -29,30 +28,22 @@ doc/%.txt: doc/%.texi
 	$(Q)$(TEXIDEP)
 	$(M)makeinfo --force --no-headers -o $@ $< 2>/dev/null
 
-doc/print_options.o: libavformat/options_table.h libavcodec/options_table.h
-
-GENTEXI  = format codec
-GENTEXI := $(GENTEXI:%=doc/avoptions_%.texi)
-
-$(GENTEXI): TAG = GENTEXI
-$(GENTEXI): doc/avoptions_%.texi: doc/print_options
-	$(M)doc/print_options $* > $@
-
 doc/%.html: TAG = HTML
-doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
+doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init
 	$(Q)$(TEXIDEP)
-	$(M)texi2html -I doc -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
+	$(M)texi2html -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
 
 doc/%.pod: TAG = POD
-doc/%.pod: doc/%.texi $(GENTEXI)
+doc/%.pod: doc/%.texi
 	$(Q)$(TEXIDEP)
-	$(M)$(SRC_PATH)/doc/texi2pod.pl -Idoc $< $@
+	$(M)$(SRC_PATH)/doc/texi2pod.pl $< $@
 
 doc/%.1: TAG = MAN
-doc/%.1: doc/%.pod $(GENTEXI)
+doc/%.1: doc/%.pod
 	$(M)pod2man --section=1 --center=" " --release=" " $< > $@
 
-$(DOCS): | doc/
+$(DOCS): | doc
+OBJDIRS += doc
 
 install-progs-$(CONFIG_DOC): install-man
 
@@ -66,7 +57,7 @@ uninstall-man:
 	$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
 
 clean::
-	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 $(CLEANSUFFIXES:%=doc/%) doc/avoptions_*.texi
+	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 $(CLEANSUFFIXES:%=doc/%)
 
 -include $(wildcard $(DOCS:%=%.d))
 

doc/RELEASE_NOTES

@@ -1,11 +1,13 @@
 Release Notes
 =============
 
-* 0.11 "Happiness"  May, 2012
+* 0.10 "Freedom"  January, 2012
 
 
 General notes
 -------------
+This release is binary compatible with 0.8 and 0.9.
+
 See the Changelog file for a list of significant changes. Note, there
 are many more new features and bugfixes than whats listed there.
 
@@ -14,3 +16,34 @@ accepted. If you are experiencing issues with any formally released version of
 FFmpeg, please try git master to check if the issue still exists. If it does,
 make your report against the development code following the usual bug reporting
 guidelines.
+
+
+API changes
+-----------
+
+A number of additional APIs have been introduced and some existing
+functions have been deprecated and are scheduled for removal in the next
+release. Significant API changes include:
+
+* new audio decoding API which decodes from an AVPacket to an AVFrame and
+is able to use AVCodecContext.get_buffer() in the similar way as video decoding.
+
+* new audio encoding API which encodes from an AVFrame to an AVPacket, thus
+allowing it to properly output timing information and side data.
+
+Please see the git history and the file doc/APIchanges for details.
+
+
+Other notable changes
+---------------------
+
+Libavcodec and libavformat built as shared libraries now hide non-public
+symbols. This will break applications using those symbols. Possible solutions
+are, in order of preference:
+1) Try finding a way of accomplishing the same with public API.
+2) If there is no corresponding public API, but you think there should be,
+post a request on the developer mailing list or IRC channel.
+3) Finally if your program needs access to FFmpeg / libavcodec / libavformat
+internals for some special reason then the best solution is to link statically.
+
+Please see the Changelog file and git history for a more detailed list of changes.

doc/avtools-common-opts.texi

@@ -41,8 +41,6 @@ streams of this type.
 @item p:@var{program_id}[:@var{stream_index}]
 If @var{stream_index} is given, then matches stream number @var{stream_index} in
 program with id @var{program_id}. Otherwise matches all streams in this program.
-@item #@var{stream_id}
-Matches the stream by format-specific ID.
 @end table
 @section Generic options
 
@@ -136,15 +134,6 @@ It also implies @code{-loglevel verbose}.
 Note: setting the environment variable @code{FFREPORT} to any value has the
 same effect.
 
-@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.
-@example
-ffmpeg -cpuflags -sse+mmx ...
-ffmpeg -cpuflags mmx ...
-ffmpeg -cpuflags 0 ...
-@end example
-
 @end table
 
 @section AVOptions
@@ -177,6 +166,3 @@ use @option{-option 0}/@option{-option 1}.
 
 Note2 old undocumented way of specifying per-stream AVOptions by prepending
 v/a/s to the options name is now obsolete and will be removed soon.
-
-@include avoptions_codec.texi
-@include avoptions_format.texi

doc/bitstream_filters.texi

@@ -71,7 +71,7 @@ stream (carrying the AVI1 header ID and lacking a DHT segment) to
 produce fully qualified JPEG images.
 
 @example
-ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg
+ffmpeg -i mjpeg-movie.avi -c:v copy -vbsf mjpeg2jpeg frame_%d.jpg
 exiftran -i -9 frame*.jpg
 ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
 @end example

doc/developer.texi

@@ -14,13 +14,12 @@
 @section API
 @itemize @bullet
 @item libavcodec is the library containing the codecs (both encoding and
-decoding). Look at @file{doc/examples/decoding_encoding.c} to see how to use
-it.
+decoding). Look at @file{libavcodec/apiexample.c} to see how to use it.
 
 @item libavformat is the library containing the file format handling (mux and
 demux code for several formats). Look at @file{ffplay.c} to use it in a
-player. See @file{doc/examples/muxing.c} to use it to generate audio or video
-streams.
+player. See @file{libavformat/output-example.c} to use it to generate
+audio or video streams.
 
 @end itemize
 
@@ -188,10 +187,8 @@ the following snippet into your @file{.vimrc}:
 set expandtab
 set shiftwidth=4
 set softtabstop=4
-set cindent
-set cinoptions=(0
-" allow tabs in Makefiles
-autocmd FileType make set noexpandtab shiftwidth=8 softtabstop=8
+" Allow tabs in Makefiles.
+autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
 " Trailing whitespace and tabs are forbidden, so highlight them.
 highlight ForbiddenWhitespace ctermbg=red guibg=red
 match ForbiddenWhitespace /\s\+$\|\t/
@@ -201,16 +198,10 @@ autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
 
 For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
 @example
-(c-add-style "ffmpeg"
-             '("k&r"
-               (c-basic-offset . 4)
-               (indent-tabs-mode nil)
-               (show-trailing-whitespace t)
-               (c-offsets-alist
-                (statement-cont . (c-lineup-assignments +)))
-               )
-             )
-(setq c-default-style "ffmpeg")
+(setq c-default-style "k&r")
+(setq-default c-basic-offset 4)
+(setq-default indent-tabs-mode nil)
+(setq-default show-trailing-whitespace t)
 @end example
 
 @section Development Policy

doc/eval.texi

@@ -106,18 +106,6 @@ the evaluation of @var{y}, return 0 otherwise.
 @item ifnot(x, y)
 Evaluate @var{x}, and if the result is zero return the result of the
 evaluation of @var{y}, return 0 otherwise.
-
-@item taylor(expr, x) taylor(expr, x, id)
-Evaluate a taylor series at x.
-expr represents the LD(id)-th derivates of f(x) at 0. If id is not specified
-then 0 is assumed.
-note, when you have the derivatives at y instead of 0
-taylor(expr, x-y) can be used
-When the series does not converge the results are undefined.
-
-@item root(expr, max)
-Finds x where f(x)=0 in the interval 0..max.
-f() must be continuous or the result is undefined.
 @end table
 
 The following constants are available:

doc/examples/Makefile

@@ -1,28 +1,17 @@
-# use pkg-config for getting CFLAGS and LDLIBS
-FFMPEG_LIBS=    libavdevice                        \
-                libavformat                        \
-                libavfilter                        \
-                libavcodec                         \
-                libavresample                      \
-                libswresample                      \
-                libswscale                         \
-                libavutil                          \
+# use pkg-config for getting CFLAGS abd LDFLAGS
+FFMPEG_LIBS=libavdevice libavformat libavfilter libavcodec libswscale libavutil
+CFLAGS+=$(shell pkg-config  --cflags $(FFMPEG_LIBS))
+LDFLAGS+=$(shell pkg-config --libs $(FFMPEG_LIBS))
 
-CFLAGS += -Wall -O2 -g
-CFLAGS += $(shell pkg-config --cflags $(FFMPEG_LIBS))
-LDLIBS += $(shell pkg-config --libs $(FFMPEG_LIBS))
-
-EXAMPLES=       decoding_encoding                  \
-                filtering_video                    \
-                filtering_audio                    \
-                metadata                           \
-                muxing                             \
+EXAMPLES=decoding_encoding filtering metadata muxing
 
 OBJS=$(addsuffix .o,$(EXAMPLES))
 
-# the following examples make explicit use of the math library
-decoding_encoding: LDLIBS += -lm
-muxing:            LDLIBS += -lm
+%: %.o
+	$(CC) $< $(LDFLAGS) -o $@
+
+%.o: %.c
+	$(CC) $< $(CFLAGS) -c -o $@
 
 .phony: all clean
 

doc/examples/decoding_encoding.c

@@ -29,13 +29,11 @@
  * format handling
  */
 
-#include <math.h>
-
-#include <libavutil/imgutils.h>
-#include <libavutil/opt.h>
-#include <libavcodec/avcodec.h>
-#include <libavutil/mathematics.h>
-#include <libavutil/samplefmt.h>
+#include "libavutil/imgutils.h"
+#include "libavutil/opt.h"
+#include "libavcodec/avcodec.h"
+#include "libavutil/mathematics.h"
+#include "libavutil/samplefmt.h"
 
 #define INBUF_SIZE 4096
 #define AUDIO_INBUF_SIZE 20480
@@ -54,7 +52,7 @@ static void audio_encode_example(const char *filename)
     float t, tincr;
     uint8_t *outbuf;
 
-    printf("Encode audio file %s\n", filename);
+    printf("Audio encoding\n");
 
     /* find the MP2 encoder */
     codec = avcodec_find_encoder(CODEC_ID_MP2);
@@ -72,7 +70,7 @@ static void audio_encode_example(const char *filename)
     c->sample_fmt = AV_SAMPLE_FMT_S16;
 
     /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
+    if (avcodec_open(c, codec) < 0) {
         fprintf(stderr, "could not open codec\n");
         exit(1);
     }
@@ -125,7 +123,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
 
     av_init_packet(&avpkt);
 
-    printf("Decode audio file %s\n", filename);
+    printf("Audio decoding\n");
 
     /* find the mpeg audio decoder */
     codec = avcodec_find_decoder(CODEC_ID_MP2);
@@ -137,7 +135,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
     c = avcodec_alloc_context3(codec);
 
     /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
+    if (avcodec_open(c, codec) < 0) {
         fprintf(stderr, "could not open codec\n");
         exit(1);
     }
@@ -213,13 +211,12 @@ static void video_encode_example(const char *filename, int codec_id)
 {
     AVCodec *codec;
     AVCodecContext *c= NULL;
-    int i, out_size, x, y, outbuf_size;
+    int i, out_size, size, x, y, outbuf_size;
     FILE *f;
     AVFrame *picture;
     uint8_t *outbuf;
-    int had_output=0;
 
-    printf("Encode video file %s\n", filename);
+    printf("Video encoding\n");
 
     /* find the mpeg1 video encoder */
     codec = avcodec_find_encoder(codec_id);
@@ -246,7 +243,7 @@ static void video_encode_example(const char *filename, int codec_id)
         av_opt_set(c->priv_data, "preset", "slow", 0);
 
     /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
+    if (avcodec_open(c, codec) < 0) {
         fprintf(stderr, "could not open codec\n");
         exit(1);
     }
@@ -258,7 +255,7 @@ static void video_encode_example(const char *filename, int codec_id)
     }
 
     /* alloc image and output buffer */
-    outbuf_size = 100000 + 12*c->width*c->height;
+    outbuf_size = 100000;
     outbuf = malloc(outbuf_size);
 
     /* the image can be allocated by any means and av_image_alloc() is
@@ -287,17 +284,15 @@ static void video_encode_example(const char *filename, int codec_id)
 
         /* encode the image */
         out_size = avcodec_encode_video(c, outbuf, outbuf_size, picture);
-        had_output |= out_size;
         printf("encoding frame %3d (size=%5d)\n", i, out_size);
         fwrite(outbuf, 1, out_size, f);
     }
 
     /* get the delayed frames */
-    for(; out_size || !had_output; i++) {
+    for(; out_size; i++) {
         fflush(stdout);
 
         out_size = avcodec_encode_video(c, outbuf, outbuf_size, NULL);
-        had_output |= out_size;
         printf("write frame %3d (size=%5d)\n", i, out_size);
         fwrite(outbuf, 1, out_size, f);
     }
@@ -351,7 +346,7 @@ static void video_decode_example(const char *outfilename, const char *filename)
     /* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
     memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
 
-    printf("Decode video file %s\n", filename);
+    printf("Video decoding\n");
 
     /* find the mpeg1 video decoder */
     codec = avcodec_find_decoder(CODEC_ID_MPEG1VIDEO);
@@ -371,7 +366,7 @@ static void video_decode_example(const char *outfilename, const char *filename)
        available in the bitstream. */
 
     /* open it */
-    if (avcodec_open2(c, codec, NULL) < 0) {
+    if (avcodec_open(c, codec) < 0) {
         fprintf(stderr, "could not open codec\n");
         exit(1);
     }
@@ -458,6 +453,9 @@ int main(int argc, char **argv)
 {
     const char *filename;
 
+    /* must be called before using avcodec lib */
+    avcodec_init();
+
     /* register all the codecs */
     avcodec_register_all();
 

doc/examples/filtering.c

@@ -27,13 +27,11 @@
  */
 
 #define _XOPEN_SOURCE 600 /* for usleep */
-#include <unistd.h>
 
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
 #include <libavfilter/avfiltergraph.h>
-#include <libavfilter/avcodec.h>
-#include <libavfilter/buffersink.h>
+#include <libavfilter/vsrc_buffer.h>
 
 const char *filter_descr = "scale=78:24";
 
@@ -47,7 +45,7 @@ static int64_t last_pts = AV_NOPTS_VALUE;
 
 static int open_input_file(const char *filename)
 {
-    int ret;
+    int ret, i;
     AVCodec *dec;
 
     if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
@@ -55,7 +53,7 @@ static int open_input_file(const char *filename)
         return ret;
     }
 
-    if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
+    if ((ret = av_find_stream_info(fmt_ctx)) < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
         return ret;
     }
@@ -70,7 +68,7 @@ static int open_input_file(const char *filename)
     dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
 
     /* init the video decoder */
-    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
+    if ((ret = avcodec_open(dec_ctx, dec)) < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n");
         return ret;
     }
@@ -120,13 +118,12 @@ static int init_filters(const char *filters_descr)
     inputs->pad_idx    = 0;
     inputs->next       = NULL;
 
-    if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
+    if ((ret = avfilter_graph_parse(filter_graph, filter_descr,
                                     &inputs, &outputs, NULL)) < 0)
         return ret;
 
     if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
         return ret;
-    return 0;
 }
 
 static void display_picref(AVFilterBufferRef *picref, AVRational time_base)
@@ -198,14 +195,15 @@ int main(int argc, char **argv)
             }
 
             if (got_frame) {
-                frame.pts = av_frame_get_best_effort_timestamp(&frame);
-
+                if (frame.pts == AV_NOPTS_VALUE)
+                    frame.pts = frame.pkt_dts == AV_NOPTS_VALUE ?
+                        frame.pkt_dts : frame.pkt_pts;
                 /* push the decoded frame into the filtergraph */
-                av_vsrc_buffer_add_frame(buffersrc_ctx, &frame, 0);
+                av_vsrc_buffer_add_frame(buffersrc_ctx, &frame);
 
                 /* pull filtered pictures from the filtergraph */
                 while (avfilter_poll_frame(buffersink_ctx->inputs[0])) {
-                    av_buffersink_get_buffer_ref(buffersink_ctx, &picref, 0);
+                    av_vsink_buffer_get_video_buffer_ref(buffersink_ctx, &picref, 0);
                     if (picref) {
                         display_picref(picref, buffersink_ctx->inputs[0]->time_base);
                         avfilter_unref_buffer(picref);
@@ -218,7 +216,7 @@ end:
     avfilter_graph_free(&filter_graph);
     if (dec_ctx)
         avcodec_close(dec_ctx);
-    avformat_close_input(&fmt_ctx);
+    av_close_input_file(fmt_ctx);
 
     if (ret < 0 && ret != AVERROR_EOF) {
         char buf[1024];

doc/examples/filtering_audio.c

@@ -1,235 +0,0 @@
-/*
- * Copyright (c) 2010 Nicolas George
- * Copyright (c) 2011 Stefano Sabatini
- * Copyright (c) 2012 Clément Bœsch
- *
- * 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
- * API example for audio decoding and filtering
- */
-
-#include <unistd.h>
-
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavfilter/avfiltergraph.h>
-#include <libavfilter/avcodec.h>
-#include <libavfilter/buffersink.h>
-#include <libavfilter/buffersrc.h>
-
-const char *filter_descr = "aresample=8000,aconvert=s16:mono";
-const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";
-
-static AVFormatContext *fmt_ctx;
-static AVCodecContext *dec_ctx;
-AVFilterContext *buffersink_ctx;
-AVFilterContext *buffersrc_ctx;
-AVFilterGraph *filter_graph;
-static int audio_stream_index = -1;
-
-static int open_input_file(const char *filename)
-{
-    int ret;
-    AVCodec *dec;
-
-    if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
-        return ret;
-    }
-
-    if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
-        return ret;
-    }
-
-    /* select the audio stream */
-    ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n");
-        return ret;
-    }
-    audio_stream_index = ret;
-    dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
-
-    /* init the audio decoder */
-    if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
-        return ret;
-    }
-
-    return 0;
-}
-
-static int init_filters(const char *filters_descr)
-{
-    char args[512];
-    int ret;
-    AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
-    AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
-    AVFilterInOut *outputs = avfilter_inout_alloc();
-    AVFilterInOut *inputs  = avfilter_inout_alloc();
-    const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
-    const int64_t *chlayouts                = avfilter_all_channel_layouts;
-    AVABufferSinkParams *abuffersink_params;
-    const AVFilterLink *outlink;
-
-    filter_graph = avfilter_graph_alloc();
-
-    /* buffer audio source: the decoded frames from the decoder will be inserted here. */
-    if (!dec_ctx->channel_layout)
-        dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
-    snprintf(args, sizeof(args), "%d:%d:0x%"PRIx64,
-             dec_ctx->sample_rate, dec_ctx->sample_fmt, dec_ctx->channel_layout);
-    ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
-                                       args, NULL, filter_graph);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
-        return ret;
-    }
-
-    /* buffer audio sink: to terminate the filter chain. */
-    abuffersink_params = av_abuffersink_params_alloc();
-    abuffersink_params->sample_fmts     = sample_fmts;
-    abuffersink_params->channel_layouts = chlayouts;
-    ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
-                                       NULL, abuffersink_params, filter_graph);
-    av_free(abuffersink_params);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
-        return ret;
-    }
-
-    /* Endpoints for the filter graph. */
-    outputs->name       = av_strdup("in");
-    outputs->filter_ctx = buffersrc_ctx;
-    outputs->pad_idx    = 0;
-    outputs->next       = NULL;
-
-    inputs->name       = av_strdup("out");
-    inputs->filter_ctx = buffersink_ctx;
-    inputs->pad_idx    = 0;
-    inputs->next       = NULL;
-
-    if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
-                                    &inputs, &outputs, NULL)) < 0)
-        return ret;
-
-    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        return ret;
-
-    /* Print summary of the sink buffer
-     * Note: args buffer is reused to store channel layout string */
-    outlink = buffersink_ctx->inputs[0];
-    av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
-    av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
-           (int)outlink->sample_rate,
-           (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
-           args);
-
-    return 0;
-}
-
-static void print_samplesref(AVFilterBufferRef *samplesref)
-{
-    const AVFilterBufferRefAudioProps *props = samplesref->audio;
-    const int n = props->nb_samples * av_get_channel_layout_nb_channels(props->channel_layout);
-    const uint16_t *p     = (uint16_t*)samplesref->data[0];
-    const uint16_t *p_end = p + n;
-
-    while (p < p_end) {
-        fputc(*p    & 0xff, stdout);
-        fputc(*p>>8 & 0xff, stdout);
-        p++;
-    }
-    fflush(stdout);
-}
-
-int main(int argc, char **argv)
-{
-    int ret;
-    AVPacket packet;
-    AVFrame frame;
-    int got_frame;
-
-    if (argc != 2) {
-        fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
-        exit(1);
-    }
-
-    avcodec_register_all();
-    av_register_all();
-    avfilter_register_all();
-
-    if ((ret = open_input_file(argv[1])) < 0)
-        goto end;
-    if ((ret = init_filters(filter_descr)) < 0)
-        goto end;
-
-    /* read all packets */
-    while (1) {
-        AVFilterBufferRef *samplesref;
-        if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
-            break;
-
-        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);
-            av_free_packet(&packet);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
-                continue;
-            }
-
-            if (got_frame) {
-                /* push the audio data from decoded frame into the filtergraph */
-                if (av_buffersrc_add_frame(buffersrc_ctx, &frame, 0) < 0) {
-                    av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
-                    break;
-                }
-
-                /* pull filtered audio from the filtergraph */
-                while (avfilter_poll_frame(buffersink_ctx->inputs[0])) {
-                    av_buffersink_get_buffer_ref(buffersink_ctx, &samplesref, 0);
-                    if (samplesref) {
-                        print_samplesref(samplesref);
-                        avfilter_unref_buffer(samplesref);
-                    }
-                }
-            }
-        }
-    }
-end:
-    avfilter_graph_free(&filter_graph);
-    if (dec_ctx)
-        avcodec_close(dec_ctx);
-    avformat_close_input(&fmt_ctx);
-
-    if (ret < 0 && ret != AVERROR_EOF) {
-        char buf[1024];
-        av_strerror(ret, buf, sizeof(buf));
-        fprintf(stderr, "Error occurred: %s\n", buf);
-        exit(1);
-    }
-
-    exit(0);
-}

doc/examples/metadata.c

@@ -50,6 +50,6 @@ int main (int argc, char **argv)
     while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
         printf("%s=%s\n", tag->key, tag->value);
 
-    avformat_close_input(&fmt_ctx);
+    avformat_free_context(fmt_ctx);
     return 0;
 }

doc/examples/muxing.c

@@ -33,9 +33,9 @@
 #include <string.h>
 #include <math.h>
 
-#include <libavutil/mathematics.h>
-#include <libavformat/avformat.h>
-#include <libswscale/swscale.h>
+#include "libavutil/mathematics.h"
+#include "libavformat/avformat.h"
+#include "libswscale/swscale.h"
 
 #undef exit
 
@@ -43,7 +43,7 @@
 #define STREAM_DURATION   200.0
 #define STREAM_FRAME_RATE 25 /* 25 images/s */
 #define STREAM_NB_FRAMES  ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
-#define STREAM_PIX_FMT    PIX_FMT_YUV420P /* default pix_fmt */
+#define STREAM_PIX_FMT PIX_FMT_YUV420P /* default pix_fmt */
 
 static int sws_flags = SWS_BICUBIC;
 
@@ -52,6 +52,8 @@ static int sws_flags = SWS_BICUBIC;
 
 static float t, tincr, tincr2;
 static int16_t *samples;
+static uint8_t *audio_outbuf;
+static int audio_outbuf_size;
 static int audio_input_frame_size;
 
 /*
@@ -61,16 +63,8 @@ static AVStream *add_audio_stream(AVFormatContext *oc, enum CodecID codec_id)
 {
     AVCodecContext *c;
     AVStream *st;
-    AVCodec *codec;
 
-    /* find the audio encoder */
-    codec = avcodec_find_encoder(codec_id);
-    if (!codec) {
-        fprintf(stderr, "codec not found\n");
-        exit(1);
-    }
-
-    st = avformat_new_stream(oc, codec);
+    st = avformat_new_stream(oc, NULL);
     if (!st) {
         fprintf(stderr, "Could not alloc stream\n");
         exit(1);
@@ -78,12 +72,14 @@ static AVStream *add_audio_stream(AVFormatContext *oc, enum CodecID codec_id)
     st->id = 1;
 
     c = st->codec;
+    c->codec_id = codec_id;
+    c->codec_type = AVMEDIA_TYPE_AUDIO;
 
     /* put sample parameters */
-    c->sample_fmt  = AV_SAMPLE_FMT_S16;
-    c->bit_rate    = 64000;
+    c->sample_fmt = AV_SAMPLE_FMT_S16;
+    c->bit_rate = 64000;
     c->sample_rate = 44100;
-    c->channels    = 2;
+    c->channels = 2;
 
     // some formats want stream headers to be separate
     if (oc->oformat->flags & AVFMT_GLOBALHEADER)
@@ -95,32 +91,54 @@ static AVStream *add_audio_stream(AVFormatContext *oc, enum CodecID codec_id)
 static void open_audio(AVFormatContext *oc, AVStream *st)
 {
     AVCodecContext *c;
+    AVCodec *codec;
 
     c = st->codec;
 
+    /* find the audio encoder */
+    codec = avcodec_find_encoder(c->codec_id);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
+        exit(1);
+    }
+
     /* open it */
-    if (avcodec_open2(c, NULL, NULL) < 0) {
+    if (avcodec_open(c, codec) < 0) {
         fprintf(stderr, "could not open codec\n");
         exit(1);
     }
 
     /* init signal generator */
-    t     = 0;
+    t = 0;
     tincr = 2 * M_PI * 110.0 / c->sample_rate;
     /* increment frequency by 110 Hz per second */
     tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
 
-    if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
-        audio_input_frame_size = 10000;
-    else
+    audio_outbuf_size = 10000;
+    audio_outbuf = av_malloc(audio_outbuf_size);
+
+    /* ugly hack for PCM codecs (will be removed ASAP with new PCM
+       support to compute the input frame size in samples */
+    if (c->frame_size <= 1) {
+        audio_input_frame_size = audio_outbuf_size / c->channels;
+        switch(st->codec->codec_id) {
+        case CODEC_ID_PCM_S16LE:
+        case CODEC_ID_PCM_S16BE:
+        case CODEC_ID_PCM_U16LE:
+        case CODEC_ID_PCM_U16BE:
+            audio_input_frame_size >>= 1;
+            break;
+        default:
+            break;
+        }
+    } else {
         audio_input_frame_size = c->frame_size;
-    samples = av_malloc(audio_input_frame_size *
-                        av_get_bytes_per_sample(c->sample_fmt) *
-                        c->channels);
+    }
+    samples = av_malloc(audio_input_frame_size * 2 * c->channels);
 }
 
-/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
- * 'nb_channels' channels. */
+/* prepare a 16 bit dummy audio frame of 'frame_size' samples and
+   'nb_channels' channels */
 static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
 {
     int j, i, v;
@@ -129,9 +147,9 @@ static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
     q = samples;
     for (j = 0; j < frame_size; j++) {
         v = (int)(sin(t) * 10000);
-        for (i = 0; i < nb_channels; i++)
+        for(i = 0; i < nb_channels; i++)
             *q++ = v;
-        t     += tincr;
+        t += tincr;
         tincr += tincr2;
     }
 }
@@ -139,28 +157,22 @@ static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
 static void write_audio_frame(AVFormatContext *oc, AVStream *st)
 {
     AVCodecContext *c;
-    AVPacket pkt = { 0 }; // data and size must be 0;
-    AVFrame *frame = avcodec_alloc_frame();
-    int got_packet;
-
+    AVPacket pkt;
     av_init_packet(&pkt);
+
     c = st->codec;
 
     get_audio_frame(samples, audio_input_frame_size, c->channels);
-    frame->nb_samples = audio_input_frame_size;
-    avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
-                             (uint8_t *)samples,
-                             audio_input_frame_size *
-                             av_get_bytes_per_sample(c->sample_fmt) *
-                             c->channels, 1);
 
-    avcodec_encode_audio2(c, &pkt, frame, &got_packet);
-    if (!got_packet)
-        return;
+    pkt.size = avcodec_encode_audio(c, audio_outbuf, audio_outbuf_size, samples);
 
+    if (c->coded_frame && c->coded_frame->pts != AV_NOPTS_VALUE)
+        pkt.pts= av_rescale_q(c->coded_frame->pts, c->time_base, st->time_base);
+    pkt.flags |= AV_PKT_FLAG_KEY;
     pkt.stream_index = st->index;
+    pkt.data = audio_outbuf;
 
-    /* Write the compressed frame to the media file. */
+    /* write the compressed frame in the media file */
     if (av_interleaved_write_frame(oc, &pkt) != 0) {
         fprintf(stderr, "Error while writing audio frame\n");
         exit(1);
@@ -172,6 +184,7 @@ static void close_audio(AVFormatContext *oc, AVStream *st)
     avcodec_close(st->codec);
 
     av_free(samples);
+    av_free(audio_outbuf);
 }
 
 /**************************************************************/
@@ -181,21 +194,14 @@ static AVFrame *picture, *tmp_picture;
 static uint8_t *video_outbuf;
 static int frame_count, video_outbuf_size;
 
-/* Add a video output stream. */
+/* add a video output stream */
 static AVStream *add_video_stream(AVFormatContext *oc, enum CodecID codec_id)
 {
     AVCodecContext *c;
     AVStream *st;
     AVCodec *codec;
 
-    /* find the video encoder */
-    codec = avcodec_find_encoder(codec_id);
-    if (!codec) {
-        fprintf(stderr, "codec not found\n");
-        exit(1);
-    }
-
-    st = avformat_new_stream(oc, codec);
+    st = avformat_new_stream(oc, NULL);
     if (!st) {
         fprintf(stderr, "Could not alloc stream\n");
         exit(1);
@@ -213,30 +219,30 @@ static AVStream *add_video_stream(AVFormatContext *oc, enum CodecID codec_id)
 
     c->codec_id = codec_id;
 
-    /* Put sample parameters. */
+    /* put sample parameters */
     c->bit_rate = 400000;
-    /* Resolution must be a multiple of two. */
-    c->width    = 352;
-    c->height   = 288;
-    /* timebase: This is the fundamental unit of time (in seconds) in terms
-     * of which frame timestamps are represented. For fixed-fps content,
-     * timebase should be 1/framerate and timestamp increments should be
-     * identical to 1. */
+    /* resolution must be a multiple of two */
+    c->width = 352;
+    c->height = 288;
+    /* time base: this is the fundamental unit of time (in seconds) in terms
+       of which frame timestamps are represented. for fixed-fps content,
+       timebase should be 1/framerate and timestamp increments should be
+       identically 1. */
     c->time_base.den = STREAM_FRAME_RATE;
     c->time_base.num = 1;
-    c->gop_size      = 12; /* emit one intra frame every twelve frames at most */
-    c->pix_fmt       = STREAM_PIX_FMT;
+    c->gop_size = 12; /* emit one intra frame every twelve frames at most */
+    c->pix_fmt = STREAM_PIX_FMT;
     if (c->codec_id == CODEC_ID_MPEG2VIDEO) {
         /* just for testing, we also add B frames */
         c->max_b_frames = 2;
     }
-    if (c->codec_id == CODEC_ID_MPEG1VIDEO) {
+    if (c->codec_id == CODEC_ID_MPEG1VIDEO){
         /* Needed to avoid using macroblocks in which some coeffs overflow.
-         * This does not happen with normal video, it just happens here as
-         * the motion of the chroma plane does not match the luma plane. */
-        c->mb_decision = 2;
+           This does not happen with normal video, it just happens here as
+           the motion of the chroma plane does not match the luma plane. */
+        c->mb_decision=2;
     }
-    /* Some formats want stream headers to be separate. */
+    // some formats want stream headers to be separate
     if (oc->oformat->flags & AVFMT_GLOBALHEADER)
         c->flags |= CODEC_FLAG_GLOBAL_HEADER;
 
@@ -252,7 +258,7 @@ static AVFrame *alloc_picture(enum PixelFormat pix_fmt, int width, int height)
     picture = avcodec_alloc_frame();
     if (!picture)
         return NULL;
-    size        = avpicture_get_size(pix_fmt, width, height);
+    size = avpicture_get_size(pix_fmt, width, height);
     picture_buf = av_malloc(size);
     if (!picture_buf) {
         av_free(picture);
@@ -265,38 +271,46 @@ static AVFrame *alloc_picture(enum PixelFormat pix_fmt, int width, int height)
 
 static void open_video(AVFormatContext *oc, AVStream *st)
 {
+    AVCodec *codec;
     AVCodecContext *c;
 
     c = st->codec;
 
+    /* find the video encoder */
+    codec = avcodec_find_encoder(c->codec_id);
+    if (!codec) {
+        fprintf(stderr, "codec not found\n");
+        exit(1);
+    }
+
     /* open the codec */
-    if (avcodec_open2(c, NULL, NULL) < 0) {
+    if (avcodec_open(c, codec) < 0) {
         fprintf(stderr, "could not open codec\n");
         exit(1);
     }
 
     video_outbuf = NULL;
     if (!(oc->oformat->flags & AVFMT_RAWPICTURE)) {
-        /* Allocate output buffer. */
-        /* XXX: API change will be done. */
-        /* Buffers passed into lav* can be allocated any way you prefer,
-         * as long as they're aligned enough for the architecture, and
-         * they're freed appropriately (such as using av_free for buffers
-         * allocated with av_malloc). */
+        /* allocate output buffer */
+        /* XXX: API change will be done */
+        /* buffers passed into lav* can be allocated any way you prefer,
+           as long as they're aligned enough for the architecture, and
+           they're freed appropriately (such as using av_free for buffers
+           allocated with av_malloc) */
         video_outbuf_size = 200000;
-        video_outbuf      = av_malloc(video_outbuf_size);
+        video_outbuf = av_malloc(video_outbuf_size);
     }
 
-    /* Allocate the encoded raw picture. */
+    /* allocate the encoded raw picture */
     picture = alloc_picture(c->pix_fmt, c->width, c->height);
     if (!picture) {
         fprintf(stderr, "Could not allocate picture\n");
         exit(1);
     }
 
-    /* If the output format is not YUV420P, then a temporary YUV420P
-     * picture is needed too. It is then converted to the required
-     * output format. */
+    /* if the output format is not YUV420P, then a temporary YUV420P
+       picture is needed too. It is then converted to the required
+       output format */
     tmp_picture = NULL;
     if (c->pix_fmt != PIX_FMT_YUV420P) {
         tmp_picture = alloc_picture(PIX_FMT_YUV420P, c->width, c->height);
@@ -307,22 +321,23 @@ static void open_video(AVFormatContext *oc, AVStream *st)
     }
 }
 
-/* Prepare a dummy image. */
-static void fill_yuv_image(AVFrame *pict, int frame_index,
-                           int width, int height)
+/* prepare a dummy image */
+static void fill_yuv_image(AVFrame *pict, int frame_index, int width, int height)
 {
     int x, y, i;
 
     i = frame_index;
 
     /* Y */
-    for (y = 0; y < height; y++)
-        for (x = 0; x < width; x++)
+    for (y = 0; y < height; y++) {
+        for (x = 0; x < width; x++) {
             pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
+        }
+    }
 
     /* Cb and Cr */
-    for (y = 0; y < height / 2; y++) {
-        for (x = 0; x < width / 2; x++) {
+    for (y = 0; y < height/2; y++) {
+        for (x = 0; x < width/2; x++) {
             pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
             pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
         }
@@ -338,13 +353,13 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)
     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. */
+        /* no more frame 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 (c->pix_fmt != PIX_FMT_YUV420P) {
             /* as we only generate a YUV420P picture, we must convert it
-             * to the codec pixel format if needed */
+               to the codec pixel format if needed */
             if (img_convert_ctx == NULL) {
                 img_convert_ctx = sws_getContext(c->width, c->height,
                                                  PIX_FMT_YUV420P,
@@ -352,8 +367,7 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)
                                                  c->pix_fmt,
                                                  sws_flags, NULL, NULL, NULL);
                 if (img_convert_ctx == NULL) {
-                    fprintf(stderr,
-                            "Cannot initialize the conversion context\n");
+                    fprintf(stderr, "Cannot initialize the conversion context\n");
                     exit(1);
                 }
             }
@@ -365,38 +379,36 @@ static void write_video_frame(AVFormatContext *oc, AVStream *st)
         }
     }
 
+
     if (oc->oformat->flags & AVFMT_RAWPICTURE) {
-        /* Raw video case - the API will change slightly in the near
-         * future for that. */
+        /* raw video case. The API will change slightly in the near
+           future for that. */
         AVPacket pkt;
         av_init_packet(&pkt);
 
-        pkt.flags        |= AV_PKT_FLAG_KEY;
-        pkt.stream_index  = st->index;
-        pkt.data          = (uint8_t *)picture;
-        pkt.size          = sizeof(AVPicture);
+        pkt.flags |= AV_PKT_FLAG_KEY;
+        pkt.stream_index = st->index;
+        pkt.data = (uint8_t *)picture;
+        pkt.size = sizeof(AVPicture);
 
         ret = av_interleaved_write_frame(oc, &pkt);
     } else {
         /* encode the image */
-        out_size = avcodec_encode_video(c, video_outbuf,
-                                        video_outbuf_size, picture);
-        /* If size is zero, it means the image was buffered. */
+        out_size = avcodec_encode_video(c, video_outbuf, video_outbuf_size, picture);
+        /* if zero size, it means the image was buffered */
         if (out_size > 0) {
             AVPacket pkt;
             av_init_packet(&pkt);
 
             if (c->coded_frame->pts != AV_NOPTS_VALUE)
-                pkt.pts = av_rescale_q(c->coded_frame->pts,
-                                       c->time_base, st->time_base);
-            if (c->coded_frame->key_frame)
+                pkt.pts= av_rescale_q(c->coded_frame->pts, c->time_base, st->time_base);
+            if(c->coded_frame->key_frame)
                 pkt.flags |= AV_PKT_FLAG_KEY;
-
             pkt.stream_index = st->index;
-            pkt.data         = video_outbuf;
-            pkt.size         = out_size;
+            pkt.data = video_outbuf;
+            pkt.size = out_size;
 
-            /* Write the compressed frame to the media file. */
+            /* write the compressed frame in the media file */
             ret = av_interleaved_write_frame(oc, &pkt);
         } else {
             ret = 0;
@@ -433,7 +445,7 @@ int main(int argc, char **argv)
     double audio_pts, video_pts;
     int i;
 
-    /* Initialize libavcodec, and register all codecs and formats. */
+    /* initialize libavcodec, and register all codecs and formats */
     av_register_all();
 
     if (argc != 2) {
@@ -458,8 +470,8 @@ int main(int argc, char **argv)
     }
     fmt = oc->oformat;
 
-    /* Add the audio and video streams using the default format codecs
-     * and initialize the codecs. */
+    /* add the audio and video streams using the default format codecs
+       and initialize the codecs */
     video_st = NULL;
     audio_st = NULL;
     if (fmt->video_codec != CODEC_ID_NONE) {
@@ -469,15 +481,15 @@ int main(int argc, char **argv)
         audio_st = add_audio_stream(oc, fmt->audio_codec);
     }
 
-    /* Now that all the parameters are set, we can open the audio and
-     * video codecs and allocate the necessary encode buffers. */
+    av_dump_format(oc, 0, filename, 1);
+
+    /* now that all the parameters are set, we can open the audio and
+       video codecs and allocate the necessary encode buffers */
     if (video_st)
         open_video(oc, video_st);
     if (audio_st)
         open_audio(oc, audio_st);
 
-    av_dump_format(oc, 0, filename, 1);
-
     /* open the output file, if needed */
     if (!(fmt->flags & AVFMT_NOFILE)) {
         if (avio_open(&oc->pb, filename, AVIO_FLAG_WRITE) < 0) {
@@ -486,20 +498,18 @@ int main(int argc, char **argv)
         }
     }
 
-    /* Write the stream header, if any. */
-    avformat_write_header(oc, NULL);
-
+    /* write the stream header, if any */
+    av_write_header(oc);
     picture->pts = 0;
-    for (;;) {
-        /* Compute current audio and video time. */
+    for(;;) {
+        /* compute current audio and video time */
         if (audio_st)
             audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
         else
             audio_pts = 0.0;
 
         if (video_st)
-            video_pts = (double)video_st->pts.val * video_st->time_base.num /
-                        video_st->time_base.den;
+            video_pts = (double)video_st->pts.val * video_st->time_base.num / video_st->time_base.den;
         else
             video_pts = 0.0;
 
@@ -516,27 +526,28 @@ int main(int argc, char **argv)
         }
     }
 
-    /* Write the trailer, if any. The trailer must be written before you
-     * close the CodecContexts open when you wrote the header; otherwise
-     * av_write_trailer() may try to use memory that was freed on
-     * av_codec_close(). */
+    /* write the trailer, if any.  the trailer must be written
+     * before you close the CodecContexts open when you wrote the
+     * header; otherwise write_trailer may try to use memory that
+     * was freed on av_codec_close() */
     av_write_trailer(oc);
 
-    /* Close each codec. */
+    /* close each codec */
     if (video_st)
         close_video(oc, video_st);
     if (audio_st)
         close_audio(oc, audio_st);
 
-    /* Free the streams. */
-    for (i = 0; i < oc->nb_streams; i++) {
+    /* free the streams */
+    for(i = 0; i < oc->nb_streams; i++) {
         av_freep(&oc->streams[i]->codec);
         av_freep(&oc->streams[i]);
     }
 
-    if (!(fmt->flags & AVFMT_NOFILE))
-        /* Close the output file. */
+    if (!(fmt->flags & AVFMT_NOFILE)) {
+        /* close the output file */
         avio_close(oc->pb);
+    }
 
     /* free the stream */
     av_free(oc);

doc/fate.texi

@@ -166,11 +166,9 @@ the synchronisation of the samples directory.
 @item THREADS
     Specify how many threads to use while running regression tests, it is
     quite useful to detect thread-related regressions.
-@item CPUFLAGS
-    Specify CPU flags.
 @end table
 
 Example:
 @example
-make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate
+make V=1 SAMPLES=/var/fate/samples THREADS=2 fate
 @end example

doc/ffmpeg.texi

@@ -143,7 +143,7 @@ Stop writing the output after its duration reaches @var{duration}.
 @var{duration} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.
 
 @item -fs @var{limit_size} (@emph{output})
-Set the file size limit, expressed in bytes.
+Set the file size limit.
 
 @item -ss @var{position} (@emph{input/output})
 When used as an input option (before @code{-i}), seeks in this input file to
@@ -164,7 +164,7 @@ streams are delayed by @var{offset} seconds.
 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])
+now|([(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH[:MM[:SS[.m...]]])|(HH[MM[SS[.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
@@ -224,23 +224,12 @@ codec-dependent.
 @var{filter_graph} is a description of the filter graph to apply to
 the stream. Use @code{-filters} to show all the available filters
 (including also sources and sinks).
-
-See also the @option{-filter_complex} option if you want to create filter graphs
-with multiple inputs and/or outputs.
 @item -pre[:@var{stream_specifier}] @var{preset_name} (@emph{output,per-stream})
 Specify the preset for matching stream(s).
 
 @item -stats (@emph{global})
 Print encoding progress/statistics. On by default.
 
-@item -debug_ts (@emph{global})
-Print timestamp information. It is off by default. This option is
-mostly useful for testing and debugging purposes, and the output
-format may change from one version to another, so it should not be
-employed by portable scripts.
-
-See also the option @code{-fdebug ts}.
-
 @item -attach @var{filename} (@emph{output})
 Add an attachment to the output file. This is supported by a few formats
 like Matroska for e.g. fonts used in rendering subtitles. Attachments
@@ -282,10 +271,70 @@ attachments.
 @item -vframes @var{number} (@emph{output})
 Set the number of video frames to record. This is an alias for @code{-frames:v}.
 @item -r[:@var{stream_specifier}] @var{fps} (@emph{input/output,per-stream})
-Set frame rate (Hz value, fraction or abbreviation), (default = 25). For output
-streams implies @code{-vsync cfr}.
+Set frame rate (Hz value, fraction or abbreviation), (default = 25).
 @item -s[:@var{stream_specifier}] @var{size} (@emph{input/output,per-stream})
 Set frame size. The format is @samp{wxh} (default - same as source).
+The following abbreviations are recognized:
+@table @samp
+@item sqcif
+128x96
+@item qcif
+176x144
+@item cif
+352x288
+@item 4cif
+704x576
+@item 16cif
+1408x1152
+@item qqvga
+160x120
+@item qvga
+320x240
+@item vga
+640x480
+@item svga
+800x600
+@item xga
+1024x768
+@item uxga
+1600x1200
+@item qxga
+2048x1536
+@item sxga
+1280x1024
+@item qsxga
+2560x2048
+@item hsxga
+5120x4096
+@item wvga
+852x480
+@item wxga
+1366x768
+@item wsxga
+1600x1024
+@item wuxga
+1920x1200
+@item woxga
+2560x1600
+@item wqsxga
+3200x2048
+@item wquxga
+3840x2400
+@item whsxga
+6400x4096
+@item whuxga
+7680x4800
+@item cga
+320x200
+@item ega
+640x350
+@item hd480
+852x480
+@item hd720
+1280x720
+@item hd1080
+1920x1080
+@end table
 
 @item -aspect[:@var{stream_specifier}] @var{aspect} (@emph{output,per-stream})
 Set the video display aspect ratio specified by @var{aspect}.
@@ -312,7 +361,25 @@ pad=width:height:x:y:color instead.
 
 @item -vn (@emph{output})
 Disable video recording.
-
+@item -bt @var{tolerance}
+Set video bitrate tolerance (in bits, default 4000k).
+Has a minimum value of: (target_bitrate/target_framerate).
+In 1-pass mode, bitrate tolerance specifies how far ratecontrol is
+willing to deviate from the target average bitrate value. This is
+not related to min/max bitrate. Lowering tolerance too much has
+an adverse effect on quality.
+@item -maxrate @var{bitrate}
+Set max video bitrate (in bit/s).
+Requires -bufsize to be set.
+@item -minrate @var{bitrate}
+Set min video bitrate (in bit/s).
+Most useful in setting up a CBR encode:
+@example
+ffmpeg -i myfile.avi -b:v 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v
+@end example
+It is of little use elsewise.
+@item -bufsize @var{size}
+Set video buffer verifier buffer size (in bits).
 @item -vcodec @var{codec} (@emph{output})
 Set the video codec. This is an alias for @code{-codec:v}.
 @item -same_quant
@@ -340,6 +407,10 @@ prefix is ``ffmpeg2pass''. The complete file name will be
 @file{PREFIX-N.log}, where N is a number specific to the output
 stream
 
+Note that this option is overwritten by a local option of the same name
+when using @code{-vcodec libx264}. That option maps to the x264 option stats
+which has a different syntax.
+
 @item -vlang @var{code}
 Set the ISO 639 language code (3 letters) of the current video stream.
 
@@ -357,29 +428,202 @@ also sources and sinks).  This is an alias for @code{-filter:v}.
 @item -pix_fmt[:@var{stream_specifier}] @var{format} (@emph{input/output,per-stream})
 Set pixel format. Use @code{-pix_fmts} to show all the supported
 pixel formats.
-If the selected pixel format can not be selected, ffmpeg will print a
-warning and select the best pixel format supported by the encoder.
-If @var{pix_fmt} is prefixed by a @code{+}, ffmpeg will exit with an error
-if the requested pixel format can not be selected, and automatic conversions
-inside filter graphs are disabled.
-If @var{pix_fmt} is a single @code{+}, ffmpeg selects the same pixel format
-as the input (or graph output) and automatic conversions are disabled.
-
 @item -sws_flags @var{flags} (@emph{input/output})
 Set SwScaler flags.
+@item -g @var{gop_size}
+Set the group of pictures size.
+@item -intra
+deprecated, use -g 1
 @item -vdt @var{n}
 Discard threshold.
+@item -qmin @var{q}
+minimum video quantizer scale (VBR)
+@item -qmax @var{q}
+maximum video quantizer scale (VBR)
+@item -qdiff @var{q}
+maximum difference between the quantizer scales (VBR)
+@item -qblur @var{blur}
+video quantizer scale blur (VBR) (range 0.0 - 1.0)
+@item -qcomp @var{compression}
+video quantizer scale compression (VBR) (default 0.5).
+Constant of ratecontrol equation. Recommended range for default rc_eq: 0.0-1.0
+
+@item -lmin @var{lambda}
+minimum video lagrange factor (VBR)
+@item -lmax @var{lambda}
+max video lagrange factor (VBR)
+@item -mblmin @var{lambda}
+minimum macroblock quantizer scale (VBR)
+@item -mblmax @var{lambda}
+maximum macroblock quantizer scale (VBR)
+
+These four options (lmin, lmax, mblmin, mblmax) use 'lambda' units,
+but you may use the QP2LAMBDA constant to easily convert from 'q' units:
+@example
+ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
+@end example
+
+@item -rc_init_cplx @var{complexity}
+initial complexity for single pass encoding
+@item -b_qfactor @var{factor}
+qp factor between P- and B-frames
+@item -i_qfactor @var{factor}
+qp factor between P- and I-frames
+@item -b_qoffset @var{offset}
+qp offset between P- and B-frames
+@item -i_qoffset @var{offset}
+qp offset between P- and I-frames
+@item -rc_eq @var{equation}
+Set rate control equation (see section "Expression Evaluation")
+(default = @code{tex^qComp}).
+
+When computing the rate control equation expression, besides the
+standard functions defined in the section "Expression Evaluation", the
+following functions are available:
+@table @var
+@item bits2qp(bits)
+@item qp2bits(qp)
+@end table
+
+and the following constants are available:
+@table @var
+@item iTex
+@item pTex
+@item tex
+@item mv
+@item fCode
+@item iCount
+@item mcVar
+@item var
+@item isI
+@item isP
+@item isB
+@item avgQP
+@item qComp
+@item avgIITex
+@item avgPITex
+@item avgPPTex
+@item avgBPTex
+@item avgTex
+@end table
 
 @item -rc_override[:@var{stream_specifier}] @var{override} (@emph{output,per-stream})
 Rate control override for specific intervals, formatted as "int,int,int"
 list separated with slashes. Two first values are the beginning and
 end frame numbers, last one is quantizer to use if positive, or quality
 factor if negative.
+@item -me_method @var{method}
+Set motion estimation method to @var{method}.
+Available methods are (from lowest to best quality):
+@table @samp
+@item zero
+Try just the (0, 0) vector.
+@item phods
+@item log
+@item x1
+@item hex
+@item umh
+@item epzs
+(default method)
+@item full
+exhaustive search (slow and marginally better than epzs)
+@end table
+
+@item -dct_algo @var{algo}
+Set DCT algorithm to @var{algo}. Available values are:
+@table @samp
+@item 0
+FF_DCT_AUTO (default)
+@item 1
+FF_DCT_FASTINT
+@item 2
+FF_DCT_INT
+@item 3
+FF_DCT_MMX
+@item 4
+FF_DCT_MLIB
+@item 5
+FF_DCT_ALTIVEC
+@end table
+
+@item -idct_algo @var{algo}
+Set IDCT algorithm to @var{algo}. Available values are:
+@table @samp
+@item 0
+FF_IDCT_AUTO (default)
+@item 1
+FF_IDCT_INT
+@item 2
+FF_IDCT_SIMPLE
+@item 3
+FF_IDCT_SIMPLEMMX
+@item 4
+FF_IDCT_LIBMPEG2MMX
+@item 5
+FF_IDCT_PS2
+@item 6
+FF_IDCT_MLIB
+@item 7
+FF_IDCT_ARM
+@item 8
+FF_IDCT_ALTIVEC
+@item 9
+FF_IDCT_SH4
+@item 10
+FF_IDCT_SIMPLEARM
+@end table
+
+@item -er @var{n}
+Set error resilience to @var{n}.
+@table @samp
+@item 1
+FF_ER_CAREFUL (default)
+@item 2
+FF_ER_COMPLIANT
+@item 3
+FF_ER_AGGRESSIVE
+@item 4
+FF_ER_VERY_AGGRESSIVE
+@end table
+
+@item -ec @var{bit_mask}
+Set error concealment to @var{bit_mask}. @var{bit_mask} is a bit mask of
+the following values:
+@table @samp
+@item 1
+FF_EC_GUESS_MVS (default = enabled)
+@item 2
+FF_EC_DEBLOCK (default = enabled)
+@end table
+
+@item -bf @var{frames}
+Use 'frames' B-frames (supported for MPEG-1, MPEG-2 and MPEG-4).
+@item -mbd @var{mode}
+macroblock decision
+@table @samp
+@item 0
+FF_MB_DECISION_SIMPLE: Use mb_cmp (cannot change it yet in ffmpeg).
+@item 1
+FF_MB_DECISION_BITS: Choose the one which needs the fewest bits.
+@item 2
+FF_MB_DECISION_RD: rate distortion
+@end table
+
+@item -4mv
+Use four motion vector by macroblock (MPEG-4 only).
+@item -part
+Use data partitioning (MPEG-4 only).
+@item -bug @var{param}
+Work around encoder bugs that are not auto-detected.
+@item -strict @var{strictness}
+How strictly to follow the standards.
+@item -aic
+Enable Advanced intra coding (h263+).
+@item -umv
+Enable Unlimited Motion Vector (h263+)
 
 @item -deinterlace
 Deinterlace pictures.
-This option is deprecated since the deinterlacing is very low quality.
-Use the yadif filter with @code{-filter:v yadif}.
 @item -ilme
 Force interlacing support in encoder (MPEG-2 and MPEG-4 only).
 Use this option if your input file is interlaced and you want
@@ -438,11 +682,6 @@ Set the audio codec. This is an alias for @code{-codec:a}.
 @item -sample_fmt[:@var{stream_specifier}] @var{sample_fmt} (@emph{output,per-stream})
 Set the audio sample format. Use @code{-sample_fmts} to get a list
 of supported sample formats.
-@item -af @var{filter_graph} (@emph{output})
-@var{filter_graph} is a description of the filter graph to apply to
-the input audio.
-Use the option "-filters" to show all the available filters (including
-also sources and sinks).  This is an alias for @code{-filter:a}.
 @end table
 
 @section Advanced Audio options:
@@ -450,6 +689,28 @@ also sources and sinks).  This is an alias for @code{-filter:a}.
 @table @option
 @item -atag @var{fourcc/tag} (@emph{output})
 Force audio tag/fourcc. This is an alias for @code{-tag:a}.
+@item -audio_service_type @var{type}
+Set the type of service that the audio stream contains.
+@table @option
+@item ma
+Main Audio Service (default)
+@item ef
+Effects
+@item vi
+Visually Impaired
+@item hi
+Hearing Impaired
+@item di
+Dialogue
+@item co
+Commentary
+@item em
+Emergency
+@item vo
+Voice Over
+@item ka
+Karaoke
+@end table
 @item -absf @var{bitstream_filter}
 Deprecated, see -bsf
 @end table
@@ -477,7 +738,7 @@ Synchronize read on input.
 @section Advanced options
 
 @table @option
-@item -map [-]@var{input_file_id}[:@var{stream_specifier}][,@var{sync_file_id}[:@var{stream_specifier}]] | @var{[linklabel]} (@emph{output})
+@item -map [-]@var{input_file_id}[:@var{stream_specifier}][,@var{sync_file_id}[:@var{stream_specifier}]] (@emph{output})
 
 Designate one or more input streams as a source for the output file. Each input
 stream is identified by the input file index @var{input_file_id} and
@@ -493,10 +754,6 @@ the source for output stream 1, etc.
 A @code{-} character before the stream identifier creates a "negative" mapping.
 It disables matching streams from already created mappings.
 
-An alternative @var{[linklabel]} form will map outputs from complex filter
-graphs (see the @option{-filter_complex} option) to the output file.
-@var{linklabel} must correspond to a defined output link label in the graph.
-
 For example, to map ALL streams from the first input file to output
 @example
 ffmpeg -i INPUT -map 0 output
@@ -534,7 +791,7 @@ Note that using this option disables the default mappings for this output file.
 
 @item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][:@var{output_file_id}.@var{stream_specifier}]
 Map an audio channel from a given input to an output. If
-@var{output_file_id}.@var{stream_specifier} is not set, the audio channel will
+@var{output_file_id}.@var{stream_specifier} are not set, the audio channel will
 be mapped on all the audio streams.
 
 Using "-1" instead of
@@ -556,18 +813,18 @@ The order of the "-map_channel" option specifies the order of the channels in
 the output stream. The output channel layout is guessed from the number of
 channels mapped (mono if one "-map_channel", stereo if two, etc.). Using "-ac"
 in combination of "-map_channel" makes the channel gain levels to be updated if
-input and output channel layouts don't match (for instance two "-map_channel"
-options and "-ac 6").
+channel layouts don't match (for instance two "-map_channel" options and "-ac
+6").
 
-You can also extract each channel of an input to specific outputs; the following
-command extracts two channels of the @var{INPUT} audio stream (file 0, stream 0)
-to the respective @var{OUTPUT_CH0} and @var{OUTPUT_CH1} outputs:
+You can also extract each channel of an @var{INPUT} to specific outputs; the
+following command extract each channel of the audio stream (file 0, stream 0)
+to the respective @var{OUTPUT_CH0} and @var{OUTPUT_CH1}:
 @example
 ffmpeg -i INPUT -map_channel 0.0.0 OUTPUT_CH0 -map_channel 0.0.1 OUTPUT_CH1
 @end example
 
-The following example splits the channels of a stereo input into two separate
-streams, which are put into the same output file:
+The following example split the channels of a stereo input into streams:
+
 @example
 ffmpeg -i stereo.wav -map 0:0 -map 0:0 -map_channel 0.0.0:0.0 -map_channel 0.0.1:0.1 -y out.ogg
 @end example
@@ -577,20 +834,9 @@ input stream; you can't for example use "-map_channel" to pick multiple input
 audio channels contained in different streams (from the same or different files)
 and merge them into a single output stream. It is therefore not currently
 possible, for example, to turn two separate mono streams into a single stereo
-stream. However splitting a stereo stream into two single channel mono streams
+stream. However spliting a stereo stream into two single channel mono streams
 is possible.
 
-If you need this feature, a possible workaround is to use the @emph{amerge}
-filter. For example, if you need to merge a media (here @file{input.mkv}) with 2
-mono audio streams into one single stereo channel audio stream (and keep the
-video stream), you can use the following command:
-@example
-ffmpeg -i input.mkv -f lavfi -i "
-amovie=input.mkv:si=1 [a1];
-amovie=input.mkv:si=2 [a2];
-[a1][a2] amerge" -c:a pcm_s16le -c:v copy output.mkv
-@end example
-
 @item -map_metadata[:@var{metadata_spec_out}] @var{infile}[:@var{metadata_spec_in}] (@emph{output,per-metadata})
 Set metadata information of the next output file from @var{infile}. Note that
 those are file indices (zero-based), not filenames.
@@ -675,15 +921,14 @@ Show benchmarking information at the end of an encode.
 Shows CPU time used and maximum memory consumption.
 Maximum memory consumption is not supported on all systems,
 it will usually display as 0 if not supported.
-@item -benchmark_all (@emph{global})
-Show benchmarking information during the encode.
-Shows CPU time used in various steps (audio/video encode/decode).
 @item -timelimit @var{duration} (@emph{global})
 Exit after ffmpeg has been running for @var{duration} seconds.
 @item -dump (@emph{global})
 Dump each input packet to stderr.
 @item -hex (@emph{global})
 When dumping packets, also dump the payload.
+@item -ps @var{size}
+Set RTP payload size in bytes.
 @item -re (@emph{input})
 Read input at native frame rate. Mainly used to simulate a grab device.
 @item -loop_input
@@ -694,10 +939,10 @@ This option is deprecated, use -loop 1.
 Repeatedly loop output for formats that support looping such as animated GIF
 (0 will loop the output infinitely).
 This option is deprecated, use -loop.
+@item -threads @var{count}
+Thread count.
 @item -vsync @var{parameter}
 Video sync method.
-For compatibility reasons old values can be specified as numbers.
-Newly added values will have to be specified as strings always.
 
 @table @option
 @item 0, passthrough
@@ -708,9 +953,6 @@ constant framerate.
 @item 2, vfr
 Frames are passed through with their timestamp or dropped so as to
 prevent 2 frames from having the same timestamp.
-@item drop
-As passthrough but destroys all timestamps, making the muxer generate
-fresh timestamps based on frame-rate.
 @item -1, auto
 Chooses between 1 and 2 depending on muxer capabilities. This is the
 default method.
@@ -725,33 +967,10 @@ Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps
 the parameter is the maximum samples per second by which the audio is changed.
 -async 1 is a special case where only the start of the audio stream is corrected
 without any later correction.
-This option has been deprecated. Use the @code{asyncts} audio filter instead.
 @item -copyts
 Copy timestamps from input to output.
-@item -copytb @var{mode}
-Specify how to set the encoder timebase when stream copying.  @var{mode} is an
-integer numeric value, and can assume one of the following values:
-
-@table @option
-@item 1
-Use the demuxer timebase.
-
-The time base is copied to the output encoder from the corresponding input
-demuxer. This is sometimes required to avoid non monotonically increasing
-timestamps when copying video streams with variable frame rate.
-
-@item 0
-Use the decoder timebase.
-
-The time base is copied to the output encoder from the corresponding input
-decoder.
-
-@item -1
-Try to make the choice automatically, in order to generate a sane output.
-@end table
-
-Default value is -1.
-
+@item -copytb
+Copy input stream time base from input to output when stream copying.
 @item -shortest
 Finish encoding when the shortest input stream ends.
 @item -dts_delta_threshold
@@ -777,10 +996,10 @@ Set bitstream filters for matching streams. @var{bistream_filters} is
 a comma-separated list of bitstream filters. Use the @code{-bsfs} option
 to get the list of bitstream filters.
 @example
-ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264
+ffmpeg -i h264.mp4 -c:v copy -vbsf h264_mp4toannexb -an out.h264
 @end example
 @example
-ffmpeg -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt
+ffmpeg -i file.mov -an -vn -sbsf mov2textsub -c:s copy -f rawvideo sub.txt
 @end example
 
 @item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{per-stream})
@@ -792,44 +1011,6 @@ Specify Timecode for writing. @var{SEP} is ':' for non drop timecode and ';'
 @example
 ffmpeg -i input.mpg -timecode 01:02:03.04 -r 30000/1001 -s ntsc output.mpg
 @end example
-
-@item -filter_complex @var{filtergraph} (@emph{global})
-Define a complex filter graph, i.e. one with arbitrary number of inputs and/or
-outputs. For simple graphs -- those with one input and one output of the same
-type -- see the @option{-filter} options. @var{filtergraph} is a description of
-the filter graph, as described in @ref{Filtergraph syntax}.
-
-Input link labels must refer to input streams using the
-@code{[file_index:stream_specifier]} syntax (i.e. the same as @option{-map}
-uses). If @var{stream_specifier} matches multiple streams, the first one will be
-used. An unlabeled input will be connected to the first unused input stream of
-the matching type.
-
-Output link labels are referred to with @option{-map}. Unlabeled outputs are
-added to the first output file.
-
-For example, to overlay an image over video
-@example
-ffmpeg -i video.mkv -i image.png -filter_complex '[0:v][1:v]overlay[out]' -map
-'[out]' out.mkv
-@end example
-Here @code{[0:v]} refers to the first video stream in the first input file,
-which is linked to the first (main) input of the overlay filter. Similarly the
-first video stream in the second input is linked to the second (overlay) input
-of overlay.
-
-Assuming there is only one video stream in each input file, we can omit input
-labels, so the above is equivalent to
-@example
-ffmpeg -i video.mkv -i image.png -filter_complex 'overlay[out]' -map
-'[out]' out.mkv
-@end example
-
-Furthermore we can omit the output label and the single output from the filter
-graph will be added to the output file automatically, so we can simply write
-@example
-ffmpeg -i video.mkv -i image.png -filter_complex 'overlay' out.mkv
-@end example
 @end table
 
 @section Preset files
@@ -891,7 +1072,7 @@ frame rate or decrease the frame size.
 @item
 If your computer is not fast enough, you can speed up the
 compression at the expense of the compression ratio. You can use
-'-me zero' to speed up motion estimation, and '-g 0' to disable
+'-me zero' to speed up motion estimation, and '-intra' to disable
 motion estimation completely (you have only I-frames, which means it
 is about as good as JPEG compression).
 
@@ -1080,18 +1261,6 @@ composed of three digits padded with zeroes to express the sequence
 number. It is the same syntax supported by the C printf function, but
 only formats accepting a normal integer are suitable.
 
-When importing an image sequence, -i also supports expanding shell-like
-wildcard patterns (globbing) internally. To lower the chance of interfering
-with your actual file names and the shell's glob expansion, you are required
-to activate glob meta characters by prefixing them with a single @code{%}
-character, like in @code{foo-%*.jpeg}, @code{foo-%?%?%?.jpeg} or
-@code{foo-00%[234%]%*.jpeg}.
-If your filename actually contains a character sequence of a @code{%} character
-followed by a glob character, you must double the @code{%} character to escape
-it. Imagine your files begin with @code{%?-foo-}, then you could use a glob
-pattern like @code{%%?-foo-%*.jpeg}. For input patterns that could be both a
-printf or a glob pattern, ffmpeg will assume it is a glob pattern.
-
 @item
 You can put many streams of the same type in the output:
 
@@ -1102,23 +1271,9 @@ ffmpeg -i test1.avi -i test2.avi -map 0.3 -map 0.2 -map 0.1 -map 0.0 -c copy tes
 The resulting output file @file{test12.avi} will contain first four streams from
 the input file in reverse order.
 
-@item
-To force CBR video output:
-@example
-ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v
-@end example
-
-@item
-The four options lmin, lmax, mblmin and mblmax use 'lambda' units,
-but you may use the QP2LAMBDA constant to easily convert from 'q' units:
-@example
-ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
-@end example
-
 @end itemize
 @c man end EXAMPLES
 
-@include syntax.texi
 @include eval.texi
 @include decoders.texi
 @include encoders.texi

doc/ffmpeg.txt

@@ -29,7 +29,7 @@
                                                                           \                        /                                   :
                                                                     +======\======================/======+      ^                      :
                                                           ------> 0 |   : source_index  : st-:---        |      :                      :
-                    OutputFile output_files[]           /           +------------------------------------+      :                      :
+                    OuputFile output_files[]            /           +------------------------------------+      :                      :
                                                        /          1 |   :               :    :           |      :                      :
                  ^  +------+------------+-----+       /             +------------------------------------+      :                      :
                  :  |      : ost_index -:-----:------/            2 |   :               :    :           |      :                      :

doc/ffplay.texi

@@ -178,7 +178,6 @@ Seek to percentage in file corresponding to fraction of width.
 
 @c man end
 
-@include syntax.texi
 @include eval.texi
 @include decoders.texi
 @include demuxers.texi

doc/ffprobe.texi

@@ -106,11 +106,6 @@ stream.
 All the container format information is printed within a section with
 name "FORMAT".
 
-@item -show_format_entry @var{name}
-Like @option{-show_format}, but only prints the specified entry of the
-container format information, rather than all. This option may be given more
-than once, then all specified entries will be shown.
-
 @item -show_packets
 Show information about each packet contained in the input multimedia
 stream.
@@ -132,14 +127,6 @@ multimedia stream.
 Each media stream information is printed within a dedicated section
 with name "STREAM".
 
-@item -count_frames
-Count the number of frames per stream and report it in the
-corresponding stream section.
-
-@item -count_packets
-Count the number of packets per stream and report it in the
-corresponding stream section.
-
 @item -show_private_data, -private
 Show private data, that is data depending on the format of the
 particular shown element.
@@ -195,22 +182,6 @@ keyN=valN
 Metadata tags are printed as a line in the corresponding FORMAT or
 STREAM section, and are prefixed by the string "TAG:".
 
-This writer accepts options as a list of @var{key}=@var{value} pairs,
-separated by ":".
-
-A description of the accepted options follows.
-
-@table @option
-
-@item nokey, nk
-If set to 1 specify not to print the key of each field. Default value
-is 0.
-
-@item noprint_wrappers, nw
-If set to 1 specify not to print the section header and footer.
-Default value is 0.
-@end table
-
 @section compact
 Compact format.
 
@@ -294,10 +265,6 @@ XML based format.
 The XML output is described in the XML schema description file
 @file{ffprobe.xsd} installed in the FFmpeg datadir.
 
-An updated version of the schema can be retrieved at the url
-@url{http://www.ffmpeg.org/schema/ffprobe.xsd}, which redirects to the
-latest schema committed into the FFmpeg development source code tree.
-
 Note that the output issued will be compliant to the
 @file{ffprobe.xsd} schema only when no special global output options
 (@option{unit}, @option{prefix}, @option{byte_binary_prefix},
@@ -324,31 +291,26 @@ This option automatically sets @option{fully_qualified} to 1.
 
 For more information about the XML format, see
 @url{http://www.w3.org/XML/}.
-@c man end WRITERS
 
 @chapter Timecode
-@c man begin TIMECODE
 
 @command{ffprobe} supports Timecode extraction:
 
 @itemize
 
-@item
-MPEG1/2 timecode is extracted from the GOP, and is available in the video
+@item MPEG1/2 timecode is extracted from the GOP, and is available in the video
 stream details (@option{-show_streams}, see @var{timecode}).
 
-@item
-MOV timecode is extracted from tmcd track, so is available in the tmcd
+@item MOV timecode is extracted from tmcd track, so is available in the tmcd
 stream metadata (@option{-show_streams}, see @var{TAG:timecode}).
 
-@item
-DV and GXF timecodes are available in format metadata
+@item DV and GXF timecodes are available in format metadata
 (@option{-show_format}, see @var{TAG:timecode}).
 
 @end itemize
-@c man end TIMECODE
 
-@include syntax.texi
+@c man end WRITERS
+
 @include decoders.texi
 @include demuxers.texi
 @include protocols.texi

doc/ffprobe.xsd

@@ -110,10 +110,7 @@
       <xsd:attribute name="time_base"        type="xsd:string" use="required"/>
       <xsd:attribute name="start_time"       type="xsd:float"/>
       <xsd:attribute name="duration"         type="xsd:float"/>
-      <xsd:attribute name="bit_rate"         type="xsd:int"/>
       <xsd:attribute name="nb_frames"        type="xsd:int"/>
-      <xsd:attribute name="nb_read_frames"   type="xsd:int"/>
-      <xsd:attribute name="nb_read_packets"  type="xsd:int"/>
     </xsd:complexType>
 
     <xsd:complexType name="formatType">

doc/ffserver.conf

@@ -373,3 +373,5 @@ ACL allow 192.168.0.0 192.168.255.255
 <Redirect index.html>
 URL http://www.ffmpeg.org/
 </Redirect>
+
+

doc/filter_design.txt

@@ -1,116 +0,0 @@
-Filter design
-=============
-
-This document explains guidelines that should be observed (or ignored with
-good reason) when writing filters for libavfilter.
-
-In this document, the word “frame” indicates either a video frame or a group
-of audio samples, as stored in an AVFilterBuffer structure.
-
-
-Format negotiation
-==================
-
-  The query_formats method should set, for each input and each output links,
-  the list supported formats.
-
-  For video links, that means pixel format. For audio links, that means
-  channel layout, and sample format (the sample packing is implied by the
-  sample format).
-
-  The lists are not just lists, they are references to shared objects. When
-  the negotiation mechanism computes the intersection of the formats
-  supported at each ends of a link, all references to both lists are
-  replaced with a reference to the intersection. And when a single format is
-  eventually chosen for a link amongst the remaining list, again, all
-  references to the list are updated.
-
-  That means that if a filter requires that its input and output have the
-  same format amongst a supported list, all it have to do is use a reference
-  to the same list of formats.
-
-
-Buffer references ownership and permissions
-===========================================
-
-  TODO
-
-
-Frame scheduling
-================
-
-  The purpose of these rules is to ensure that frames flow in the filter
-  graph without getting stuck and accumulating somewhere.
-
-  Simple filters that output one frame for each input frame should not have
-  to worry about it.
-
-  start_frame / filter_samples
-  ----------------------------
-
-    These methods are called when a frame is pushed to the filter's input.
-    They can be called at any time except in a reentrant way.
-
-    If the input frame is enough to produce output, then the filter should
-    push the output frames on the output link immediately.
-
-    As an exception to the previous rule, if the input frame is enough to
-    produce several output frames, then the filter needs output only at
-    least one per link. The additional frames can be left buffered in the
-    filter; these buffered frames must be flushed immediately if a new input
-    produces new output.
-
-    (Example: framerate-doubling filter: start_frame must (1) flush the
-    second copy of the previous frame, if it is still there, (2) push the
-    first copy of the incoming frame, (3) keep the second copy for later.)
-
-    If the input frame is not enough to produce output, the filter must not
-    call request_frame to get more. It must just process the frame or queue
-    it. The task of requesting more frames is left to the filter's
-    request_frame method or the application.
-
-    If a filter has several inputs, the filter must be ready for frames
-    arriving randomly on any input. Therefore, any filter with several input
-    will most likely require some kind of queuing mechanism. It is perfectly
-    acceptable to have a limited queue and to drop frames when the inputs
-    are too unbalanced.
-
-  request_frame
-  -------------
-
-    This method is called when a frame is wanted on an output.
-
-    For an input, it should directly call start_frame or filter_samples on
-    the corresponding output.
-
-    For a filter, if there are queued frames already ready, one of these
-    frames should be pushed. If not, the filter should request a frame on
-    one of its input, repeatedly until at least one frame has been pushed.
-
-    Return values:
-    if request_frame could produce a frame, it should return 0;
-    if it could not for temporary reasons, it should return AVERROR(EAGAIN);
-    if it could not because there are no more frames, it should return
-    AVERROR_EOF.
-
-    The typical implementation of request_frame for a filter with several
-    inputs will look like that:
-
-        if (frames_queued) {
-            push_one_frame();
-            return 0;
-        }
-        while (!frame_pushed) {
-            input = input_where_a_frame_is_most_needed();
-            ret = avfilter_request_frame(input);
-            if (ret == AVERROR_EOF) {
-                process_eof_on_input();
-            } else if (ret < 0) {
-                return ret;
-            }
-        }
-        return 0;
-
-    Note that, except for filters that can have queued frames, request_frame
-    does not push frames: it requests them to its input, and as a reaction,
-    the start_frame / filter_samples method will be called and do the work.

doc/general.texi

@@ -13,8 +13,7 @@
 
 FFmpeg can be hooked up with a number of external libraries to add support
 for more formats. None of them are used by default, their use has to be
-explicitly requested by passing the appropriate flags to
-@command{./configure}.
+explicitly requested by passing the appropriate flags to @file{./configure}.
 
 @section OpenJPEG
 
@@ -144,8 +143,6 @@ library:
     @tab Multimedia format used by Delphine Software games.
 @item CD+G                      @tab   @tab X
     @tab Video format used by CD+G karaoke disks
-@item Commodore CDXL            @tab   @tab X
-    @tab Amiga CD video format
 @item Core Audio Format         @tab X @tab X
     @tab Apple Core Audio Format
 @item CRC testing format        @tab X @tab
@@ -211,7 +208,6 @@ library:
 @item MAXIS XA                  @tab   @tab X
     @tab Used in Sim City 3000; file extension .xa.
 @item MD Studio                 @tab   @tab X
-@item Metal Gear Solid: The Twin Snakes @tab @tab X
 @item Mobotix .mxg              @tab   @tab X
 @item Monkey's Audio            @tab   @tab X
 @item Motion Pixels MVI         @tab   @tab X
@@ -353,7 +349,6 @@ library:
 @item eXtended BINary text (XBIN) @tab @tab X
 @item YUV4MPEG pipe             @tab X @tab X
 @item Psygnosis YOP             @tab   @tab X
-@item ZeroCodec Lossless Video  @tab   @tab X
 @end multitable
 
 @code{X} means that encoding (resp. decoding) is supported.
@@ -373,8 +368,6 @@ following image formats are supported:
     @tab Microsoft BMP image
 @item DPX          @tab X @tab X
     @tab Digital Picture Exchange
-@item EXR          @tab   @tab X
-    @tab OpenEXR
 @item JPEG         @tab X @tab X
     @tab Progressive JPEG is not supported.
 @item JPEG 2000    @tab X @tab X
@@ -400,14 +393,12 @@ following image formats are supported:
     @tab V.Flash PTX format
 @item SGI          @tab X @tab X
     @tab SGI RGB image format
-@item Sun Rasterfile  @tab X @tab X
+@item Sun Rasterfile  @tab   @tab X
     @tab Sun RAS image format
 @item TIFF         @tab X @tab X
     @tab YUV, JPEG and some extension is not supported yet.
 @item Truevision Targa  @tab X @tab X
     @tab Targa (.TGA) image format
-@item XBM  @tab X @tab X
-    @tab X BitMap image format
 @item XWD  @tab X @tab X
     @tab X Window Dump image format
 @end multitable
@@ -453,8 +444,6 @@ following image formats are supported:
     @tab fourcc: AVrp
 @item AVS (Audio Video Standard) video  @tab     @tab  X
     @tab Video encoding used by the Creature Shock game.
-@item AYUV                   @tab  X  @tab  X
-    @tab Microsoft uncompressed packed 4:4:4:4
 @item Beam Software VB       @tab     @tab  X
 @item Bethesda VID video     @tab     @tab  X
     @tab Used in some games from Bethesda Softworks.
@@ -469,8 +458,6 @@ following image formats are supported:
     @tab fourcc: CSCD
 @item CD+G                   @tab     @tab  X
     @tab Video codec for CD+G karaoke disks
-@item CDXL                   @tab     @tab  X
-    @tab Amiga CD video codec
 @item Chinese AVS video      @tab  E  @tab  X
     @tab AVS1-P2, JiZhun profile, encoding through external library libxavs
 @item Delphine Software International CIN video  @tab     @tab  X
@@ -483,7 +470,7 @@ following image formats are supported:
 @item DFA                    @tab     @tab  X
     @tab Codec used in Chronomaster game.
 @item Dirac                  @tab  E  @tab  X
-    @tab supported through external library libschroedinger
+    @tab supported through external libdirac/libschroedinger libraries
 @item Deluxe Paint Animation @tab     @tab  X
 @item DNxHD                  @tab   X @tab  X
     @tab aka SMPTE VC3
@@ -504,13 +491,12 @@ following image formats are supported:
 @item Escape 124             @tab     @tab  X
 @item Escape 130             @tab     @tab  X
 @item FFmpeg video codec #1  @tab  X  @tab  X
-    @tab lossless codec (fourcc: FFV1)
+    @tab experimental lossless codec (fourcc: FFV1)
 @item Flash Screen Video v1  @tab  X  @tab  X
     @tab fourcc: FSV1
 @item Flash Screen Video v2  @tab  X  @tab  X
 @item Flash Video (FLV)      @tab  X  @tab  X
     @tab Sorenson H.263 used in Flash
-@item Forward Uncompressed   @tab     @tab  X
 @item Fraps                  @tab     @tab  X
 @item H.261                  @tab  X  @tab  X
 @item H.263 / H.263-1996     @tab  X  @tab  X
@@ -626,7 +612,6 @@ following image formats are supported:
 @item Ut Video               @tab     @tab  X
 @item v210 QuickTime uncompressed 4:2:2 10-bit     @tab  X  @tab  X
 @item v308 QuickTime uncompressed 4:4:4            @tab  X  @tab  X
-@item v408 QuickTime uncompressed 4:4:4:4          @tab  X  @tab  X
 @item v410 QuickTime uncompressed 4:4:4 10-bit     @tab  X  @tab  X
 @item VBLE Lossless Codec    @tab     @tab  X
 @item VMware Screen Codec / VMware Video  @tab     @tab  X
@@ -796,7 +781,6 @@ following image formats are supported:
     @tab Real 28800 bit/s codec
 @item RealAudio 3.0 (dnet)   @tab IX  @tab  X
     @tab Real low bitrate AC-3 codec
-@item RealAudio Lossless     @tab     @tab  X
 @item RealAudio SIPR / ACELP.NET @tab     @tab  X
 @item Shorten                @tab     @tab  X
 @item Sierra VMD audio       @tab     @tab  X
@@ -819,7 +803,6 @@ following image formats are supported:
 @item Westwood Audio (SND1)  @tab     @tab  X
 @item Windows Media Audio 1  @tab  X  @tab  X
 @item Windows Media Audio 2  @tab  X  @tab  X
-@item Windows Media Audio Lossless @tab  @tab  X
 @item Windows Media Audio Pro @tab    @tab  X
 @item Windows Media Audio Voice @tab  @tab  X
 @end multitable
@@ -838,8 +821,7 @@ performance on systems without hardware floating point support).
 @item SSA/ASS      @tab X @tab X @tab X @tab X
 @item DVB          @tab X @tab X @tab X @tab X
 @item DVD          @tab X @tab X @tab X @tab X
-@item JACOsub      @tab X @tab X @tab   @tab X
-@item MicroDVD     @tab X @tab X @tab   @tab X
+@item MicroDVD     @tab X @tab X @tab   @tab
 @item PGS          @tab   @tab   @tab   @tab X
 @item SubRip (SRT) @tab X @tab X @tab X @tab X
 @item XSUB         @tab   @tab   @tab X @tab X
@@ -872,12 +854,11 @@ performance on systems without hardware floating point support).
 @item ALSA              @tab X      @tab X
 @item BKTR              @tab X      @tab
 @item DV1394            @tab X      @tab
-@item Linux framebuffer @tab X      @tab
 @item JACK              @tab X      @tab
-@item LIBCDIO           @tab X
 @item LIBDC1394         @tab X      @tab
 @item OSS               @tab X      @tab X
 @item Pulseaudio        @tab X      @tab
+@item Video4Linux       @tab X      @tab
 @item Video4Linux2      @tab X      @tab
 @item VfW capture       @tab X      @tab
 @item X11 grabbing      @tab X      @tab
@@ -893,7 +874,7 @@ performance on systems without hardware floating point support).
 @item GXF               @tab X      @tab X
 @item MOV               @tab X      @tab
 @item MPEG1/2           @tab X      @tab X
-@item MXF               @tab X      @tab X
+@item MXF               @tab        @tab X
 @end multitable
 
 @bye

doc/git-howto.texi

@@ -65,14 +65,6 @@ git clone git@@source.ffmpeg.org:ffmpeg <target>
 This will put the FFmpeg sources into the directory @var{<target>} and let
 you push back your changes to the remote repository.
 
-Make sure that you do not have Windows line endings in your checkouts,
-otherwise you may experience spurious compilation failures. One way to
-achieve this is to run
-
-@example
-git config --global core.autocrlf false
-@end example
-
 
 @section Updating the source tree to the latest revision
 

doc/indevs.texi

@@ -59,7 +59,7 @@ BSD video input device.
 
 Windows DirectShow input device.
 
-DirectShow support is enabled when FFmpeg is built with the mingw-w64 project.
+DirectShow support is enabled when FFmpeg is built with mingw-w64.
 Currently only audio and video devices are supported.
 
 Multiple devices may be opened as separate inputs, but they may also be
@@ -504,9 +504,9 @@ command:
 ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
 @end example
 
-@section video4linux2
+@section video4linux and video4linux2
 
-Video4Linux2 input video device.
+Video4Linux and Video4Linux2 input video devices.
 
 The name of the device to grab is a file device node, usually Linux
 systems tend to automatically create such nodes when the device
@@ -514,28 +514,36 @@ systems tend to automatically create such nodes when the device
 kind @file{/dev/video@var{N}}, where @var{N} is a number associated to
 the device.
 
-Video4Linux2 devices usually support a limited set of
+Video4Linux and Video4Linux2 devices only support a limited set of
 @var{width}x@var{height} sizes and framerates. You can check which are
-supported using @command{-list_formats all} for Video4Linux2 devices.
+supported for example with the command @command{dov4l} for Video4Linux
+devices and using @command{-list_formats all} for Video4Linux2 devices.
 
-Some usage examples of the video4linux2 devices with ffmpeg and ffplay:
+If the size for the device is set to 0x0, the input device will
+try to auto-detect the size to use.
+Only for the video4linux2 device, if the frame rate is set to 0/0 the
+input device will use the frame rate value already set in the driver.
 
-The time base for the timestamps is 1 microsecond. Depending on the kernel
-version and configuration, the timestamps may be derived from the real time
-clock (origin at the Unix Epoch) or the monotonic clock (origin usually at
-boot time, unaffected by NTP or manual changes to the clock). The
-@option{-timestamps abs} or @option{-ts abs} option can be used to force
-conversion into the real time clock.
+Video4Linux support is deprecated since Linux 2.6.30, and will be
+dropped in later versions.
 
 Note that if FFmpeg is build with v4l-utils support ("--enable-libv4l2"
 option), it will always be used.
-@example
-# Grab and show the input of a video4linux2 device.
-ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0
 
-# Grab and record the input of a video4linux2 device, leave the
-framerate and size as previously set.
-ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg
+Follow some usage examples of the video4linux devices with the ff*
+tools.
+@example
+# Grab and show the input of a video4linux device, frame rate is set
+# to the default of 25/1.
+ffplay -s 320x240 -f video4linux /dev/video0
+
+# Grab and show the input of a video4linux2 device, auto-adjust size.
+ffplay -f video4linux2 /dev/video0
+
+# Grab and record the input of a video4linux2 device, auto-adjust size,
+# frame rate value defaults to 0/0 so it is read from the video4linux2
+# driver.
+ffmpeg -f video4linux2 -i /dev/video0 out.mpeg
 @end example
 
 "v4l" and "v4l2" can be used as aliases for the respective "video4linux" and

doc/issue_tracker.txt

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

doc/muxers.texi

@@ -56,37 +56,31 @@ See also the @ref{framecrc} muxer.
 @anchor{framecrc}
 @section framecrc
 
-Per-packet CRC (Cyclic Redundancy Check) testing format.
+Per-frame CRC (Cyclic Redundancy Check) testing format.
 
-This muxer computes and prints the Adler-32 CRC for each audio
-and video packet. By default audio frames are converted to signed
+This muxer computes and prints the Adler-32 CRC for each decoded audio
+and video frame. By default audio frames are converted to signed
 16-bit raw audio and video frames to raw video before computing the
 CRC.
 
 The output of the muxer consists of a line for each audio and video
-packet of the form:
-@example
-@var{stream_index}, @var{packet_dts}, @var{packet_pts}, @var{packet_duration}, @var{packet_size}, 0x@var{CRC}
-@end example
+frame of the form: @var{stream_index}, @var{frame_dts},
+@var{frame_size}, 0x@var{CRC}, where @var{CRC} is a hexadecimal
+number 0-padded to 8 digits containing the CRC of the decoded frame.
 
-@var{CRC} is a hexadecimal number 0-padded to 8 digits containing the
-CRC of the packet.
-
-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}:
+For example to compute the CRC of each decoded frame in the input, and
+store it in the file @file{out.crc}:
 @example
 ffmpeg -i INPUT -f framecrc out.crc
 @end example
 
-To print the information to stdout, use the command:
+You can print the CRC of each decoded frame to stdout with the command:
 @example
 ffmpeg -i INPUT -f framecrc -
 @end example
 
-With @command{ffmpeg}, you can select the output format to which the
-audio and video frames are encoded before computing the CRC for each
-packet by specifying the audio and video codec. For example, to
+You can select the output format of each frame with @command{ffmpeg} by
+specifying the audio and video codec and format. For example, to
 compute the CRC of each decoded input audio frame converted to PCM
 unsigned 8-bit and of each decoded input video frame converted to
 MPEG-2 video, use the command:
@@ -96,39 +90,6 @@ ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc -
 
 See also the @ref{crc} muxer.
 
-@anchor{framemd5}
-@section framemd5
-
-Per-packet MD5 testing format.
-
-This muxer computes and prints the MD5 hash for each audio
-and video packet. By default audio frames are converted to signed
-16-bit raw audio and video frames to raw video before computing the
-hash.
-
-The output of the muxer consists of a line for each audio and video
-packet of the form:
-@example
-@var{stream_index}, @var{packet_dts}, @var{packet_pts}, @var{packet_duration}, @var{packet_size}, @var{MD5}
-@end example
-
-@var{MD5} is a hexadecimal number representing the computed MD5 hash
-for the packet.
-
-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}:
-@example
-ffmpeg -i INPUT -f framemd5 out.md5
-@end example
-
-To print the information to stdout, use the command:
-@example
-ffmpeg -i INPUT -f framemd5 -
-@end example
-
-See also the @ref{md5} muxer.
-
 @anchor{image2}
 @section image2
 
@@ -187,104 +148,18 @@ each of the YUV420P components. To read or write this image file format,
 specify the name of the '.Y' file. The muxer will automatically open the
 '.U' and '.V' files as required.
 
-@anchor{md5}
-@section md5
+@section mov
 
-MD5 testing format.
+MOV / MP4 muxer
 
-This muxer computes and prints the MD5 hash of all the input audio
-and video frames. By default audio frames are converted to signed
-16-bit raw audio and video frames to raw video before computing the
-hash.
-
-The output of the muxer consists of a single line of the form:
-MD5=@var{MD5}, where @var{MD5} is a hexadecimal number representing
-the computed MD5 hash.
-
-For example to compute the MD5 hash of the input converted to raw
-audio and video, and store it in the file @file{out.md5}:
-@example
-ffmpeg -i INPUT -f md5 out.md5
-@end example
-
-You can print the MD5 to stdout with the command:
-@example
-ffmpeg -i INPUT -f md5 -
-@end example
-
-See also the @ref{framemd5} muxer.
-
-@section MOV/MP4/ISMV
-
-The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4
-file has all the metadata about all packets stored in one location
-(written at the end of the file, it can be moved to the start for
-better playback using the @command{qt-faststart} tool). A fragmented
-file consists of a number of fragments, where packets and metadata
-about these packets are stored together. Writing a fragmented
-file has the advantage that the file is decodable even if the
-writing is interrupted (while a normal MOV/MP4 is undecodable if
-it is not properly finished), and it requires less memory when writing
-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.
-
-Fragmentation is enabled by setting one of the AVOptions that define
-how to cut the file into fragments:
+The muxer options are:
 
 @table @option
 @item -moov_size @var{bytes}
 Reserves space for the moov atom at the beginning of the file instead of placing the
 moov atom at the end. If the space reserved is insufficient, muxing will fail.
-@item -movflags frag_keyframe
-Start a new fragment at each video keyframe.
-@item -frag_duration @var{duration}
-Create fragments that are @var{duration} microseconds long.
-@item -frag_size @var{size}
-Create fragments that contain up to @var{size} bytes of payload data.
-@item -movflags frag_custom
-Allow the caller to manually choose when to cut fragments, by
-calling @code{av_write_frame(ctx, NULL)} to write a fragment with
-the packets written so far. (This is only useful with other
-applications integrating libavformat, not from @command{ffmpeg}.)
-@item -min_frag_duration @var{duration}
-Don't create fragments that are shorter than @var{duration} microseconds long.
 @end table
 
-If more than one condition is specified, fragments are cut when
-one of the specified conditions is fulfilled. The exception to this is
-@code{-min_frag_duration}, which has to be fulfilled for any of the other
-conditions to apply.
-
-Additionally, the way the output file is written can be adjusted
-through a few other options:
-
-@table @option
-@item -movflags empty_moov
-Write an initial moov atom directly at the start of the file, without
-describing any samples in it. Generally, an mdat/moov pair is written
-at the start of the file, as a normal MOV/MP4 file, containing only
-a short portion of the file. With this option set, there is no initial
-mdat atom, and the moov atom only describes the tracks but has
-a zero duration.
-
-Files written with this option set do not work in QuickTime.
-This option is implicitly set when writing ismv (Smooth Streaming) files.
-@item -movflags separate_moof
-Write a separate moof (movie fragment) atom for each track. Normally,
-packets for all tracks are written in a moof atom (which is slightly
-more efficient), but with this option set, the muxer writes one moof/mdat
-pair for each track, making it easier to separate tracks.
-
-This option is implicitly set when writing ismv (Smooth Streaming) files.
-@end table
-
-Smooth Streaming content can be pushed in real time to a publishing
-point on IIS with this muxer. Example:
-@example
-ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1)
-@end example
-
 @section mpegts
 
 MPEG transport stream muxer.
@@ -435,47 +310,11 @@ Set segment duration to @var{t} seconds.
 Generate also a listfile named @var{name}.
 @item segment_list_size @var{size}
 Overwrite the listfile once it reaches @var{size} entries.
-@item segment_wrap @var{limit}
-Wrap around segment index once it reaches @var{limit}.
 @end table
 
 @example
 ffmpeg -i in.mkv -c copy -map 0 -f segment -list out.list out%03d.nut
 @end example
 
-@section mp3
-
-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.
-
-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 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
-can be any number of those streams, each will correspond to a single APIC frame.
-The stream metadata tags @var{title} and @var{comment} map to APIC
-@var{description} and @var{picture type} respectively. See
-@url{http://id3.org/id3v2.4.0-frames} for allowed picture types.
-
-Note that the APIC frames must be written at the beginning, so the muxer will
-buffer the audio frames until it gets all the pictures. It is therefore advised
-to provide the pictures as soon as possible to avoid excessive buffering.
-
-Examples:
-
-Write an mp3 with an ID3v2.3 header and an ID3v1 footer:
-@example
-ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3
-@end example
-
-Attach a picture to an mp3:
-@example
-ffmpeg -i input.mp3 -i cover.png -c copy -metadata:s:v title="Album cover"
--metadata:s:v comment="Cover (Front)" out.mp3
-@end example
 
 @c man end MUXERS

doc/platform.texi

@@ -27,11 +27,11 @@ to configure.
 @section BSD
 
 BSD make will not build FFmpeg, you need to install and use GNU Make
-(@command{gmake}).
+(@file{gmake}).
 
 @section (Open)Solaris
 
-GNU Make is required to build FFmpeg, so you have to invoke (@command{gmake}),
+GNU Make is required to build FFmpeg, so you have to invoke (@file{gmake}),
 standard Solaris Make will not work. When building with a non-c99 front-end
 (gcc, generic suncc) add either @code{--extra-libs=/usr/lib/values-xpg6.o}
 or @code{--extra-libs=/usr/lib/64/values-xpg6.o} to the configure options
@@ -89,7 +89,7 @@ section and the FAQ.
 
 FFmpeg does not build out-of-the-box with the packages the automated MinGW
 installer provides. It also requires coreutils to be installed and many other
-packages updated to the latest version. The minimum versions for some packages
+packages updated to the latest version. The minimum version for some packages
 are listed below:
 
 @itemize
@@ -109,11 +109,14 @@ Notes:
 @item Building natively using MSYS can be sped up by disabling implicit rules
 in the Makefile by calling @code{make -r} instead of plain @code{make}. This
 speed up is close to non-existent for normal one-off builds and is only
-noticeable when running make for a second time (for example during
+noticeable when running make for a second time (for example in
 @code{make install}).
 
 @item In order to compile FFplay, you must have the MinGW development library
-of @uref{http://www.libsdl.org/, SDL} and @code{pkg-config} installed.
+of @uref{http://www.libsdl.org/, SDL}.
+Edit the @file{bin/sdl-config} script so that it points to the correct prefix
+where SDL was installed. Verify that @file{sdl-config} can be launched from
+the MSYS command line.
 
 @item By using @code{./configure --enable-shared} when configuring FFmpeg,
 you can build the FFmpeg libraries (e.g. libavutil, libavcodec,
@@ -137,7 +140,7 @@ you might have to modify the procedures slightly.
 
 @subsection Using static libraries
 
-Assuming you have just built and installed FFmpeg in @file{/usr/local}:
+Assuming you have just built and installed FFmpeg in @file{/usr/local}.
 
 @enumerate
 
@@ -272,7 +275,7 @@ To create import libraries that work with the @code{/OPT:REF} option
 
 @enumerate
 
-@item Open @emph{Visual Studio 2005 Command Prompt}.
+@item Open @file{Visual Studio 2005 Command Prompt}.
 
 Alternatively, in a normal command line prompt, call @file{vcvars32.bat}
 which sets up the environment variables for the Visual C++ tools
@@ -282,14 +285,17 @@ which sets up the environment variables for the Visual C++ tools
 @item Enter the @file{bin} directory where the created LIB and DLL files
 are stored.
 
-@item Generate new import libraries with @command{lib.exe}:
+@item Generate new import libraries with @file{lib.exe}:
 
 @example
-lib /machine:i386 /def:..\lib\foo-version.def  /out:foo.lib
+lib /machine:i386 /def:..\lib\avcodec-53.def  /out:avcodec.lib
+lib /machine:i386 /def:..\lib\avdevice-53.def /out:avdevice.lib
+lib /machine:i386 /def:..\lib\avfilter-2.def  /out:avfilter.lib
+lib /machine:i386 /def:..\lib\avformat-53.def /out:avformat.lib
+lib /machine:i386 /def:..\lib\avutil-51.def   /out:avutil.lib
+lib /machine:i386 /def:..\lib\swscale-2.def   /out:swscale.lib
 @end example
 
-Replace @code{foo-version} and @code{foo} with the respective library names.
-
 @end enumerate
 
 @anchor{Cross compilation for Windows with Linux}
@@ -331,8 +337,8 @@ Then run
 
 to make a static build.
 
-To build shared libraries add a special compiler flag to work around current
-@code{gcc4-core} package bugs in addition to the normal configure flags:
+The current @code{gcc4-core} package is buggy and needs this flag to build
+shared libraries:
 
 @example
 ./configure --enable-shared --disable-static --extra-cflags=-fno-reorder-functions
@@ -348,12 +354,16 @@ These library packages are only available from
 @uref{http://sourceware.org/cygwinports/, Cygwin Ports}:
 
 @example
-yasm, libSDL-devel, libfaac-devel, libaacplus-devel, libgsm-devel, libmp3lame-devel,
-libschroedinger1.0-devel, speex-devel, libtheora-devel, libxvidcore-devel
+yasm, libSDL-devel, libdirac-devel, libfaac-devel, libaacplus-devel, libgsm-devel,
+libmp3lame-devel, libschroedinger1.0-devel, speex-devel, libtheora-devel,
+libxvidcore-devel
 @end example
 
-The recommendation for x264 is to build it from source, as it evolves too
-quickly for Cygwin Ports to be up to date.
+The recommendation for libnut and x264 is to build them from source by
+yourself, as they evolve too quickly for Cygwin Ports to be up to date.
+
+Cygwin 1.7.x has IPv6 support. You can add IPv6 to Cygwin 1.5.x by means
+of the @code{libgetaddrinfo-devel} package, available at Cygwin Ports.
 
 @section Crosscompilation for Windows under Cygwin
 

doc/print_options.c

@@ -1,125 +0,0 @@
-/*
- * Copyright (c) 2012 Anton Khirnov
- *
- * This file is part of Libav.
- *
- * Libav is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * Libav is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/*
- * generate texinfo manpages for avoptions
- */
-
-#include <stddef.h>
-#include <string.h>
-#include <float.h>
-
-#include "libavformat/avformat.h"
-#include "libavcodec/avcodec.h"
-#include "libavutil/opt.h"
-
-static void print_usage(void)
-{
-    fprintf(stderr, "Usage: enum_options type\n"
-            "type: format codec\n");
-    exit(1);
-}
-
-static void print_option(const AVOption *opts, const AVOption *o, int per_stream)
-{
-    printf("@item -%s%s @var{", o->name, per_stream ? "[:stream_specifier]" : "");
-    switch (o->type) {
-    case AV_OPT_TYPE_BINARY:   printf("hexadecimal string"); break;
-    case AV_OPT_TYPE_STRING:   printf("string");             break;
-    case AV_OPT_TYPE_INT:
-    case AV_OPT_TYPE_INT64:    printf("integer");            break;
-    case AV_OPT_TYPE_FLOAT:
-    case AV_OPT_TYPE_DOUBLE:   printf("float");              break;
-    case AV_OPT_TYPE_RATIONAL: printf("rational number");    break;
-    case AV_OPT_TYPE_FLAGS:    printf("flags");              break;
-    default:                   printf("value");              break;
-    }
-    printf("} (@emph{");
-
-    if (o->flags & AV_OPT_FLAG_DECODING_PARAM) {
-        printf("input");
-        if (o->flags & AV_OPT_FLAG_ENCODING_PARAM)
-            printf("/");
-    }
-    if (o->flags & AV_OPT_FLAG_ENCODING_PARAM) printf("output");
-    if (o->flags & AV_OPT_FLAG_AUDIO_PARAM)    printf(",audio");
-    if (o->flags & AV_OPT_FLAG_VIDEO_PARAM)    printf(",video");
-    if (o->flags & AV_OPT_FLAG_SUBTITLE_PARAM) printf(",subtitles");
-
-    printf("})\n");
-    if (o->help)
-        printf("%s\n", o->help);
-
-    if (o->unit) {
-        const AVOption *u;
-        printf("\nPossible values:\n@table @samp\n");
-
-        for (u = opts; u->name; u++) {
-            if (u->type == AV_OPT_TYPE_CONST && u->unit && !strcmp(u->unit, o->unit))
-                printf("@item %s\n%s\n", u->name, u->help ? u->help : "");
-        }
-        printf("@end table\n");
-    }
-}
-
-static void show_opts(const AVOption *opts, int per_stream)
-{
-    const AVOption *o;
-
-    printf("@table @option\n");
-    for (o = opts; o->name; o++) {
-        if (o->type != AV_OPT_TYPE_CONST)
-            print_option(opts, o, per_stream);
-    }
-    printf("@end table\n");
-}
-
-static void show_format_opts(void)
-{
-#include "libavformat/options_table.h"
-
-    printf("@section Format AVOptions\n");
-    show_opts(options, 0);
-}
-
-static void show_codec_opts(void)
-{
-#include "libavcodec/options_table.h"
-
-    printf("@section Codec AVOptions\n");
-    show_opts(options, 1);
-}
-
-int main(int argc, char **argv)
-{
-    if (argc < 2)
-        print_usage();
-
-    printf("@c DO NOT EDIT THIS FILE!\n"
-           "@c It was generated by print_options.\n\n");
-    if (!strcmp(argv[1], "format"))
-        show_format_opts();
-    else if (!strcmp(argv[1], "codec"))
-        show_codec_opts();
-    else
-        print_usage();
-
-    return 0;
-}

doc/protocols.texi

@@ -19,34 +19,20 @@ supported protocols.
 
 A description of the currently available protocols follows.
 
-@section bluray
+@section applehttp
 
-Read BluRay playlist.
+Read Apple HTTP Live Streaming compliant segmented stream as
+a uniform one. The M3U8 playlists describing the segments can be
+remote HTTP resources or local files, accessed using the standard
+file protocol.
+HTTP is default, specific protocol can be declared by specifying
+"+@var{proto}" after the applehttp URI scheme name, where @var{proto}
+is either "file" or "http".
 
-The accepted options are:
-@table @option
-
-@item angle
-BluRay angle
-
-@item chapter
-Start chapter (1...N)
-
-@item playlist
-Playlist to read (BDMV/PLAYLIST/?????.mpls)
-
-@end table
-
-Examples:
-
-Read longest playlist from BluRay mounted to /mnt/bluray:
 @example
-bluray:/mnt/bluray
-@end example
-
-Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
-@example
--playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
+applehttp://host/path/to/remote/resource.m3u8
+applehttp+http://host/path/to/remote/resource.m3u8
+applehttp+file://path/to/local/resource.m3u8
 @end example
 
 @section concat
@@ -95,26 +81,6 @@ specified with the name "FILE.mpeg" is interpreted as the URL
 
 Gopher protocol.
 
-@section hls
-
-Read Apple HTTP Live Streaming compliant segmented stream as
-a uniform one. The M3U8 playlists describing the segments can be
-remote HTTP resources or local files, accessed using the standard
-file protocol.
-The nested protocol is declared by specifying
-"+@var{proto}" after the hls URI scheme name, where @var{proto}
-is either "file" or "http".
-
-@example
-hls+http://host/path/to/remote/resource.m3u8
-hls+file://path/to/local/resource.m3u8
-@end example
-
-Using this protocol is discouraged - the hls demuxer should work
-just as well (if not, please report the issues) and is more complete.
-To use the hls demuxer instead, simply use the direct URLs to the
-m3u8 files.
-
 @section http
 
 HTTP (Hyper Text Transfer Protocol).
@@ -318,9 +284,9 @@ Accept packets only from negotiated peer address and port.
 @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
-the @code{max_delay} field of AVFormatContext).
+(since they may arrive out of order, or packets may get lost totally). In
+order for this to be enabled, a maximum delay must be specified in the
+@code{max_delay} field of AVFormatContext.
 
 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
 streams to display can be chosen with @code{-vst} @var{n} and

doc/swresample.txt

@@ -32,7 +32,7 @@ Special Converter         v
                         Output
 
 Planar/Packed convertion is done when needed during sample format convertion
-Every step can be skipped without memcpy when its not needed.
+Every step can be skiped without memcpy when its not needed.
 Either Resampling and Rematrixing can be performed first depending on which
 way its faster.
 The Buffers are needed for resampling due to resamplng being a process that

doc/swscale.txt

@@ -96,3 +96,4 @@ would benefit from it.
 Also, as already hinted at, initFilter() accepts an optional convolutional
 filter as input that can be used for contrast, saturation, blur, sharpening
 shift, chroma vs. luma shift, ...
+

doc/syntax.texi

@@ -1,158 +0,0 @@
-@chapter Syntax
-@c man begin SYNTAX
-
-When evaluating specific formats, FFmpeg uses internal library parsing
-functions, shared by the tools. This section documents the syntax of
-some of these formats.
-
-@anchor{date syntax}
-@section Date
-
-The accepted syntax is:
-@example
-[(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z]
-now
-@end example
-
-If the value is "now" it takes the current time.
-
-Time is local time unless 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.
-
-@anchor{time duration syntax}
-@section Time duration
-
-The accepted syntax is:
-@example
-[-]HH:MM:SS[.m...]
-[-]S+[.m...]
-@end example
-
-@var{HH} expresses the number of hours, @var{MM} the number a of minutes
-and @var{SS} the number of seconds.
-
-@anchor{video size syntax}
-@section Video size
-Specify the size of the sourced video, it may be a string of the form
-@var{width}x@var{height}, or the name of a size abbreviation.
-
-The following abbreviations are recognized:
-@table @samp
-@item sqcif
-128x96
-@item qcif
-176x144
-@item cif
-352x288
-@item 4cif
-704x576
-@item 16cif
-1408x1152
-@item qqvga
-160x120
-@item qvga
-320x240
-@item vga
-640x480
-@item svga
-800x600
-@item xga
-1024x768
-@item uxga
-1600x1200
-@item qxga
-2048x1536
-@item sxga
-1280x1024
-@item qsxga
-2560x2048
-@item hsxga
-5120x4096
-@item wvga
-852x480
-@item wxga
-1366x768
-@item wsxga
-1600x1024
-@item wuxga
-1920x1200
-@item woxga
-2560x1600
-@item wqsxga
-3200x2048
-@item wquxga
-3840x2400
-@item whsxga
-6400x4096
-@item whuxga
-7680x4800
-@item cga
-320x200
-@item ega
-640x350
-@item hd480
-852x480
-@item hd720
-1280x720
-@item hd1080
-1920x1080
-@end table
-
-@anchor{video rate syntax}
-@section Video rate
-
-Specify the frame rate of a video, expressed as the number of frames
-generated per second. It has to be a string in the format
-@var{frame_rate_num}/@var{frame_rate_den}, an integer number, a float
-number or a valid video frame rate abbreviation.
-
-The following abbreviations are recognized:
-@table @samp
-@item ntsc
-30000/1001
-@item pal
-25/1
-@item qntsc
-30000/1
-@item qpal
-25/1
-@item sntsc
-30000/1
-@item spal
-25/1
-@item film
-24/1
-@item ntsc-film
-24000/1
-@end table
-
-@anchor{ratio syntax}
-@section Ratio
-
-A ratio can be expressed as an expression, or in the form
-@var{numerator}:@var{denominator}.
-
-Note that a ratio with infinite (1/0) or negative value is
-considered valid, so you should check on the returned value if you
-want to exclude those values.
-
-The undefined value can be expressed using the "0:0" string.
-
-@anchor{color syntax}
-@section Color
-
-It can be the name of a color (case insensitive match) or a
-[0x|#]RRGGBB[AA] sequence, possibly followed by "@" and a string
-representing the alpha component.
-
-The alpha component may be a string composed by "0x" followed by an
-hexadecimal number or a decimal number between 0.0 and 1.0, which
-represents the opacity value (0x00/0.0 means completely transparent,
-0xff/1.0 completely opaque).
-If the alpha component is not specified then 0xff is assumed.
-
-The string "random" will result in a random color.
-
-@c man end SYNTAX

doc/texi2pod.pl

@@ -36,7 +36,7 @@ $shift = "";
 %defs = ();
 $fnno = 1;
 $inf = "";
-@ibase = ();
+$ibase = "";
 
 while ($_ = shift) {
     if (/^-D(.*)$/) {
@@ -52,8 +52,6 @@ while ($_ = shift) {
         die "flags may only contain letters, digits, hyphens, dashes and underscores\n"
             unless $flag =~ /^[a-zA-Z0-9_-]+$/;
         $defs{$flag} = $value;
-    } elsif (/^-I(.*)$/) {
-        push @ibase, $1 ne "" ? $1 : shift;
     } elsif (/^-/) {
         usage();
     } else {
@@ -63,12 +61,10 @@ while ($_ = shift) {
     }
 }
 
-push @ibase, ".";
-
 if (defined $in) {
     $inf = gensym();
     open($inf, "<$in") or die "opening \"$in\": $!\n";
-    push @ibase, $1 if $in =~ m|^(.+)/[^/]+$|;
+    $ibase = $1 if $in =~ m|^(.+)/[^/]+$|;
 } else {
     $inf = \*STDIN;
 }
@@ -78,7 +74,7 @@ if (defined $out) {
 }
 
 while(defined $inf) {
-INF: while(<$inf>) {
+while(<$inf>) {
     # Certain commands are discarded without further processing.
     /^\@(?:
          [a-z]+index            # @*index: useful only in complete manual
@@ -108,10 +104,11 @@ INF: while(<$inf>) {
         push @instack, $inf;
         $inf = gensym();
 
-        for (@ibase) {
-            open($inf, "<" . $_ . "/" . $1) and next INF;
-        }
-        die "cannot open $1: $!\n";
+        # Try cwd and $ibase.
+        open($inf, "<" . $1)
+            or open($inf, "<" . $ibase . "/" . $1)
+                or die "cannot open $1 or $ibase/$1: $!\n";
+        next;
     };
 
     # Look for blocks surrounded by @c man begin SECTION ... @c man end.

doc/viterbi.txt

@@ -107,3 +107,4 @@ one with score 3)
 
 Author: Michael niedermayer
 Copyright LGPL
+

ffplay.c

@@ -40,6 +40,7 @@
 #include "libavformat/avformat.h"
 #include "libavdevice/avdevice.h"
 #include "libswscale/swscale.h"
+#include "libavcodec/audioconvert.h"
 #include "libavutil/opt.h"
 #include "libavcodec/avfft.h"
 #include "libswresample/swresample.h"
@@ -63,6 +64,7 @@ const char program_name[] = "ffplay";
 const int program_birth_year = 2003;
 
 #define MAX_QUEUE_SIZE (15 * 1024 * 1024)
+#define MIN_AUDIOQ_SIZE (20 * 16 * 1024)
 #define MIN_FRAMES 5
 
 /* SDL audio buffer size, in samples. Should be small to have precise
@@ -104,7 +106,6 @@ typedef struct VideoPicture {
     int skip;
     SDL_Overlay *bmp;
     int width, height; /* source height & width */
-    AVRational sample_aspect_ratio;
     int allocated;
     int reallocate;
     enum PixelFormat pix_fmt;
@@ -132,7 +133,6 @@ typedef struct VideoState {
     AVInputFormat *iformat;
     int no_background;
     int abort_request;
-    int force_refresh;
     int paused;
     int last_paused;
     int seek_req;
@@ -232,14 +232,8 @@ typedef struct VideoState {
 #endif
 
     int refresh;
-    int last_video_stream, last_audio_stream, last_subtitle_stream;
 } VideoState;
 
-typedef struct AllocEventProps {
-    VideoState *is;
-    AVFrame *frame;
-} AllocEventProps;
-
 static int opt_help(const char *opt, const char *arg);
 
 /* options specified by the user */
@@ -304,12 +298,13 @@ void av_noreturn exit_program(int ret)
     exit(ret);
 }
 
-static int packet_queue_put_private(PacketQueue *q, AVPacket *pkt)
+static int packet_queue_put(PacketQueue *q, AVPacket *pkt)
 {
     AVPacketList *pkt1;
 
-    if (q->abort_request)
-       return -1;
+    /* duplicate the packet */
+    if (pkt != &flush_pkt && av_dup_packet(pkt) < 0)
+        return -1;
 
     pkt1 = av_malloc(sizeof(AVPacketList));
     if (!pkt1)
@@ -317,7 +312,11 @@ static int packet_queue_put_private(PacketQueue *q, AVPacket *pkt)
     pkt1->pkt = *pkt;
     pkt1->next = NULL;
 
+
+    SDL_LockMutex(q->mutex);
+
     if (!q->last_pkt)
+
         q->first_pkt = pkt1;
     else
         q->last_pkt->next = pkt1;
@@ -326,25 +325,9 @@ static int packet_queue_put_private(PacketQueue *q, AVPacket *pkt)
     q->size += pkt1->pkt.size + sizeof(*pkt1);
     /* XXX: should duplicate packet data in DV case */
     SDL_CondSignal(q->cond);
-    return 0;
-}
 
-static int packet_queue_put(PacketQueue *q, AVPacket *pkt)
-{
-    int ret;
-
-    /* duplicate the packet */
-    if (pkt != &flush_pkt && av_dup_packet(pkt) < 0)
-        return -1;
-
-    SDL_LockMutex(q->mutex);
-    ret = packet_queue_put_private(q, pkt);
     SDL_UnlockMutex(q->mutex);
-
-    if (pkt != &flush_pkt && ret < 0)
-        av_free_packet(pkt);
-
-    return ret;
+    return 0;
 }
 
 /* packet queue handling */
@@ -353,7 +336,7 @@ static void packet_queue_init(PacketQueue *q)
     memset(q, 0, sizeof(PacketQueue));
     q->mutex = SDL_CreateMutex();
     q->cond = SDL_CreateCond();
-    q->abort_request = 1;
+    packet_queue_put(q, &flush_pkt);
 }
 
 static void packet_queue_flush(PacketQueue *q)
@@ -373,7 +356,7 @@ static void packet_queue_flush(PacketQueue *q)
     SDL_UnlockMutex(q->mutex);
 }
 
-static void packet_queue_destroy(PacketQueue *q)
+static void packet_queue_end(PacketQueue *q)
 {
     packet_queue_flush(q);
     SDL_DestroyMutex(q->mutex);
@@ -391,14 +374,6 @@ static void packet_queue_abort(PacketQueue *q)
     SDL_UnlockMutex(q->mutex);
 }
 
-static void packet_queue_start(PacketQueue *q)
-{
-    SDL_LockMutex(q->mutex);
-    q->abort_request = 0;
-    packet_queue_put_private(q, &flush_pkt);
-    SDL_UnlockMutex(q->mutex);
-}
-
 /* return < 0 if aborted, 0 if no packet and > 0 if packet.  */
 static int packet_queue_get(PacketQueue *q, AVPacket *pkt, int block)
 {
@@ -692,11 +667,21 @@ static void video_image_display(VideoState *is)
 
     vp = &is->pictq[is->pictq_rindex];
     if (vp->bmp) {
-        if (vp->sample_aspect_ratio.num == 0)
-            aspect_ratio = 0;
-        else
-            aspect_ratio = av_q2d(vp->sample_aspect_ratio);
+#if CONFIG_AVFILTER
+         if (vp->picref->video->sample_aspect_ratio.num == 0)
+             aspect_ratio = 0;
+         else
+             aspect_ratio = av_q2d(vp->picref->video->sample_aspect_ratio);
+#else
 
+        /* XXX: use variable in the frame */
+        if (is->video_st->sample_aspect_ratio.num)
+            aspect_ratio = av_q2d(is->video_st->sample_aspect_ratio);
+        else if (is->video_st->codec->sample_aspect_ratio.num)
+            aspect_ratio = av_q2d(is->video_st->codec->sample_aspect_ratio);
+        else
+            aspect_ratio = 0;
+#endif
         if (aspect_ratio <= 0.0)
             aspect_ratio = 1.0;
         aspect_ratio *= (float)vp->width / (float)vp->height;
@@ -882,8 +867,7 @@ static void video_audio_display(VideoState *s)
             }
         }
         SDL_UpdateRect(screen, s->xpos, s->ytop, 1, s->height);
-        if (!s->paused)
-            s->xpos++;
+        s->xpos++;
         if (s->xpos >= s->width)
             s->xpos= s->xleft;
     }
@@ -897,9 +881,6 @@ static void stream_close(VideoState *is)
     is->abort_request = 1;
     SDL_WaitThread(is->read_tid, NULL);
     SDL_WaitThread(is->refresh_tid, NULL);
-    packet_queue_destroy(&is->videoq);
-    packet_queue_destroy(&is->audioq);
-    packet_queue_destroy(&is->subtitleq);
 
     /* free all pictures */
     for (i = 0; i < VIDEO_PICTURE_QUEUE_SIZE; i++) {
@@ -953,7 +934,6 @@ static int video_open(VideoState *is, int force_set_video_mode)
 {
     int flags = SDL_HWSURFACE | SDL_ASYNCBLIT | SDL_HWACCEL;
     int w,h;
-    VideoPicture *vp = &is->pictq[is->pictq_rindex];
 
     if (is_full_screen) flags |= SDL_FULLSCREEN;
     else                flags |= SDL_RESIZABLE;
@@ -964,9 +944,15 @@ static int video_open(VideoState *is, int force_set_video_mode)
     } else if (!is_full_screen && screen_width) {
         w = screen_width;
         h = screen_height;
-    } else if (vp->width) {
-        w = vp->width;
-        h = vp->height;
+#if CONFIG_AVFILTER
+    } else if (is->out_video_filter && is->out_video_filter->inputs[0]) {
+        w = is->out_video_filter->inputs[0]->w;
+        h = is->out_video_filter->inputs[0]->h;
+#else
+    } else if (is->video_st && is->video_st->codec->width) {
+        w = is->video_st->codec->width;
+        h = is->video_st->codec->height;
+#endif
     } else {
         w = 640;
         h = 480;
@@ -1007,7 +993,7 @@ static int refresh_thread(void *opaque)
         SDL_Event event;
         event.type = FF_REFRESH_EVENT;
         event.user.data1 = opaque;
-        if (!is->refresh && (!is->paused || is->force_refresh)) {
+        if (!is->refresh) {
             is->refresh = 1;
             SDL_PushEvent(&event);
         }
@@ -1170,9 +1156,6 @@ retry:
                 goto retry;
             }
 
-            if (is->paused)
-                goto display;
-
             /* compute nominal last_duration */
             last_duration = vp->pts - is->frame_last_pts;
             if (last_duration > 0 && last_duration < 10.0) {
@@ -1251,13 +1234,11 @@ retry:
                 }
             }
 
-display:
             /* display picture */
             if (!display_disable)
                 video_display(is);
 
-            if (!is->paused)
-                pictq_next_picture(is);
+            pictq_next_picture(is);
         }
     } else if (is->audio_st) {
         /* draw the next audio frame */
@@ -1269,7 +1250,6 @@ display:
         if (!display_disable)
             video_display(is);
     }
-    is->force_refresh = 0;
     if (show_status) {
         static int64_t last_time;
         int64_t cur_time;
@@ -1307,10 +1287,9 @@ display:
 
 /* allocate a picture (needs to do that in main thread to avoid
    potential locking problems */
-static void alloc_picture(AllocEventProps *event_props)
+static void alloc_picture(void *opaque)
 {
-    VideoState *is = event_props->is;
-    AVFrame *frame = event_props->frame;
+    VideoState *is = opaque;
     VideoPicture *vp;
 
     vp = &is->pictq[is->pictq_windex];
@@ -1322,14 +1301,16 @@ static void alloc_picture(AllocEventProps *event_props)
     if (vp->picref)
         avfilter_unref_buffer(vp->picref);
     vp->picref = NULL;
+
+    vp->width   = is->out_video_filter->inputs[0]->w;
+    vp->height  = is->out_video_filter->inputs[0]->h;
+    vp->pix_fmt = is->out_video_filter->inputs[0]->format;
+#else
+    vp->width   = is->video_st->codec->width;
+    vp->height  = is->video_st->codec->height;
+    vp->pix_fmt = is->video_st->codec->pix_fmt;
 #endif
 
-    vp->width   = frame->width;
-    vp->height  = frame->height;
-    vp->pix_fmt = frame->format;
-
-    video_open(event_props->is, 0);
-
     vp->bmp = SDL_CreateYUVOverlay(vp->width, vp->height,
                                    SDL_YV12_OVERLAY,
                                    screen);
@@ -1391,22 +1372,22 @@ static int queue_picture(VideoState *is, AVFrame *src_frame, double pts1, int64_
 
     /* alloc or resize hardware picture buffer */
     if (!vp->bmp || vp->reallocate ||
-        vp->width  != src_frame->width ||
-        vp->height != src_frame->height) {
+#if CONFIG_AVFILTER
+        vp->width  != is->out_video_filter->inputs[0]->w ||
+        vp->height != is->out_video_filter->inputs[0]->h) {
+#else
+        vp->width != is->video_st->codec->width ||
+        vp->height != is->video_st->codec->height) {
+#endif
         SDL_Event event;
-        AllocEventProps event_props;
-
-        event_props.frame = src_frame;
-        event_props.is = is;
 
         vp->allocated  = 0;
         vp->reallocate = 0;
 
         /* the allocation must be done in the main thread to avoid
-           locking problems. We wait in this block for the event to complete,
-           so we can pass a pointer to event_props to it. */
+           locking problems */
         event.type = FF_ALLOC_EVENT;
-        event.user.data1 = &event_props;
+        event.user.data1 = is;
         SDL_PushEvent(&event);
 
         /* wait until the picture is allocated */
@@ -1428,7 +1409,7 @@ static int queue_picture(VideoState *is, AVFrame *src_frame, double pts1, int64_
 
     /* if the frame is not skipped, then display it */
     if (vp->bmp) {
-        AVPicture pict = { { 0 } };
+        AVPicture pict;
 #if CONFIG_AVFILTER
         if (vp->picref)
             avfilter_unref_buffer(vp->picref);
@@ -1438,6 +1419,7 @@ static int queue_picture(VideoState *is, AVFrame *src_frame, double pts1, int64_
         /* get a pointer on the bitmap */
         SDL_LockYUVOverlay (vp->bmp);
 
+        memset(&pict, 0, sizeof(AVPicture));
         pict.data[0] = vp->bmp->pixels[0];
         pict.data[1] = vp->bmp->pixels[2];
         pict.data[2] = vp->bmp->pixels[1];
@@ -1450,7 +1432,6 @@ static int queue_picture(VideoState *is, AVFrame *src_frame, double pts1, int64_
         // FIXME use direct rendering
         av_picture_copy(&pict, (AVPicture *)src_frame,
                         vp->pix_fmt, vp->width, vp->height);
-        vp->sample_aspect_ratio = vp->picref->video->sample_aspect_ratio;
 #else
         sws_flags = av_get_int(sws_opts, "sws_flags", NULL);
         is->img_convert_ctx = sws_getCachedContext(is->img_convert_ctx,
@@ -1462,7 +1443,6 @@ static int queue_picture(VideoState *is, AVFrame *src_frame, double pts1, int64_
         }
         sws_scale(is->img_convert_ctx, src_frame->data, src_frame->linesize,
                   0, vp->height, pict.data, pict.linesize);
-        vp->sample_aspect_ratio = av_guess_sample_aspect_ratio(is->ic, is->video_st, src_frame);
 #endif
         /* update the bitmap content */
         SDL_UnlockYUVOverlay(vp->bmp);
@@ -1515,7 +1495,7 @@ static int get_video_frame(VideoState *is, AVFrame *frame, int64_t *pts, AVPacke
         int ret = 1;
 
         if (decoder_reorder_pts == -1) {
-            *pts = av_frame_get_best_effort_timestamp(frame);
+            *pts = *(int64_t*)av_opt_ptr(avcodec_get_frame_class(), frame, "best_effort_timestamp");
         } else if (decoder_reorder_pts) {
             *pts = frame->pkt_pts;
         } else {
@@ -1565,7 +1545,7 @@ static int input_get_buffer(AVCodecContext *codec, AVFrame *pic)
     AVFilterContext *ctx = codec->opaque;
     AVFilterBufferRef  *ref;
     int perms = AV_PERM_WRITE;
-    int i, w, h, stride[AV_NUM_DATA_POINTERS];
+    int i, w, h, stride[4];
     unsigned edge;
     int pixel_size;
 
@@ -1584,7 +1564,7 @@ static int input_get_buffer(AVCodecContext *codec, AVFrame *pic)
     w = codec->width;
     h = codec->height;
 
-    if(av_image_check_size(w, h, 0, codec) || codec->pix_fmt<0)
+    if(av_image_check_size(w, h, 0, codec))
         return -1;
 
     avcodec_align_dimensions2(codec, &w, &h, stride);
@@ -1605,7 +1585,6 @@ static int input_get_buffer(AVCodecContext *codec, AVFrame *pic)
         unsigned hshift = (i == 1 || i == 2) ? av_pix_fmt_descriptors[ref->format].log2_chroma_w : 0;
         unsigned vshift = (i == 1 || i == 2) ? av_pix_fmt_descriptors[ref->format].log2_chroma_h : 0;
 
-        pic->base[i]     = ref->data[i];
         if (ref->data[i]) {
             ref->data[i]    += ((edge * pixel_size) >> hshift) + ((edge * ref->linesize[i]) >> vshift);
         }
@@ -1615,10 +1594,6 @@ static int input_get_buffer(AVCodecContext *codec, AVFrame *pic)
     pic->opaque = ref;
     pic->type   = FF_BUFFER_TYPE_USER;
     pic->reordered_opaque = codec->reordered_opaque;
-    pic->width               = codec->width;
-    pic->height              = codec->height;
-    pic->format              = codec->pix_fmt;
-    pic->sample_aspect_ratio = codec->sample_aspect_ratio;
     if (codec->pkt) pic->pkt_pts = codec->pkt->pts;
     else            pic->pkt_pts = AV_NOPTS_VALUE;
     return 0;
@@ -1696,19 +1671,18 @@ static int input_request_frame(AVFilterLink *link)
     if (priv->use_dr1 && priv->frame->opaque) {
         picref = avfilter_ref_buffer(priv->frame->opaque, ~0);
     } else {
-        picref = avfilter_get_video_buffer(link, AV_PERM_WRITE, priv->frame->width, priv->frame->height);
+        picref = avfilter_get_video_buffer(link, AV_PERM_WRITE, link->w, link->h);
         av_image_copy(picref->data, picref->linesize,
-                      (const uint8_t **)(void **)priv->frame->data, priv->frame->linesize,
-                      picref->format, priv->frame->width, priv->frame->height);
+                      priv->frame->data, priv->frame->linesize,
+                      picref->format, link->w, link->h);
     }
     av_free_packet(&pkt);
 
     avfilter_copy_frame_props(picref, priv->frame);
-    picref->video->sample_aspect_ratio = av_guess_sample_aspect_ratio(priv->is->ic, priv->is->video_st, priv->frame);
     picref->pts = pts;
 
     avfilter_start_frame(link, picref);
-    avfilter_draw_slice(link, 0, picref->video->h, 1);
+    avfilter_draw_slice(link, 0, link->h, 1);
     avfilter_end_frame(link);
 
     return 0;
@@ -1760,11 +1734,11 @@ static AVFilter input_filter =
 
 static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const char *vfilters)
 {
-    static const enum PixelFormat pix_fmts[] = { PIX_FMT_YUV420P, PIX_FMT_NONE };
     char sws_flags_str[128];
     int ret;
+    enum PixelFormat pix_fmts[] = { PIX_FMT_YUV420P, PIX_FMT_NONE };
     AVBufferSinkParams *buffersink_params = av_buffersink_params_alloc();
-    AVFilterContext *filt_src = NULL, *filt_out = NULL, *filt_format;;
+    AVFilterContext *filt_src = NULL, *filt_out = NULL;
     snprintf(sws_flags_str, sizeof(sws_flags_str), "flags=%d", sws_flags);
     graph->scale_sws_opts = av_strdup(sws_flags_str);
 
@@ -1773,27 +1747,17 @@ static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const c
         return ret;
 
 #if FF_API_OLD_VSINK_API
-    ret = avfilter_graph_create_filter(&filt_out,
-                                       avfilter_get_by_name("buffersink"),
-                                       "out", NULL, pix_fmts, graph);
+    ret = avfilter_graph_create_filter(&filt_out, avfilter_get_by_name("buffersink"), "out",
+                                       NULL, pix_fmts, graph);
 #else
     buffersink_params->pixel_fmts = pix_fmts;
-    ret = avfilter_graph_create_filter(&filt_out,
-                                       avfilter_get_by_name("buffersink"),
-                                       "out", NULL, buffersink_params, graph);
+    ret = avfilter_graph_create_filter(&filt_out, avfilter_get_by_name("buffersink"), "out",
+                                       NULL, buffersink_params, graph);
 #endif
     av_freep(&buffersink_params);
     if (ret < 0)
         return ret;
 
-    if ((ret = avfilter_graph_create_filter(&filt_format,
-                                            avfilter_get_by_name("format"),
-                                            "format", "yuv420p", NULL, graph)) < 0)
-        return ret;
-    if ((ret = avfilter_link(filt_format, 0, filt_out, 0)) < 0)
-        return ret;
-
-
     if (vfilters) {
         AVFilterInOut *outputs = avfilter_inout_alloc();
         AVFilterInOut *inputs  = avfilter_inout_alloc();
@@ -1804,14 +1768,14 @@ static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const c
         outputs->next    = NULL;
 
         inputs->name    = av_strdup("out");
-        inputs->filter_ctx = filt_format;
+        inputs->filter_ctx = filt_out;
         inputs->pad_idx = 0;
         inputs->next    = NULL;
 
         if ((ret = avfilter_graph_parse(graph, vfilters, &inputs, &outputs, NULL)) < 0)
             return ret;
     } else {
-        if ((ret = avfilter_link(filt_src, 0, filt_format, 0)) < 0)
+        if ((ret = avfilter_link(filt_src, 0, filt_out, 0)) < 0)
             return ret;
     }
 
@@ -1839,13 +1803,8 @@ static int video_thread(void *arg)
     int last_w = is->video_st->codec->width;
     int last_h = is->video_st->codec->height;
 
-    if ((ret = configure_video_filters(graph, is, vfilters)) < 0) {
-        SDL_Event event;
-        event.type = FF_QUIT_EVENT;
-        event.user.data1 = is;
-        SDL_PushEvent(&event);
+    if ((ret = configure_video_filters(graph, is, vfilters)) < 0)
         goto the_end;
-    }
     filt_out = is->out_video_filter;
 #endif
 
@@ -1875,14 +1834,11 @@ static int video_thread(void *arg)
         if (picref) {
             avfilter_fill_frame_from_video_buffer_ref(frame, picref);
             pts_int = picref->pts;
-            tb      = filt_out->inputs[0]->time_base;
             pos     = picref->pos;
             frame->opaque = picref;
-
-            ret = 1;
         }
 
-        if (ret >= 0 && av_cmp_q(tb, is->video_st->time_base)) {
+        if (av_cmp_q(tb, is->video_st->time_base)) {
             av_unused int64_t pts1 = pts_int;
             pts_int = av_rescale_q(pts_int, tb, is->video_st->time_base);
             av_dlog(NULL, "video_thread(): "
@@ -1894,8 +1850,6 @@ static int video_thread(void *arg)
         ret = get_video_frame(is, frame, &pts_int, &pkt);
         pos = pkt.pos;
         av_free_packet(&pkt);
-        if (ret == 0)
-            continue;
 #endif
 
         if (ret < 0)
@@ -1921,9 +1875,7 @@ static int video_thread(void *arg)
             stream_toggle_pause(is);
     }
  the_end:
-    avcodec_flush_buffers(is->video_st->codec);
 #if CONFIG_AVFILTER
-    av_freep(&vfilters);
     avfilter_graph_free(&graph);
 #endif
     av_free(frame);
@@ -2167,8 +2119,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
             /* if no pts, then compute it */
             pts = is->audio_clock;
             *pts_ptr = pts;
-            is->audio_clock += (double)data_size /
-                (dec->channels * dec->sample_rate * av_get_bytes_per_sample(dec->sample_fmt));
+            is->audio_clock += (double)data_size / (dec->channels * dec->sample_rate * av_get_bytes_per_sample(dec->sample_fmt));
 #ifdef DEBUG
             {
                 static double last_clock;
@@ -2194,10 +2145,8 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
         if ((new_packet = packet_queue_get(&is->audioq, pkt, 1)) < 0)
             return -1;
 
-        if (pkt->data == flush_pkt.data) {
+        if (pkt->data == flush_pkt.data)
             avcodec_flush_buffers(dec);
-            flush_complete = 0;
-        }
 
         *pkt_temp = *pkt;
 
@@ -2269,9 +2218,9 @@ static int stream_component_open(VideoState *is, int stream_index)
     opts = filter_codec_opts(codec_opts, codec, ic, ic->streams[stream_index]);
 
     switch(avctx->codec_type){
-        case AVMEDIA_TYPE_AUDIO   : is->last_audio_stream    = stream_index; if(audio_codec_name   ) codec= avcodec_find_decoder_by_name(   audio_codec_name); break;
-        case AVMEDIA_TYPE_SUBTITLE: is->last_subtitle_stream = stream_index; if(subtitle_codec_name) codec= avcodec_find_decoder_by_name(subtitle_codec_name); break;
-        case AVMEDIA_TYPE_VIDEO   : is->last_video_stream    = stream_index; if(video_codec_name   ) codec= avcodec_find_decoder_by_name(   video_codec_name); break;
+        case AVMEDIA_TYPE_AUDIO   : if(audio_codec_name   ) codec= avcodec_find_decoder_by_name(   audio_codec_name); break;
+        case AVMEDIA_TYPE_SUBTITLE: if(subtitle_codec_name) codec= avcodec_find_decoder_by_name(subtitle_codec_name); break;
+        case AVMEDIA_TYPE_VIDEO   : if(video_codec_name   ) codec= avcodec_find_decoder_by_name(   video_codec_name); break;
     }
     if (!codec)
         return -1;
@@ -2295,7 +2244,6 @@ static int stream_component_open(VideoState *is, int stream_index)
         avctx->flags |= CODEC_FLAG_EMU_EDGE;
 
     if (avctx->codec_type == AVMEDIA_TYPE_AUDIO) {
-        memset(&is->audio_pkt_temp, 0, sizeof(is->audio_pkt_temp));
         env = SDL_getenv("SDL_AUDIO_CHANNELS");
         if (env)
             wanted_channel_layout = av_get_default_channel_layout(SDL_atoi(env));
@@ -2372,20 +2320,20 @@ static int stream_component_open(VideoState *is, int stream_index)
         is->audio_diff_threshold = 2.0 * SDL_AUDIO_BUFFER_SIZE / wanted_spec.freq;
 
         memset(&is->audio_pkt, 0, sizeof(is->audio_pkt));
-        packet_queue_start(&is->audioq);
+        packet_queue_init(&is->audioq);
         SDL_PauseAudio(0);
         break;
     case AVMEDIA_TYPE_VIDEO:
         is->video_stream = stream_index;
         is->video_st = ic->streams[stream_index];
 
-        packet_queue_start(&is->videoq);
+        packet_queue_init(&is->videoq);
         is->video_tid = SDL_CreateThread(video_thread, is);
         break;
     case AVMEDIA_TYPE_SUBTITLE:
         is->subtitle_stream = stream_index;
         is->subtitle_st = ic->streams[stream_index];
-        packet_queue_start(&is->subtitleq);
+        packet_queue_init(&is->subtitleq);
 
         is->subtitle_tid = SDL_CreateThread(subtitle_thread, is);
         break;
@@ -2410,10 +2358,10 @@ static void stream_component_close(VideoState *is, int stream_index)
 
         SDL_CloseAudio();
 
-        packet_queue_flush(&is->audioq);
-        av_free_packet(&is->audio_pkt);
+        packet_queue_end(&is->audioq);
         if (is->swr_ctx)
             swr_free(&is->swr_ctx);
+        av_free_packet(&is->audio_pkt);
         av_freep(&is->audio_buf1);
         is->audio_buf = NULL;
         av_freep(&is->frame);
@@ -2436,7 +2384,7 @@ static void stream_component_close(VideoState *is, int stream_index)
 
         SDL_WaitThread(is->video_tid, NULL);
 
-        packet_queue_flush(&is->videoq);
+        packet_queue_end(&is->videoq);
         break;
     case AVMEDIA_TYPE_SUBTITLE:
         packet_queue_abort(&is->subtitleq);
@@ -2451,7 +2399,7 @@ static void stream_component_close(VideoState *is, int stream_index)
 
         SDL_WaitThread(is->subtitle_tid, NULL);
 
-        packet_queue_flush(&is->subtitleq);
+        packet_queue_end(&is->subtitleq);
         break;
     default:
         break;
@@ -2498,9 +2446,9 @@ static int read_thread(void *arg)
     int orig_nb_streams;
 
     memset(st_index, -1, sizeof(st_index));
-    is->last_video_stream = is->video_stream = -1;
-    is->last_audio_stream = is->audio_stream = -1;
-    is->last_subtitle_stream = is->subtitle_stream = -1;
+    is->video_stream = -1;
+    is->audio_stream = -1;
+    is->subtitle_stream = -1;
 
     ic = avformat_alloc_context();
     ic->interrupt_callback.callback = decode_interrupt_cb;
@@ -2654,9 +2602,9 @@ static int read_thread(void *arg)
 
         /* if the queue are full, no need to read more */
         if (   is->audioq.size + is->videoq.size + is->subtitleq.size > MAX_QUEUE_SIZE
-            || (   (is->audioq   .nb_packets > MIN_FRAMES || is->audio_stream < 0 || is->audioq.abort_request)
-                && (is->videoq   .nb_packets > MIN_FRAMES || is->video_stream < 0 || is->videoq.abort_request)
-                && (is->subtitleq.nb_packets > MIN_FRAMES || is->subtitle_stream < 0 || is->subtitleq.abort_request))) {
+            || (   (is->audioq   .size  > MIN_AUDIOQ_SIZE || is->audio_stream < 0)
+                && (is->videoq   .nb_packets > MIN_FRAMES || is->video_stream < 0)
+                && (is->subtitleq.nb_packets > MIN_FRAMES || is->subtitle_stream < 0))) {
             /* wait 10 ms */
             SDL_Delay(10);
             continue;
@@ -2761,10 +2709,6 @@ static VideoState *stream_open(const char *filename, AVInputFormat *iformat)
     is->subpq_mutex = SDL_CreateMutex();
     is->subpq_cond  = SDL_CreateCond();
 
-    packet_queue_init(&is->videoq);
-    packet_queue_init(&is->audioq);
-    packet_queue_init(&is->subtitleq);
-
     is->av_sync_type = av_sync_type;
     is->read_tid     = SDL_CreateThread(read_thread, is);
     if (!is->read_tid) {
@@ -2778,19 +2722,16 @@ static void stream_cycle_channel(VideoState *is, int codec_type)
 {
     AVFormatContext *ic = is->ic;
     int start_index, stream_index;
-    int old_index;
     AVStream *st;
 
-    if (codec_type == AVMEDIA_TYPE_VIDEO) {
-        start_index = is->last_video_stream;
-        old_index = is->video_stream;
-    } else if (codec_type == AVMEDIA_TYPE_AUDIO) {
-        start_index = is->last_audio_stream;
-        old_index = is->audio_stream;
-    } else {
-        start_index = is->last_subtitle_stream;
-        old_index = is->subtitle_stream;
-    }
+    if (codec_type == AVMEDIA_TYPE_VIDEO)
+        start_index = is->video_stream;
+    else if (codec_type == AVMEDIA_TYPE_AUDIO)
+        start_index = is->audio_stream;
+    else
+        start_index = is->subtitle_stream;
+    if (start_index < (codec_type == AVMEDIA_TYPE_SUBTITLE ? -1 : 0))
+        return;
     stream_index = start_index;
     for (;;) {
         if (++stream_index >= is->ic->nb_streams)
@@ -2798,12 +2739,9 @@ static void stream_cycle_channel(VideoState *is, int codec_type)
             if (codec_type == AVMEDIA_TYPE_SUBTITLE)
             {
                 stream_index = -1;
-                is->last_subtitle_stream = -1;
                 goto the_end;
-            }
-            if (start_index == -1)
-                return;
-            stream_index = 0;
+            } else
+                stream_index = 0;
         }
         if (stream_index == start_index)
             return;
@@ -2825,7 +2763,7 @@ static void stream_cycle_channel(VideoState *is, int codec_type)
         }
     }
  the_end:
-    stream_component_close(is, old_index);
+    stream_component_close(is, start_index);
     stream_component_open(is, stream_index);
 }
 
@@ -2889,7 +2827,6 @@ static void event_loop(VideoState *cur_stream)
                 break;
             case SDLK_f:
                 toggle_full_screen(cur_stream);
-                cur_stream->force_refresh = 1;
                 break;
             case SDLK_p:
             case SDLK_SPACE:
@@ -2909,7 +2846,6 @@ static void event_loop(VideoState *cur_stream)
                 break;
             case SDLK_w:
                 toggle_audio_display(cur_stream);
-                cur_stream->force_refresh = 1;
                 break;
             case SDLK_PAGEUP:
                 incr = 600.0;
@@ -2952,9 +2888,6 @@ static void event_loop(VideoState *cur_stream)
                 break;
             }
             break;
-        case SDL_VIDEOEXPOSE:
-            cur_stream->force_refresh = 1;
-            break;
         case SDL_MOUSEBUTTONDOWN:
             if (exit_on_mousedown) {
                 do_exit(cur_stream);
@@ -2997,13 +2930,13 @@ static void event_loop(VideoState *cur_stream)
                                           SDL_HWSURFACE|SDL_RESIZABLE|SDL_ASYNCBLIT|SDL_HWACCEL);
                 screen_width  = cur_stream->width  = event.resize.w;
                 screen_height = cur_stream->height = event.resize.h;
-                cur_stream->force_refresh = 1;
             break;
         case SDL_QUIT:
         case FF_QUIT_EVENT:
             do_exit(cur_stream);
             break;
         case FF_ALLOC_EVENT:
+            video_open(event.user.data1, 0);
             alloc_picture(event.user.data1);
             break;
         case FF_REFRESH_EVENT:
@@ -3282,7 +3215,7 @@ int main(int argc, char **argv)
     }
 
     av_init_packet(&flush_pkt);
-    flush_pkt.data = (char *)(intptr_t)"FLUSH";
+    flush_pkt.data = "FLUSH";
 
     is = stream_open(input_filename, file_iformat);
     if (!is) {

ffpresets/libvpx-1080p.ffpreset

@@ -0,0 +1,17 @@
+vcodec=libvpx
+g=120
+rc_lookahead=16
+quality=good
+speed=0
+profile=1
+qmax=51
+qmin=11
+slices=4
+vb=2M
+
+#ignored unless using -pass 2
+maxrate=24M
+minrate=100k
+arnr_max_frames=7
+arnr_strength=5
+arnr_type=3

ffpresets/libvpx-1080p50_60.ffpreset

@@ -0,0 +1,17 @@
+vcodec=libvpx
+g=120
+rc_lookahead=25
+quality=good
+speed=0
+profile=1
+qmax=51
+qmin=11
+slices=4
+vb=2M
+
+#ignored unless using -pass 2
+maxrate=24M
+minrate=100k
+arnr_max_frames=7
+arnr_strength=5
+arnr_type=3

ffpresets/libvpx-360p.ffpreset

@@ -0,0 +1,16 @@
+vcodec=libvpx
+g=120
+rc_lookahead=16
+quality=good
+speed=0
+profile=0
+qmax=63
+qmin=0
+vb=768k
+
+#ignored unless using -pass 2
+maxrate=1.5M
+minrate=40k
+arnr_max_frames=7
+arnr_strength=5
+arnr_type=3

ffpresets/libvpx-720p.ffpreset

@@ -0,0 +1,17 @@
+vcodec=libvpx
+g=120
+rc_lookahead=16
+quality=good
+speed=0
+profile=0
+qmax=51
+qmin=11
+slices=4
+vb=2M
+
+#ignored unless using -pass 2
+maxrate=24M
+minrate=100k
+arnr_max_frames=7
+arnr_strength=5
+arnr_type=3

ffpresets/libvpx-720p50_60.ffpreset

@@ -0,0 +1,17 @@
+vcodec=libvpx
+g=120
+rc_lookahead=25
+quality=good
+speed=0
+profile=0
+qmax=51
+qmin=11
+slices=4
+vb=2M
+
+#ignored unless using -pass 2
+maxrate=24M
+minrate=100k
+arnr_max_frames=7
+arnr_strength=5
+arnr_type=3

ffserver.c

@@ -30,7 +30,7 @@
 #include <string.h>
 #include <stdlib.h>
 #include "libavformat/avformat.h"
-// FIXME those are internal headers, ffserver _really_ shouldn't use them
+// FIXME those are internal headers, avserver _really_ shouldn't use them
 #include "libavformat/ffm.h"
 #include "libavformat/network.h"
 #include "libavformat/os_support.h"
@@ -339,7 +339,8 @@ static int resolve_host(struct in_addr *sin_addr, const char *hostname)
     if (!ff_inet_aton(hostname, sin_addr)) {
 #if HAVE_GETADDRINFO
         struct addrinfo *ai, *cur;
-        struct addrinfo hints = { 0 };
+        struct addrinfo hints;
+        memset(&hints, 0, sizeof(hints));
         hints.ai_family = AF_INET;
         if (getaddrinfo(hostname, NULL, &hints, &ai))
             return -1;
@@ -561,9 +562,11 @@ static void start_multicast(void)
     default_port = 6000;
     for(stream = first_stream; stream != NULL; stream = stream->next) {
         if (stream->is_multicast) {
+            unsigned random0 = av_lfg_get(&random_state);
+            unsigned random1 = av_lfg_get(&random_state);
             /* open the RTP connection */
             snprintf(session_id, sizeof(session_id), "%08x%08x",
-                     av_lfg_get(&random_state), av_lfg_get(&random_state));
+                     random0, random1);
 
             /* choose a port if none given */
             if (stream->multicast_port == 0) {
@@ -1871,7 +1874,7 @@ static int http_parse_request(HTTPContext *c)
 
 static void fmt_bytecount(AVIOContext *pb, int64_t count)
 {
-    static const char suffix[] = " kMGTP";
+    static const char *suffix = " kMGTP";
     const char *s;
 
     for (s = suffix; count >= 100000 && s[1]; count /= 1000, s++);
@@ -2126,13 +2129,12 @@ static int open_input_stream(HTTPContext *c, const char *info)
     char buf[128];
     char input_filename[1024];
     AVFormatContext *s = NULL;
-    int buf_size, i, ret;
+    int i, ret;
     int64_t stream_pos;
 
     /* find file name */
     if (c->stream->feed) {
         strcpy(input_filename, c->stream->feed->feed_filename);
-        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)
@@ -2144,7 +2146,6 @@ static int open_input_stream(HTTPContext *c, const char *info)
             stream_pos = av_gettime() - c->stream->prebuffer * (int64_t)1000;
     } else {
         strcpy(input_filename, c->stream->feed_filename);
-        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)
@@ -2160,10 +2161,6 @@ static int open_input_stream(HTTPContext *c, const char *info)
         http_log("could not open %s: %d\n", input_filename, ret);
         return -1;
     }
-
-    /* set buffer size */
-    if (buf_size > 0) ffio_set_buf_size(s->pb, buf_size);
-
     s->flags |= AVFMT_FLAG_GENPTS;
     c->fmt_in = s;
     if (strcmp(s->iformat->name, "ffm") && avformat_find_stream_info(c->fmt_in, NULL) < 0) {
@@ -2585,11 +2582,8 @@ static int http_start_receive_data(HTTPContext *c)
     if (c->stream->truncate) {
         /* truncate feed file */
         ffm_write_write_index(c->feed_fd, FFM_PACKET_SIZE);
+        ftruncate(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;
-        }
     } 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));
@@ -2837,7 +2831,7 @@ static int rtsp_parse_request(HTTPContext *c)
     char protocol[32];
     char line[1024];
     int len;
-    RTSPMessageHeader header1 = { 0 }, *header = &header1;
+    RTSPMessageHeader header1, *header = &header1;
 
     c->buffer_ptr[0] = '\0';
     p = c->buffer;
@@ -2863,6 +2857,7 @@ static int rtsp_parse_request(HTTPContext *c)
     }
 
     /* parse each header line */
+    memset(header, 0, sizeof(*header));
     /* skip to next line */
     while (*p != '\n' && *p != '\0')
         p++;
@@ -3093,9 +3088,12 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url,
  found:
 
     /* generate session id if needed */
-    if (h->session_id[0] == '\0')
+    if (h->session_id[0] == '\0') {
+        unsigned random0 = av_lfg_get(&random_state);
+        unsigned random1 = av_lfg_get(&random_state);
         snprintf(h->session_id, sizeof(h->session_id), "%08x%08x",
-                 av_lfg_get(&random_state), av_lfg_get(&random_state));
+                 random0, random1);
+    }
 
     /* find rtp session, and create it if none found */
     rtp_c = find_rtp_session(h->session_id);
@@ -3464,6 +3462,9 @@ static AVStream *add_av_stream1(FFStream *stream, AVCodecContext *codec, int cop
 {
     AVStream *fst;
 
+    if(stream->nb_streams >= FF_ARRAY_ELEMS(stream->streams))
+        return NULL;
+
     fst = av_mallocz(sizeof(AVStream));
     if (!fst)
         return NULL;
@@ -3669,8 +3670,6 @@ static void build_feed_streams(void)
             int matches = 0;
 
             if (avformat_open_input(&s, feed->feed_filename, NULL, NULL) >= 0) {
-                /* set buffer size */
-                ffio_set_buf_size(s->pb, FFM_PACKET_SIZE);
                 /* Now see if it matches */
                 if (s->nb_streams == feed->nb_streams) {
                     matches = 1;
@@ -3811,6 +3810,9 @@ static void add_codec(FFStream *stream, AVCodecContext *av)
 {
     AVStream *st;
 
+    if(stream->nb_streams >= FF_ARRAY_ELEMS(stream->streams))
+        return;
+
     /* compute default parameters */
     switch(av->codec_type) {
     case AVMEDIA_TYPE_AUDIO:
@@ -4230,8 +4232,8 @@ static int parse_ffconfig(const char *filename)
                 }
 
                 stream->fmt = ffserver_guess_format(NULL, stream->filename, NULL);
-                avcodec_get_context_defaults3(&video_enc, NULL);
-                avcodec_get_context_defaults3(&audio_enc, NULL);
+                avcodec_get_context_defaults2(&video_enc, AVMEDIA_TYPE_VIDEO);
+                avcodec_get_context_defaults2(&audio_enc, AVMEDIA_TYPE_AUDIO);
 
                 audio_id = CODEC_ID_NONE;
                 video_id = CODEC_ID_NONE;
@@ -4665,7 +4667,7 @@ static const OptionDef options[] = {
 
 int main(int argc, char **argv)
 {
-    struct sigaction sigact = { { 0 } };
+    struct sigaction sigact;
 
     parse_loglevel(argc, argv, options);
     av_register_all();
@@ -4683,6 +4685,7 @@ int main(int argc, char **argv)
 
     av_lfg_init(&random_state, av_get_random_seed());
 
+    memset(&sigact, 0, sizeof(sigact));
     sigact.sa_handler = handle_child_exit;
     sigact.sa_flags = SA_NOCLDSTOP | SA_RESTART;
     sigaction(SIGCHLD, &sigact, 0);

libavcodec/8bps.c

@@ -38,20 +38,20 @@
 #include "avcodec.h"
 
 
-static const enum PixelFormat pixfmt_rgb24[] = {
-    PIX_FMT_BGR24, PIX_FMT_RGB32, PIX_FMT_NONE };
+static const enum PixelFormat pixfmt_rgb24[] = {PIX_FMT_BGR24, PIX_FMT_RGB32, PIX_FMT_NONE};
 
 /*
  * Decoder context
  */
 typedef struct EightBpsContext {
-    AVCodecContext *avctx;
-    AVFrame pic;
 
-    unsigned char planes;
-    unsigned char planemap[4];
+        AVCodecContext *avctx;
+        AVFrame pic;
 
-    uint32_t pal[256];
+        unsigned char planes;
+        unsigned char planemap[4];
+
+        uint32_t pal[256];
 } EightBpsContext;
 
 
@@ -60,90 +60,87 @@ typedef struct EightBpsContext {
  * Decode a frame
  *
  */
-static int decode_frame(AVCodecContext *avctx, void *data,
-                        int *data_size, AVPacket *avpkt)
+static int decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPacket *avpkt)
 {
-    const uint8_t *buf = avpkt->data;
-    int buf_size       = avpkt->size;
-    EightBpsContext * const c = avctx->priv_data;
-    const unsigned char *encoded = buf;
-    unsigned char *pixptr, *pixptr_end;
-    unsigned int height = avctx->height; // Real image height
-    unsigned int dlen, p, row;
-    const unsigned char *lp, *dp;
-    unsigned char count;
-    unsigned int planes     = c->planes;
-    unsigned char *planemap = c->planemap;
+        const uint8_t *buf = avpkt->data;
+        int buf_size = avpkt->size;
+        EightBpsContext * const c = avctx->priv_data;
+        const unsigned char *encoded = buf;
+        unsigned char *pixptr, *pixptr_end;
+        unsigned int height = avctx->height; // Real image height
+        unsigned int dlen, p, row;
+        const unsigned char *lp, *dp;
+        unsigned char count;
+        unsigned int planes = c->planes;
+        unsigned char *planemap = c->planemap;
 
-    if (c->pic.data[0])
-        avctx->release_buffer(avctx, &c->pic);
+        if(c->pic.data[0])
+                avctx->release_buffer(avctx, &c->pic);
 
-    c->pic.reference    = 0;
-    c->pic.buffer_hints = FF_BUFFER_HINTS_VALID;
-    if (avctx->get_buffer(avctx, &c->pic) < 0){
-        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
-        return -1;
-    }
+        c->pic.reference = 0;
+        c->pic.buffer_hints = FF_BUFFER_HINTS_VALID;
+        if(avctx->get_buffer(avctx, &c->pic) < 0){
+                av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+                return -1;
+        }
 
-    /* Set data pointer after line lengths */
-    dp = encoded + planes * (height << 1);
+        /* Set data pointer after line lengths */
+        dp = encoded + planes * (height << 1);
 
-    for (p = 0; p < planes; p++) {
-        /* Lines length pointer for this plane */
-        lp = encoded + p * (height << 1);
+        for (p = 0; p < planes; p++) {
+                /* Lines length pointer for this plane */
+                lp = encoded + p * (height << 1);
 
-        /* Decode a plane */
-        for (row = 0; row < height; row++) {
-            pixptr = c->pic.data[0] + row * c->pic.linesize[0] + planemap[p];
-            pixptr_end = pixptr + c->pic.linesize[0];
-            dlen = av_be2ne16(*(const unsigned short *)(lp + row * 2));
-            /* Decode a row of this plane */
-            while (dlen > 0) {
-                if (dp + 1 >= buf + buf_size)
-                    return -1;
-                if ((count = *dp++) <= 127) {
-                    count++;
-                    dlen -= count + 1;
-                    if (pixptr + count * planes > pixptr_end)
-                        break;
-                    if (dp + count > buf + buf_size)
-                        return -1;
-                    while (count--) {
-                        *pixptr = *dp++;
-                        pixptr += planes;
-                    }
-                } else {
-                    count = 257 - count;
-                    if (pixptr + count * planes > pixptr_end)
-                        break;
-                    while (count--) {
-                        *pixptr = *dp;
-                        pixptr += planes;
-                    }
-                    dp++;
-                    dlen -= 2;
+                /* Decode a plane */
+                for(row = 0; row < height; row++) {
+                        pixptr = c->pic.data[0] + row * c->pic.linesize[0] + planemap[p];
+                        pixptr_end = pixptr + c->pic.linesize[0];
+                        dlen = av_be2ne16(*(const unsigned short *)(lp+row*2));
+                        /* Decode a row of this plane */
+                        while(dlen > 0) {
+                                if(dp + 1 >= buf+buf_size) return -1;
+                                if ((count = *dp++) <= 127) {
+                                        count++;
+                                        dlen -= count + 1;
+                                        if (pixptr + count * planes > pixptr_end)
+                                            break;
+                                        if(dp + count > buf+buf_size) return -1;
+                                        while(count--) {
+                                                *pixptr = *dp++;
+                                                pixptr += planes;
+                                        }
+                                } else {
+                                        count = 257 - count;
+                                        if (pixptr + count * planes > pixptr_end)
+                                            break;
+                                        while(count--) {
+                                                *pixptr = *dp;
+                                                pixptr += planes;
+                                        }
+                                        dp++;
+                                        dlen -= 2;
+                                }
+                        }
                 }
-            }
-        }
-    }
-
-    if (avctx->bits_per_coded_sample <= 8) {
-        const uint8_t *pal = av_packet_get_side_data(avpkt,
-                                                     AV_PKT_DATA_PALETTE,
-                                                     NULL);
-        if (pal) {
-            c->pic.palette_has_changed = 1;
-            memcpy(c->pal, pal, AVPALETTE_SIZE);
         }
 
-        memcpy (c->pic.data[1], c->pal, AVPALETTE_SIZE);
-    }
+        if (avctx->bits_per_coded_sample <= 8) {
+                const uint8_t *pal = av_packet_get_side_data(avpkt,
+                                                             AV_PKT_DATA_PALETTE,
+                                                             NULL);
+                if (pal) {
+                        c->pic.palette_has_changed = 1;
+                        memcpy(c->pal, pal, AVPALETTE_SIZE);
+                }
 
-    *data_size = sizeof(AVFrame);
-    *(AVFrame*)data = c->pic;
+                memcpy (c->pic.data[1], c->pal, AVPALETTE_SIZE);
+        }
 
-    /* always report that the buffer was completely consumed */
-    return buf_size;
+        *data_size = sizeof(AVFrame);
+        *(AVFrame*)data = c->pic;
+
+        /* always report that the buffer was completely consumed */
+        return buf_size;
 }
 
 
@@ -154,47 +151,47 @@ static int decode_frame(AVCodecContext *avctx, void *data,
  */
 static av_cold int decode_init(AVCodecContext *avctx)
 {
-    EightBpsContext * const c = avctx->priv_data;
+        EightBpsContext * const c = avctx->priv_data;
 
-    c->avctx       = avctx;
-    c->pic.data[0] = NULL;
+        c->avctx = avctx;
 
-    avcodec_get_frame_defaults(&c->pic);
-    switch (avctx->bits_per_coded_sample) {
-    case 8:
-        avctx->pix_fmt = PIX_FMT_PAL8;
-        c->planes      = 1;
-        c->planemap[0] = 0; // 1st plane is palette indexes
-        break;
-    case 24:
-        avctx->pix_fmt = avctx->get_format(avctx, pixfmt_rgb24);
-        c->planes      = 3;
-        c->planemap[0] = 2; // 1st plane is red
-        c->planemap[1] = 1; // 2nd plane is green
-        c->planemap[2] = 0; // 3rd plane is blue
-        break;
-    case 32:
-        avctx->pix_fmt = PIX_FMT_RGB32;
-        c->planes      = 4;
+        avcodec_get_frame_defaults(&c->pic);
+        c->pic.data[0] = NULL;
+
+        switch (avctx->bits_per_coded_sample) {
+                case 8:
+                        avctx->pix_fmt = PIX_FMT_PAL8;
+                        c->planes = 1;
+                        c->planemap[0] = 0; // 1st plane is palette indexes
+                        break;
+                case 24:
+                        avctx->pix_fmt = avctx->get_format(avctx, pixfmt_rgb24);
+                        c->planes = 3;
+                        c->planemap[0] = 2; // 1st plane is red
+                        c->planemap[1] = 1; // 2nd plane is green
+                        c->planemap[2] = 0; // 3rd plane is blue
+                        break;
+                case 32:
+                        avctx->pix_fmt = PIX_FMT_RGB32;
+                        c->planes = 4;
 #if HAVE_BIGENDIAN
-        c->planemap[0] = 1; // 1st plane is red
-        c->planemap[1] = 2; // 2nd plane is green
-        c->planemap[2] = 3; // 3rd plane is blue
-        c->planemap[3] = 0; // 4th plane is alpha
+                        c->planemap[0] = 1; // 1st plane is red
+                        c->planemap[1] = 2; // 2nd plane is green
+                        c->planemap[2] = 3; // 3rd plane is blue
+                        c->planemap[3] = 0; // 4th plane is alpha
 #else
-        c->planemap[0] = 2; // 1st plane is red
-        c->planemap[1] = 1; // 2nd plane is green
-        c->planemap[2] = 0; // 3rd plane is blue
-        c->planemap[3] = 3; // 4th plane is alpha
+                        c->planemap[0] = 2; // 1st plane is red
+                        c->planemap[1] = 1; // 2nd plane is green
+                        c->planemap[2] = 0; // 3rd plane is blue
+                        c->planemap[3] = 3; // 4th plane is alpha
 #endif
-        break;
-    default:
-        av_log(avctx, AV_LOG_ERROR, "Error: Unsupported color depth: %u.\n",
-               avctx->bits_per_coded_sample);
-        return -1;
-    }
+                        break;
+                default:
+                        av_log(avctx, AV_LOG_ERROR, "Error: Unsupported color depth: %u.\n", avctx->bits_per_coded_sample);
+                        return -1;
+        }
 
-    return 0;
+  return 0;
 }
 
 
@@ -207,12 +204,12 @@ static av_cold int decode_init(AVCodecContext *avctx)
  */
 static av_cold int decode_end(AVCodecContext *avctx)
 {
-    EightBpsContext * const c = avctx->priv_data;
+        EightBpsContext * const c = avctx->priv_data;
 
-    if (c->pic.data[0])
-        avctx->release_buffer(avctx, &c->pic);
+        if (c->pic.data[0])
+                avctx->release_buffer(avctx, &c->pic);
 
-    return 0;
+        return 0;
 }
 
 

libavcodec/Makefile

@@ -3,14 +3,7 @@ include $(SUBDIR)../config.mak
 NAME = avcodec
 FFLIBS = avutil
 
-HEADERS = avcodec.h                                                     \
-          avfft.h                                                       \
-          dxva2.h                                                       \
-          vaapi.h                                                       \
-          vda.h                                                         \
-          vdpau.h                                                       \
-          version.h                                                     \
-          xvmc.h                                                        \
+HEADERS = avcodec.h avfft.h dxva2.h opt.h vaapi.h vda.h vdpau.h version.h xvmc.h
 
 OBJS = allcodecs.o                                                      \
        audioconvert.o                                                   \
@@ -46,7 +39,6 @@ OBJS-$(CONFIG_GOLOMB)                  += golomb.o
 OBJS-$(CONFIG_H264DSP)                 += h264dsp.o h264idct.o
 OBJS-$(CONFIG_H264PRED)                += h264pred.o
 OBJS-$(CONFIG_HUFFMAN)                 += huffman.o
-OBJS-$(CONFIG_LIBXVID)                 += libxvid_rc.o
 OBJS-$(CONFIG_LPC)                     += lpc.o
 OBJS-$(CONFIG_LSP)                     += lsp.o
 OBJS-$(CONFIG_MDCT)                    += mdct_fixed.o mdct_float.o
@@ -64,13 +56,11 @@ OBJS-$(CONFIG_VDPAU)                   += vdpau.o
 OBJS-$(CONFIG_A64MULTI_ENCODER)        += a64multienc.o elbg.o
 OBJS-$(CONFIG_A64MULTI5_ENCODER)       += a64multienc.o elbg.o
 OBJS-$(CONFIG_AAC_DECODER)             += aacdec.o aactab.o aacsbr.o aacps.o \
-                                          aacadtsdec.o mpeg4audio.o kbdwin.o \
-                                          sbrdsp.o aacpsdsp.o
+                                          aacadtsdec.o mpeg4audio.o kbdwin.o
 OBJS-$(CONFIG_AAC_ENCODER)             += aacenc.o aaccoder.o    \
                                           aacpsy.o aactab.o      \
                                           psymodel.o iirfilter.o \
-                                          mpeg4audio.o kbdwin.o  \
-                                          audio_frame_queue.o
+                                          mpeg4audio.o kbdwin.o
 OBJS-$(CONFIG_AASC_DECODER)            += aasc.o msrledec.o
 OBJS-$(CONFIG_AC3_DECODER)             += ac3dec.o ac3dec_data.o ac3.o kbdwin.o
 OBJS-$(CONFIG_AC3_ENCODER)             += ac3enc_float.o ac3enc.o ac3tab.o \
@@ -104,15 +94,11 @@ OBJS-$(CONFIG_AURA2_DECODER)           += aura.o
 OBJS-$(CONFIG_AVRP_DECODER)            += r210dec.o
 OBJS-$(CONFIG_AVRP_ENCODER)            += r210enc.o
 OBJS-$(CONFIG_AVS_DECODER)             += avs.o
-OBJS-$(CONFIG_AVUI_DECODER)            += avuidec.o
-OBJS-$(CONFIG_AVUI_ENCODER)            += avuienc.o
-OBJS-$(CONFIG_AYUV_DECODER)            += v408dec.o
-OBJS-$(CONFIG_AYUV_ENCODER)            += v408enc.o
 OBJS-$(CONFIG_BETHSOFTVID_DECODER)     += bethsoftvideo.o
 OBJS-$(CONFIG_BFI_DECODER)             += bfi.o
 OBJS-$(CONFIG_BINK_DECODER)            += bink.o binkdsp.o
-OBJS-$(CONFIG_BINKAUDIO_DCT_DECODER)   += binkaudio.o wma.o wma_common.o
-OBJS-$(CONFIG_BINKAUDIO_RDFT_DECODER)  += binkaudio.o wma.o wma_common.o
+OBJS-$(CONFIG_BINKAUDIO_DCT_DECODER)   += binkaudio.o wma.o
+OBJS-$(CONFIG_BINKAUDIO_RDFT_DECODER)  += binkaudio.o wma.o
 OBJS-$(CONFIG_BINTEXT_DECODER)         += bintext.o cga_data.o
 OBJS-$(CONFIG_BMP_DECODER)             += bmp.o msrledec.o
 OBJS-$(CONFIG_BMP_ENCODER)             += bmpenc.o
@@ -122,15 +108,13 @@ OBJS-$(CONFIG_C93_DECODER)             += c93.o
 OBJS-$(CONFIG_CAVS_DECODER)            += cavs.o cavsdec.o cavsdsp.o \
                                           mpeg12data.o mpegvideo.o
 OBJS-$(CONFIG_CDGRAPHICS_DECODER)      += cdgraphics.o
-OBJS-$(CONFIG_CDXL_DECODER)            += cdxl.o
 OBJS-$(CONFIG_CINEPAK_DECODER)         += cinepak.o
 OBJS-$(CONFIG_CLJR_DECODER)            += cljr.o
 OBJS-$(CONFIG_CLJR_ENCODER)            += cljr.o
 OBJS-$(CONFIG_COOK_DECODER)            += cook.o
 OBJS-$(CONFIG_CSCD_DECODER)            += cscd.o
 OBJS-$(CONFIG_CYUV_DECODER)            += cyuv.o
-OBJS-$(CONFIG_DCA_DECODER)             += dca.o synth_filter.o dcadsp.o \
-                                          dca_parser.o
+OBJS-$(CONFIG_DCA_DECODER)             += dca.o synth_filter.o dcadsp.o
 OBJS-$(CONFIG_DCA_ENCODER)             += dcaenc.o
 OBJS-$(CONFIG_DIRAC_DECODER)           += diracdec.o dirac.o diracdsp.o \
                                           dirac_arith.o mpeg12data.o dwt.o
@@ -148,8 +132,8 @@ OBJS-$(CONFIG_DVBSUB_DECODER)          += dvbsubdec.o
 OBJS-$(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-$(CONFIG_DVVIDEO_DECODER)         += dv.o dvdata.o
+OBJS-$(CONFIG_DVVIDEO_ENCODER)         += dv.o dvdata.o
 OBJS-$(CONFIG_DXA_DECODER)             += dxa.o
 OBJS-$(CONFIG_DXTORY_DECODER)          += dxtory.o
 OBJS-$(CONFIG_EAC3_DECODER)            += eac3dec.o eac3_data.o
@@ -170,13 +154,12 @@ OBJS-$(CONFIG_EIGHTSVX_FIB_DECODER)    += 8svx.o
 OBJS-$(CONFIG_EIGHTSVX_RAW_DECODER)    += 8svx.o
 OBJS-$(CONFIG_ESCAPE124_DECODER)       += escape124.o
 OBJS-$(CONFIG_ESCAPE130_DECODER)       += escape130.o
-OBJS-$(CONFIG_EXR_DECODER)             += exr.o
 OBJS-$(CONFIG_FFV1_DECODER)            += ffv1.o rangecoder.o
 OBJS-$(CONFIG_FFV1_ENCODER)            += ffv1.o rangecoder.o
 OBJS-$(CONFIG_FFVHUFF_DECODER)         += huffyuv.o
 OBJS-$(CONFIG_FFVHUFF_ENCODER)         += huffyuv.o
 OBJS-$(CONFIG_FFWAVESYNTH_DECODER)     += ffwavesynth.o
-OBJS-$(CONFIG_FLAC_DECODER)            += flacdec.o flacdata.o flac.o
+OBJS-$(CONFIG_FLAC_DECODER)            += flacdec.o flacdata.o flac.o vorbis_data.o
 OBJS-$(CONFIG_FLAC_ENCODER)            += flacenc.o flacdata.o flac.o vorbis_data.o
 OBJS-$(CONFIG_FLASHSV_DECODER)         += flashsv.o
 OBJS-$(CONFIG_FLASHSV_ENCODER)         += flashsvenc.o
@@ -188,15 +171,15 @@ OBJS-$(CONFIG_FRAPS_DECODER)           += fraps.o
 OBJS-$(CONFIG_FRWU_DECODER)            += frwu.o
 OBJS-$(CONFIG_G723_1_DECODER)          += g723_1.o acelp_vectors.o \
                                           celp_filters.o celp_math.o
-OBJS-$(CONFIG_G723_1_ENCODER)          += g723_1.o acelp_vectors.o celp_math.o
+OBJS-$(CONFIG_G723_1_ENCODER)          += g723_1.o
 OBJS-$(CONFIG_G729_DECODER)            += g729dec.o lsp.o celp_math.o acelp_filters.o acelp_pitch_delay.o acelp_vectors.o g729postfilter.o
 OBJS-$(CONFIG_GIF_DECODER)             += gifdec.o lzw.o
 OBJS-$(CONFIG_GIF_ENCODER)             += gif.o lzwenc.o
 OBJS-$(CONFIG_GSM_DECODER)             += gsmdec.o gsmdec_data.o msgsmdec.o
 OBJS-$(CONFIG_GSM_MS_DECODER)          += gsmdec.o gsmdec_data.o msgsmdec.o
-OBJS-$(CONFIG_H261_DECODER)            += h261dec.o h261.o h261data.o  \
+OBJS-$(CONFIG_H261_DECODER)            += h261dec.o h261.o \
                                           mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_H261_ENCODER)            += h261enc.o h261.o h261data.o  \
+OBJS-$(CONFIG_H261_ENCODER)            += h261enc.o h261.o             \
                                           mpegvideo_enc.o motion_est.o \
                                           ratecontrol.o mpeg12data.o   \
                                           mpegvideo.o
@@ -231,7 +214,6 @@ OBJS-$(CONFIG_INDEO4_DECODER)          += indeo4.o ivi_common.o ivi_dsp.o
 OBJS-$(CONFIG_INDEO5_DECODER)          += indeo5.o ivi_common.o ivi_dsp.o
 OBJS-$(CONFIG_INTERPLAY_DPCM_DECODER)  += dpcm.o
 OBJS-$(CONFIG_INTERPLAY_VIDEO_DECODER) += interplayvideo.o
-OBJS-$(CONFIG_JACOSUB_DECODER)         += jacosubdec.o ass.o
 OBJS-$(CONFIG_JPEG2000_DECODER)        += j2kdec.o mqcdec.o mqc.o j2k.o j2k_dwt.o
 OBJS-$(CONFIG_JPEG2000_ENCODER)        += j2kenc.o mqcenc.o mqc.o j2k.o j2k_dwt.o
 OBJS-$(CONFIG_JPEGLS_DECODER)          += jpeglsdec.o jpegls.o \
@@ -250,7 +232,6 @@ OBJS-$(CONFIG_MACE3_DECODER)           += mace.o
 OBJS-$(CONFIG_MACE6_DECODER)           += mace.o
 OBJS-$(CONFIG_MDEC_DECODER)            += mdec.o mpeg12.o mpeg12data.o \
                                           mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_MICRODVD_DECODER)        += microdvddec.o ass.o
 OBJS-$(CONFIG_MIMIC_DECODER)           += mimic.o
 OBJS-$(CONFIG_MJPEG_DECODER)           += mjpegdec.o mjpeg.o
 OBJS-$(CONFIG_MJPEG_ENCODER)           += mjpegenc.o mjpeg.o           \
@@ -314,22 +295,19 @@ OBJS-$(CONFIG_MPEG4_VAAPI_HWACCEL)     += vaapi_mpeg4.o
 OBJS-$(CONFIG_MSMPEG4V1_DECODER)       += msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_MSMPEG4V2_DECODER)       += msmpeg4.o msmpeg4data.o h263dec.o \
                                           h263.o ituh263dec.o mpeg4videodec.o
-OBJS-$(CONFIG_MSMPEG4V2_ENCODER)       += msmpeg4.o msmpeg4enc.o msmpeg4data.o \
-                                          h263dec.o h263.o ituh263dec.o        \
-                                          mpeg4videodec.o
+OBJS-$(CONFIG_MSMPEG4V2_ENCODER)       += msmpeg4.o msmpeg4data.o h263dec.o \
+                                          h263.o ituh263dec.o mpeg4videodec.o
 OBJS-$(CONFIG_MSMPEG4V3_DECODER)       += msmpeg4.o msmpeg4data.o h263dec.o \
                                           h263.o ituh263dec.o mpeg4videodec.o
-OBJS-$(CONFIG_MSMPEG4V3_ENCODER)       += msmpeg4.o msmpeg4enc.o msmpeg4data.o \
-                                          h263dec.o h263.o ituh263dec.o        \
-                                          mpeg4videodec.o
+OBJS-$(CONFIG_MSMPEG4V3_ENCODER)       += msmpeg4.o msmpeg4data.o h263dec.o \
+                                          h263.o ituh263dec.o mpeg4videodec.o
 OBJS-$(CONFIG_MSRLE_DECODER)           += msrle.o msrledec.o
 OBJS-$(CONFIG_MSVIDEO1_DECODER)        += msvideo1.o
 OBJS-$(CONFIG_MSVIDEO1_ENCODER)        += msvideo1enc.o elbg.o
 OBJS-$(CONFIG_MSZH_DECODER)            += lcldec.o
 OBJS-$(CONFIG_MXPEG_DECODER)           += mxpegdec.o mjpegdec.o mjpeg.o
 OBJS-$(CONFIG_NELLYMOSER_DECODER)      += nellymoserdec.o nellymoser.o
-OBJS-$(CONFIG_NELLYMOSER_ENCODER)      += nellymoserenc.o nellymoser.o \
-                                          audio_frame_queue.o
+OBJS-$(CONFIG_NELLYMOSER_ENCODER)      += nellymoserenc.o nellymoser.o
 OBJS-$(CONFIG_NUV_DECODER)             += nuv.o rtjpeg.o
 OBJS-$(CONFIG_PAM_DECODER)             += pnmdec.o pnm.o
 OBJS-$(CONFIG_PAM_ENCODER)             += pamenc.o pnm.o
@@ -343,15 +321,13 @@ OBJS-$(CONFIG_PGMYUV_DECODER)          += pnmdec.o pnm.o
 OBJS-$(CONFIG_PGMYUV_ENCODER)          += pnmenc.o pnm.o
 OBJS-$(CONFIG_PGSSUB_DECODER)          += pgssubdec.o
 OBJS-$(CONFIG_PICTOR_DECODER)          += pictordec.o cga_data.o
-OBJS-$(CONFIG_PNG_DECODER)             += png.o pngdec.o pngdsp.o
+OBJS-$(CONFIG_PNG_DECODER)             += png.o pngdec.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_PRORES_DECODER)          += proresdec2.o proresdsp.o
-OBJS-$(CONFIG_PRORES_LGPL_DECODER)     += proresdec_lgpl.o proresdsp.o proresdata.o
-OBJS-$(CONFIG_PRORES_ENCODER)          += proresenc_anatoliy.o
-OBJS-$(CONFIG_PRORES_ANATOLIY_ENCODER) += proresenc_anatoliy.o
-OBJS-$(CONFIG_PRORES_KOSTYA_ENCODER)   += proresenc_kostya.o proresdata.o proresdsp.o
+OBJS-$(CONFIG_PRORES_DECODER)          += proresdec2.o
+OBJS-$(CONFIG_PRORES_LGPL_DECODER)     += proresdec_lgpl.o proresdsp.o
+OBJS-$(CONFIG_PRORES_ENCODER)          += proresenc.o
 OBJS-$(CONFIG_PTX_DECODER)             += ptx.o
 OBJS-$(CONFIG_QCELP_DECODER)           += qcelpdec.o celp_math.o         \
                                           celp_filters.o acelp_vectors.o \
@@ -368,10 +344,8 @@ OBJS-$(CONFIG_R10K_ENCODER)            += r210enc.o
 OBJS-$(CONFIG_R210_DECODER)            += r210dec.o
 OBJS-$(CONFIG_R210_ENCODER)            += r210enc.o
 OBJS-$(CONFIG_RA_144_DECODER)          += ra144dec.o ra144.o celp_filters.o
-OBJS-$(CONFIG_RA_144_ENCODER)          += ra144enc.o ra144.o celp_filters.o \
-                                          audio_frame_queue.o
+OBJS-$(CONFIG_RA_144_ENCODER)          += ra144enc.o ra144.o celp_filters.o
 OBJS-$(CONFIG_RA_288_DECODER)          += ra288.o celp_math.o celp_filters.o
-OBJS-$(CONFIG_RALF_DECODER)            += ralf.o
 OBJS-$(CONFIG_RAWVIDEO_DECODER)        += rawdec.o
 OBJS-$(CONFIG_RAWVIDEO_ENCODER)        += rawenc.o
 OBJS-$(CONFIG_RL2_DECODER)             += rl2.o
@@ -413,7 +387,6 @@ OBJS-$(CONFIG_SP5X_DECODER)            += sp5xdec.o mjpegdec.o mjpeg.o
 OBJS-$(CONFIG_SRT_DECODER)             += srtdec.o ass.o
 OBJS-$(CONFIG_SRT_ENCODER)             += srtenc.o ass_split.o
 OBJS-$(CONFIG_SUNRAST_DECODER)         += sunrast.o
-OBJS-$(CONFIG_SUNRAST_ENCODER)         += sunrastenc.o
 OBJS-$(CONFIG_SVQ1_DECODER)            += svq1dec.o svq1.o h263.o \
                                           mpegvideo.o error_resilience.o
 OBJS-$(CONFIG_SVQ1_ENCODER)            += svq1enc.o svq1.o    \
@@ -448,8 +421,6 @@ OBJS-$(CONFIG_V210_DECODER)            += v210dec.o
 OBJS-$(CONFIG_V210_ENCODER)            += v210enc.o
 OBJS-$(CONFIG_V308_DECODER)            += v308dec.o
 OBJS-$(CONFIG_V308_ENCODER)            += v308enc.o
-OBJS-$(CONFIG_V408_DECODER)            += v408dec.o
-OBJS-$(CONFIG_V408_ENCODER)            += v408enc.o
 OBJS-$(CONFIG_V410_DECODER)            += v410dec.o
 OBJS-$(CONFIG_V410_ENCODER)            += v410enc.o
 OBJS-$(CONFIG_V210X_DECODER)           += v210x.o
@@ -477,12 +448,12 @@ OBJS-$(CONFIG_VP6_DECODER)             += vp6.o vp56.o vp56data.o vp56dsp.o \
 OBJS-$(CONFIG_VP8_DECODER)             += vp8.o vp8dsp.o vp56rac.o
 OBJS-$(CONFIG_VQA_DECODER)             += vqavideo.o
 OBJS-$(CONFIG_WAVPACK_DECODER)         += wavpack.o
-OBJS-$(CONFIG_WMALOSSLESS_DECODER)     += wmalosslessdec.o wma_common.o
-OBJS-$(CONFIG_WMAPRO_DECODER)          += wmaprodec.o wma.o wma_common.o
-OBJS-$(CONFIG_WMAV1_DECODER)           += wmadec.o wma.o wma_common.o aactab.o
-OBJS-$(CONFIG_WMAV1_ENCODER)           += wmaenc.o wma.o wma_common.o aactab.o
-OBJS-$(CONFIG_WMAV2_DECODER)           += wmadec.o wma.o wma_common.o aactab.o
-OBJS-$(CONFIG_WMAV2_ENCODER)           += wmaenc.o wma.o wma_common.o aactab.o
+OBJS-$(CONFIG_WMALOSSLESS_DECODER)     += wmalosslessdec.o wma.o
+OBJS-$(CONFIG_WMAPRO_DECODER)          += wmaprodec.o wma.o
+OBJS-$(CONFIG_WMAV1_DECODER)           += wmadec.o wma.o aactab.o
+OBJS-$(CONFIG_WMAV1_ENCODER)           += wmaenc.o wma.o aactab.o
+OBJS-$(CONFIG_WMAV2_DECODER)           += wmadec.o wma.o aactab.o
+OBJS-$(CONFIG_WMAV2_ENCODER)           += wmaenc.o wma.o aactab.o
 OBJS-$(CONFIG_WMAVOICE_DECODER)        += wmavoice.o \
                                           celp_math.o celp_filters.o \
                                           acelp_vectors.o acelp_filters.o
@@ -491,7 +462,7 @@ OBJS-$(CONFIG_WMV2_DECODER)            += wmv2dec.o wmv2.o        \
                                           msmpeg4.o msmpeg4data.o \
                                           intrax8.o intrax8dsp.o
 OBJS-$(CONFIG_WMV2_ENCODER)            += wmv2enc.o wmv2.o \
-                                          msmpeg4.o msmpeg4enc.o msmpeg4data.o \
+                                          msmpeg4.o msmpeg4data.o \
                                           mpeg4videodec.o ituh263dec.o h263dec.o
 OBJS-$(CONFIG_WNV1_DECODER)            += wnv1.o
 OBJS-$(CONFIG_WS_SND1_DECODER)         += ws-snd1.o
@@ -499,8 +470,6 @@ OBJS-$(CONFIG_XAN_DPCM_DECODER)        += dpcm.o
 OBJS-$(CONFIG_XAN_WC3_DECODER)         += xan.o
 OBJS-$(CONFIG_XAN_WC4_DECODER)         += xxan.o
 OBJS-$(CONFIG_XBIN_DECODER)            += bintext.o cga_data.o
-OBJS-$(CONFIG_XBM_DECODER)             += xbmdec.o
-OBJS-$(CONFIG_XBM_ENCODER)             += xbmenc.o
 OBJS-$(CONFIG_XL_DECODER)              += xl.o
 OBJS-$(CONFIG_XSUB_DECODER)            += xsubdec.o
 OBJS-$(CONFIG_XSUB_ENCODER)            += xsubenc.o
@@ -511,7 +480,6 @@ OBJS-$(CONFIG_Y41P_ENCODER)            += y41penc.o
 OBJS-$(CONFIG_YOP_DECODER)             += yop.o
 OBJS-$(CONFIG_YUV4_DECODER)            += yuv4dec.o
 OBJS-$(CONFIG_YUV4_ENCODER)            += yuv4enc.o
-OBJS-$(CONFIG_ZEROCODEC_DECODER)       += zerocodec.o
 OBJS-$(CONFIG_ZLIB_DECODER)            += lcldec.o
 OBJS-$(CONFIG_ZLIB_ENCODER)            += lclenc.o
 OBJS-$(CONFIG_ZMBV_DECODER)            += zmbv.o
@@ -610,11 +578,11 @@ OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER)       += adpcmenc.o adpcm_data.o
 # libavformat dependencies
 OBJS-$(CONFIG_ADTS_MUXER)              += mpeg4audio.o
 OBJS-$(CONFIG_ADX_DEMUXER)             += adx.o
-OBJS-$(CONFIG_CAF_DEMUXER)             += mpeg4audio.o mpegaudiodata.o
-OBJS-$(CONFIG_DV_DEMUXER)              += dv_profile.o
-OBJS-$(CONFIG_DV_MUXER)                += dv_profile.o timecode.o
-OBJS-$(CONFIG_FLAC_DEMUXER)            += flacdec.o flacdata.o flac.o vorbis_data.o \
-                                          vorbis_parser.o xiph.o
+OBJS-$(CONFIG_CAF_DEMUXER)             += mpeg4audio.o mpegaudiodata.o  \
+                                          ac3tab.o
+OBJS-$(CONFIG_DV_DEMUXER)              += dvdata.o
+OBJS-$(CONFIG_DV_MUXER)                += dvdata.o timecode.o
+OBJS-$(CONFIG_FLAC_DEMUXER)            += flacdec.o flacdata.o flac.o vorbis_data.o
 OBJS-$(CONFIG_FLAC_MUXER)              += flacdec.o flacdata.o flac.o vorbis_data.o
 OBJS-$(CONFIG_FLV_DEMUXER)             += mpeg4audio.o
 OBJS-$(CONFIG_GXF_DEMUXER)             += mpeg12data.o
@@ -626,7 +594,6 @@ OBJS-$(CONFIG_MATROSKA_DEMUXER)        += mpeg4audio.o mpegaudiodata.o
 OBJS-$(CONFIG_MATROSKA_MUXER)          += xiph.o mpeg4audio.o \
                                           flacdec.o flacdata.o flac.o \
                                           mpegaudiodata.o vorbis_data.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_MUXER)               += mpeg4audio.o mpegaudiodata.o
@@ -635,8 +602,7 @@ OBJS-$(CONFIG_MPEGTS_DEMUXER)          += mpeg4audio.o mpegaudiodata.o
 OBJS-$(CONFIG_MXF_MUXER)               += timecode.o
 OBJS-$(CONFIG_NUT_MUXER)               += mpegaudiodata.o
 OBJS-$(CONFIG_OGG_DEMUXER)             += flacdec.o flacdata.o flac.o \
-                                          dirac.o mpeg12data.o vorbis_parser.o \
-                                          xiph.o vorbis_data.o
+                                          dirac.o mpeg12data.o vorbis_data.o
 OBJS-$(CONFIG_OGG_MUXER)               += xiph.o flacdec.o flacdata.o flac.o \
                                           vorbis_data.o
 OBJS-$(CONFIG_RTP_MUXER)               += mpeg4audio.o mpegvideo.o xiph.o
@@ -649,40 +615,38 @@ OBJS-$(CONFIG_WTV_DEMUXER)             += mpeg4audio.o mpegaudiodata.o
 # external codec libraries
 OBJS-$(CONFIG_LIBAACPLUS_ENCODER)         += libaacplus.o
 OBJS-$(CONFIG_LIBCELT_DECODER)            += libcelt_dec.o
-OBJS-$(CONFIG_LIBFAAC_ENCODER)            += libfaac.o audio_frame_queue.o
+OBJS-$(CONFIG_LIBDIRAC_DECODER)           += libdiracdec.o
+OBJS-$(CONFIG_LIBDIRAC_ENCODER)           += libdiracenc.o libdirac_libschro.o
+OBJS-$(CONFIG_LIBFAAC_ENCODER)            += libfaac.o
 OBJS-$(CONFIG_LIBGSM_DECODER)             += libgsm.o
 OBJS-$(CONFIG_LIBGSM_ENCODER)             += libgsm.o
 OBJS-$(CONFIG_LIBGSM_MS_DECODER)          += libgsm.o
 OBJS-$(CONFIG_LIBGSM_MS_ENCODER)          += libgsm.o
-OBJS-$(CONFIG_LIBMP3LAME_ENCODER)         += libmp3lame.o mpegaudiodecheader.o \
-                                             audio_frame_queue.o
-OBJS-$(CONFIG_LIBOPENCORE_AMRNB_DECODER)  += libopencore-amr.o \
-                                             audio_frame_queue.o
-OBJS-$(CONFIG_LIBOPENCORE_AMRNB_ENCODER)  += libopencore-amr.o \
-                                             audio_frame_queue.o
+OBJS-$(CONFIG_LIBMP3LAME_ENCODER)         += libmp3lame.o
+OBJS-$(CONFIG_LIBOPENCORE_AMRNB_DECODER)  += libopencore-amr.o
+OBJS-$(CONFIG_LIBOPENCORE_AMRNB_ENCODER)  += libopencore-amr.o
 OBJS-$(CONFIG_LIBOPENCORE_AMRWB_DECODER)  += libopencore-amr.o
 OBJS-$(CONFIG_LIBOPENJPEG_DECODER)        += libopenjpegdec.o
 OBJS-$(CONFIG_LIBOPENJPEG_ENCODER)        += libopenjpegenc.o
 OBJS-$(CONFIG_LIBSCHROEDINGER_DECODER)    += libschroedingerdec.o \
-                                             libschroedinger.o
+                                             libschroedinger.o    \
+                                             libdirac_libschro.o
 OBJS-$(CONFIG_LIBSCHROEDINGER_ENCODER)    += libschroedingerenc.o \
-                                             libschroedinger.o
+                                             libschroedinger.o    \
+                                             libdirac_libschro.o
 OBJS-$(CONFIG_LIBSPEEX_DECODER)           += libspeexdec.o
-OBJS-$(CONFIG_LIBSPEEX_ENCODER)           += libspeexenc.o audio_frame_queue.o
+OBJS-$(CONFIG_LIBSPEEX_ENCODER)           += libspeexenc.o
 OBJS-$(CONFIG_LIBSTAGEFRIGHT_H264_DECODER)+= libstagefright.o
 OBJS-$(CONFIG_LIBTHEORA_ENCODER)          += libtheoraenc.o
-OBJS-$(CONFIG_LIBUTVIDEO_DECODER)         += libutvideodec.o
-OBJS-$(CONFIG_LIBUTVIDEO_ENCODER)         += libutvideoenc.o
-OBJS-$(CONFIG_LIBVO_AACENC_ENCODER)       += libvo-aacenc.o mpeg4audio.o \
-                                             audio_frame_queue.o
+OBJS-$(CONFIG_LIBUTVIDEO_DECODER)         += libutvideo.o
+OBJS-$(CONFIG_LIBVO_AACENC_ENCODER)       += libvo-aacenc.o mpeg4audio.o
 OBJS-$(CONFIG_LIBVO_AMRWBENC_ENCODER)     += libvo-amrwbenc.o
-OBJS-$(CONFIG_LIBVORBIS_ENCODER)          += libvorbis.o audio_frame_queue.o \
-                                             vorbis_data.o vorbis_parser.o
+OBJS-$(CONFIG_LIBVORBIS_ENCODER)          += libvorbis.o vorbis_data.o
 OBJS-$(CONFIG_LIBVPX_DECODER)             += libvpxdec.o
 OBJS-$(CONFIG_LIBVPX_ENCODER)             += libvpxenc.o
 OBJS-$(CONFIG_LIBX264_ENCODER)            += libx264.o
 OBJS-$(CONFIG_LIBXAVS_ENCODER)            += libxavs.o
-OBJS-$(CONFIG_LIBXVID_ENCODER)            += libxvid.o
+OBJS-$(CONFIG_LIBXVID)                    += libxvidff.o libxvid_rc.o
 
 # parsers
 OBJS-$(CONFIG_AAC_PARSER)              += aac_parser.o aac_ac3_parser.o \
@@ -691,7 +655,6 @@ OBJS-$(CONFIG_AC3_PARSER)              += ac3_parser.o ac3tab.o \
                                           aac_ac3_parser.o
 OBJS-$(CONFIG_ADX_PARSER)              += adx_parser.o adx.o
 OBJS-$(CONFIG_CAVSVIDEO_PARSER)        += cavs_parser.o
-OBJS-$(CONFIG_COOK_PARSER)             += cook_parser.o
 OBJS-$(CONFIG_DCA_PARSER)              += dca_parser.o
 OBJS-$(CONFIG_DIRAC_PARSER)            += dirac_parser.o
 OBJS-$(CONFIG_DNXHD_PARSER)            += dnxhd_parser.o
@@ -715,7 +678,6 @@ OBJS-$(CONFIG_MPEG4VIDEO_PARSER)       += mpeg4video_parser.o h263.o \
                                           mpegvideo.o error_resilience.o \
                                           mpeg4videodec.o mpeg4video.o \
                                           ituh263dec.o h263dec.o
-OBJS-$(CONFIG_PNG_PARSER)              += png_parser.o
 OBJS-$(CONFIG_MPEGAUDIO_PARSER)        += mpegaudio_parser.o \
                                           mpegaudiodecheader.o mpegaudiodata.o
 OBJS-$(CONFIG_MPEGVIDEO_PARSER)        += mpegvideo_parser.o    \
@@ -727,7 +689,6 @@ OBJS-$(CONFIG_RV40_PARSER)             += rv34_parser.o
 OBJS-$(CONFIG_VC1_PARSER)              += vc1_parser.o vc1.o vc1data.o \
                                           msmpeg4.o msmpeg4data.o mpeg4video.o \
                                           h263.o mpegvideo.o error_resilience.o
-OBJS-$(CONFIG_VORBIS_PARSER)           += vorbis_parser.o xiph.o
 OBJS-$(CONFIG_VP3_PARSER)              += vp3_parser.o
 OBJS-$(CONFIG_VP8_PARSER)              += vp8_parser.o
 
@@ -753,6 +714,8 @@ OBJS-$(HAVE_PTHREADS)                  += pthread.o
 OBJS-$(HAVE_W32THREADS)                += pthread.o
 OBJS-$(HAVE_OS2THREADS)                += pthread.o
 
+OBJS-$(CONFIG_MLIB)                    += mlib/dsputil_mlib.o           \
+
 # inverse.o contains the ff_inverse table definition, which is used by
 # the FASTDIV macro (from libavutil); since referencing the external
 # table has a negative effect on performance, copy it in libavcodec as
@@ -762,43 +725,27 @@ OBJS-$(!CONFIG_SMALL)                  += inverse.o
 SKIPHEADERS                            += %_tablegen.h                  \
                                           %_tables.h                    \
                                           aac_tablegen_decl.h           \
-                                          codec_names.h                 \
                                           fft-internal.h                \
                                           tableprint.h                  \
-                                          $(ARCH)/vp56_arith.h          \
-
+                                          $(ARCH)/vp56_arith.h
 SKIPHEADERS-$(CONFIG_DXVA2)            += dxva2.h dxva2_internal.h
+SKIPHEADERS-$(CONFIG_LIBDIRAC)         += libdirac.h
 SKIPHEADERS-$(CONFIG_LIBSCHROEDINGER)  += libschroedinger.h
-SKIPHEADERS-$(CONFIG_LIBUTVIDEO)       += libutvideo.h
-SKIPHEADERS-$(CONFIG_MPEG_XVMC_DECODER) += xvmc.h
 SKIPHEADERS-$(CONFIG_VAAPI)            += vaapi_internal.h
-SKIPHEADERS-$(CONFIG_VDA)              += vda.h vda_internal.h
+SKIPHEADERS-$(CONFIG_VDA)              += vda_internal.h
 SKIPHEADERS-$(CONFIG_VDPAU)            += vdpau.h
-SKIPHEADERS-$(HAVE_OS2THREADS)         += os2threads.h
+SKIPHEADERS-$(CONFIG_XVMC)             += xvmc.h
 SKIPHEADERS-$(HAVE_W32THREADS)         += w32pthreads.h
 
-TESTPROGS = cabac                                                       \
-            dct                                                         \
-            fft                                                         \
-            fft-fixed                                                   \
-            golomb                                                      \
-            iirfilter                                                   \
-            rangecoder                                                  \
-            snowenc                                                     \
-
+TESTPROGS = cabac dct fft fft-fixed h264 iirfilter rangecoder snowenc
 TESTPROGS-$(HAVE_MMX) += motion
 TESTOBJS = dctref.o
 
-HOSTPROGS = aac_tablegen                                                \
-            aacps_tablegen                                              \
-            cbrt_tablegen                                               \
-            cos_tablegen                                                \
-            dv_tablegen                                                 \
-            motionpixels_tablegen                                       \
-            mpegaudio_tablegen                                          \
-            pcm_tablegen                                                \
-            qdm2_tablegen                                               \
-            sinewin_tablegen                                            \
+HOSTPROGS = aac_tablegen aacps_tablegen cbrt_tablegen cos_tablegen      \
+            dv_tablegen motionpixels_tablegen mpegaudio_tablegen        \
+            pcm_tablegen qdm2_tablegen sinewin_tablegen
+
+DIRS = alpha arm bfin mlib ppc ps2 sh4 sparc x86
 
 CLEANFILES = *_tables.c *_tables.h *_tablegen$(HOSTEXESUF)
 

libavcodec/a64enc.h

@@ -50,9 +50,6 @@ typedef struct A64Context {
     uint8_t *mc_colram;
     uint8_t *mc_palette;
     int mc_pal_size;
-
-    /* pts of the next packet that will be output */
-    int64_t next_pts;
 } A64Context;
 
 #endif /* AVCODEC_A64ENC_H */

libavcodec/a64multienc.c

@@ -28,7 +28,6 @@
 #include "a64colors.h"
 #include "a64tables.h"
 #include "elbg.h"
-#include "internal.h"
 #include "libavutil/intreadwrite.h"
 
 #define DITHERSTEPS   8
@@ -222,8 +221,6 @@ static av_cold int a64multi_init_encoder(AVCodecContext *avctx)
     if (!avctx->codec_tag)
          avctx->codec_tag = AV_RL32("a64m");
 
-    c->next_pts = AV_NOPTS_VALUE;
-
     return 0;
 }
 
@@ -242,19 +239,19 @@ static void a64_compress_colram(unsigned char *buf, int *charmap, uint8_t *colra
     }
 }
 
-static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
-                                 const AVFrame *pict, int *got_packet)
+static int a64multi_encode_frame(AVCodecContext *avctx, unsigned char *buf,
+                                 int buf_size, void *data)
 {
     A64Context *c = avctx->priv_data;
-    AVFrame *const p = &c->picture;
+    AVFrame *pict = data;
+    AVFrame *const p = (AVFrame *) & c->picture;
 
     int frame;
     int x, y;
     int b_height;
     int b_width;
 
-    int req_size, ret;
-    uint8_t *buf = NULL;
+    int req_size;
 
     int *charmap     = c->mc_charmap;
     uint8_t *colram  = c->mc_colram;
@@ -277,7 +274,7 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
     }
 
     /* no data, means end encoding asap */
-    if (!pict) {
+    if (!data) {
         /* all done, end encoding */
         if (!c->mc_lifetime) return 0;
         /* no more frames in queue, prepare to flush remaining frames */
@@ -295,8 +292,6 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
             p->key_frame = 1;
             to_meta_with_crop(avctx, p, meta + 32000 * c->mc_frame_counter);
             c->mc_frame_counter++;
-            if (c->next_pts == AV_NOPTS_VALUE)
-                c->next_pts = pict->pts;
             /* lifetime is not reached so wait for next frame first */
             return 0;
         }
@@ -307,11 +302,6 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
         req_size = 0;
         /* any frames to encode? */
         if (c->mc_lifetime) {
-            req_size = charset_size + c->mc_lifetime*(screen_size + colram_size);
-            if ((ret = ff_alloc_packet2(avctx, pkt, req_size)) < 0)
-                return ret;
-            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);
@@ -320,12 +310,15 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
             render_charset(avctx, charset, colram);
 
             /* copy charset to buf */
-            memcpy(buf, charset, charset_size);
+            memcpy(buf,charset, charset_size);
 
             /* advance pointers */
             buf      += charset_size;
             charset  += charset_size;
+            req_size += charset_size;
         }
+        /* no charset so clean buf */
+        else memset(buf, 0, charset_size);
 
         /* write x frames to buf */
         for (frame = 0; frame < c->mc_lifetime; frame++) {
@@ -358,12 +351,11 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
         /* reset counter */
         c->mc_frame_counter = 0;
 
-        pkt->pts = pkt->dts = c->next_pts;
-        c->next_pts         = AV_NOPTS_VALUE;
-
-        pkt->size   = req_size;
-        pkt->flags |= AV_PKT_FLAG_KEY;
-        *got_packet = !!req_size;
+        if (req_size > buf_size) {
+            av_log(avctx, AV_LOG_ERROR, "buf size too small (need %d, got %d)\n", req_size, buf_size);
+            return -1;
+        }
+        return req_size;
     }
     return 0;
 }
@@ -374,7 +366,7 @@ AVCodec ff_a64multi_encoder = {
     .id             = CODEC_ID_A64_MULTI,
     .priv_data_size = sizeof(A64Context),
     .init           = a64multi_init_encoder,
-    .encode2        = a64multi_encode_frame,
+    .encode         = a64multi_encode_frame,
     .close          = a64multi_close_encoder,
     .pix_fmts       = (const enum PixelFormat[]) {PIX_FMT_GRAY8, PIX_FMT_NONE},
     .long_name      = NULL_IF_CONFIG_SMALL("Multicolor charset for Commodore 64"),
@@ -387,7 +379,7 @@ AVCodec ff_a64multi5_encoder = {
     .id             = CODEC_ID_A64_MULTI5,
     .priv_data_size = sizeof(A64Context),
     .init           = a64multi_init_encoder,
-    .encode2        = a64multi_encode_frame,
+    .encode         = a64multi_encode_frame,
     .close          = a64multi_close_encoder,
     .pix_fmts       = (const enum PixelFormat[]) {PIX_FMT_GRAY8, PIX_FMT_NONE},
     .long_name      = NULL_IF_CONFIG_SMALL("Multicolor charset for Commodore 64, extended with 5th color (colram)"),

libavcodec/aac.h

@@ -112,15 +112,6 @@ enum OCStatus {
     OC_LOCKED,      ///< Output configuration locked in place
 };
 
-typedef struct {
-    MPEG4AudioConfig m4ac;
-    uint8_t layout_map[MAX_ELEM_ID*4][3];
-    int layout_map_tags;
-    int channels;
-    uint64_t channel_layout;
-    enum OCStatus status;
-} OutputConfiguration;
-
 /**
  * Predictor State
  */
@@ -263,6 +254,8 @@ typedef struct {
     AVCodecContext *avctx;
     AVFrame frame;
 
+    MPEG4AudioConfig m4ac;
+
     int is_saved;                 ///< Set if elements have stored overlap from previous frame.
     DynamicRangeControl che_drc;
 
@@ -270,6 +263,9 @@ typedef struct {
      * @name Channel element related data
      * @{
      */
+    enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
+                                                   *   first index as the first 4 raw data block types
+                                                   */
     ChannelElement          *che[4][MAX_ELEM_ID];
     ChannelElement  *tag_che_map[4][MAX_ELEM_ID];
     int tags_mapped;
@@ -304,7 +300,7 @@ typedef struct {
 
     DECLARE_ALIGNED(32, float, temp)[128];
 
-    OutputConfiguration oc[2];
+    enum OCStatus output_configured;
     int warned_num_aac_frames;
 } AACContext;
 

libavcodec/aac_ac3_parser.c

@@ -93,7 +93,7 @@ get_next:
             avctx->channels = s->channels;
             avctx->channel_layout = s->channel_layout;
         }
-        s1->duration = s->samples;
+        avctx->frame_size = s->samples;
         avctx->audio_service_type = s->service_type;
     }
 

libavcodec/aaccoder.c

@@ -713,18 +713,16 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
                                           const float lambda)
 {
     int start = 0, i, w, w2, g;
-    int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->channels;
-    float dists[128] = { 0 }, uplims[128];
+    int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->channels * (lambda / 120.f);
+    float dists[128], uplims[128];
     float maxvals[128];
     int fflag, minscaler;
     int its  = 0;
     int allz = 0;
     float minthr = INFINITY;
 
-    // for values above this the decoder might end up in an endless loop
-    // due to always having more bits than what can be encoded.
-    destbits = FFMIN(destbits, 5800);
     //XXX: some heuristic to determine initial quantizers will reduce search time
+    memset(dists, 0, sizeof(dists));
     //determine zero bands and upper limits
     for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
         for (g = 0;  g < sce->ics.num_swb; g++) {
@@ -878,7 +876,7 @@ static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
     } else {
         for (w = 0; w < 8; w++) {
             const float *coeffs = sce->coeffs + w*128;
-            curband = start = 0;
+            start = 0;
             for (i = 0; i < 128; i++) {
                 if (i - start >= sce->ics.swb_sizes[curband]) {
                     start += sce->ics.swb_sizes[curband];

libavcodec/aacdectab.h

@@ -80,14 +80,14 @@ static const float * const tns_tmp2_map[4] = {
 
 static const int8_t tags_per_config[16] = { 0, 1, 1, 2, 3, 3, 4, 5, 0, 0, 0, 0, 0, 0, 0, 0 };
 
-static const uint8_t aac_channel_layout_map[7][5][3] = {
-    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, },
-    { { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, },
-    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, },
-    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_SCE, 1, AAC_CHANNEL_BACK }, },
-    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_BACK }, },
-    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE  }, },
-    { { TYPE_SCE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 0, AAC_CHANNEL_FRONT }, { TYPE_CPE, 1, AAC_CHANNEL_FRONT }, { TYPE_CPE, 2, AAC_CHANNEL_BACK }, { TYPE_LFE, 0, AAC_CHANNEL_LFE  }, },
+static const uint8_t aac_channel_layout_map[7][5][2] = {
+    { { TYPE_SCE, 0 }, },
+    { { TYPE_CPE, 0 }, },
+    { { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, },
+    { { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, { TYPE_SCE, 1 }, },
+    { { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, { TYPE_CPE, 1 }, },
+    { { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, { TYPE_LFE, 0 }, { TYPE_CPE, 1 }, },
+    { { TYPE_CPE, 0 }, { TYPE_SCE, 0 }, { TYPE_LFE, 0 }, { TYPE_CPE, 2 }, { TYPE_CPE, 1 }, },
 };
 
 static const uint64_t aac_channel_layout[8] = {
@@ -97,7 +97,7 @@ static const uint64_t aac_channel_layout[8] = {
     AV_CH_LAYOUT_4POINT0,
     AV_CH_LAYOUT_5POINT0_BACK,
     AV_CH_LAYOUT_5POINT1_BACK,
-    AV_CH_LAYOUT_7POINT1_WIDE_BACK,
+    AV_CH_LAYOUT_7POINT1_WIDE,
     0,
 };
 

libavcodec/aacenc.c

@@ -34,7 +34,6 @@
 #include "avcodec.h"
 #include "put_bits.h"
 #include "dsputil.h"
-#include "internal.h"
 #include "mpeg4audio.h"
 #include "kbdwin.h"
 #include "sinewin.h"
@@ -145,7 +144,7 @@ static const uint8_t aac_chan_configs[6][5] = {
 };
 
 /**
- * Table to remap channels from libavcodec's default order to AAC order.
+ * Table to remap channels from Libav's default order to AAC order.
  */
 static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
     { 0 },
@@ -224,9 +223,8 @@ WINDOW_FUNC(eight_short)
     const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
     const float *in = audio + 448;
     float *out = sce->ret;
-    int w;
 
-    for (w = 0; w < 8; w++) {
+    for (int w = 0; w < 8; w++) {
         dsp->vector_fmul        (out, in, w ? pwindow : swindow, 128);
         out += 128;
         in  += 128;
@@ -475,9 +473,10 @@ static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
 
 /*
  * Deinterleave input samples.
- * Channels are reordered from libavcodec's default order to AAC order.
+ * Channels are reordered from Libav's default order to AAC order.
  */
-static void deinterleave_input_samples(AACEncContext *s, const AVFrame *frame)
+static void deinterleave_input_samples(AACEncContext *s,
+                                       const float *samples)
 {
     int ch, i;
     const int sinc = s->channels;
@@ -485,46 +484,38 @@ static void deinterleave_input_samples(AACEncContext *s, const AVFrame *frame)
 
     /* deinterleave and remap input samples */
     for (ch = 0; ch < sinc; ch++) {
+        const float *sptr = samples + channel_map[ch];
+
         /* copy last 1024 samples of previous frame to the start of the current frame */
         memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
 
         /* deinterleave */
-        i = 2048;
-        if (frame) {
-            const float *sptr = ((const float *)frame->data[0]) + channel_map[ch];
-            for (; i < 2048 + frame->nb_samples; i++) {
-                s->planar_samples[ch][i] = *sptr;
-                sptr += sinc;
-            }
+        for (i = 2048; i < 3072; i++) {
+            s->planar_samples[ch][i] = *sptr;
+            sptr += sinc;
         }
-        memset(&s->planar_samples[ch][i], 0,
-               (3072 - i) * sizeof(s->planar_samples[0][0]));
     }
 }
 
-static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
-                            const AVFrame *frame, int *got_packet_ptr)
+static int aac_encode_frame(AVCodecContext *avctx,
+                            uint8_t *frame, int buf_size, void *data)
 {
     AACEncContext *s = avctx->priv_data;
     float **samples = s->planar_samples, *samples2, *la, *overlap;
     ChannelElement *cpe;
-    int i, ch, w, g, chans, tag, start_ch, ret;
+    int i, ch, w, g, chans, tag, start_ch;
     int chan_el_counter[4];
     FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
 
-    if (s->last_frame == 2)
+    if (s->last_frame)
         return 0;
 
-    /* add current frame to queue */
-    if (frame) {
-        if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
-            return ret;
+    if (data) {
+        deinterleave_input_samples(s, data);
+        if (s->psypp)
+            ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
     }
 
-    deinterleave_input_samples(s, frame);
-    if (s->psypp)
-        ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
-
     if (!avctx->frame_number)
         return 0;
 
@@ -540,7 +531,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
             overlap  = &samples[cur_channel][0];
             samples2 = overlap + 1024;
             la       = samples2 + (448+64);
-            if (!frame)
+            if (!data)
                 la = NULL;
             if (tag == TYPE_LFE) {
                 wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;
@@ -571,15 +562,9 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         }
         start_ch += chans;
     }
-    if ((ret = ff_alloc_packet2(avctx, avpkt, 768 * s->channels))) {
-        av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
-        return ret;
-    }
     do {
         int frame_bits;
-
-        init_put_bits(&s->pb, avpkt->data, avpkt->size);
-
+        init_put_bits(&s->pb, frame, buf_size*8);
         if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
             put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
         start_ch = 0;
@@ -659,15 +644,10 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         s->lambda = FFMIN(s->lambda, 65536.f);
     }
 
-    if (!frame)
-        s->last_frame++;
+    if (!data)
+        s->last_frame = 1;
 
-    ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
-                       &avpkt->duration);
-
-    avpkt->size = put_bits_count(&s->pb) >> 3;
-    *got_packet_ptr = 1;
-    return 0;
+    return put_bits_count(&s->pb)>>3;
 }
 
 static av_cold int aac_encode_end(AVCodecContext *avctx)
@@ -681,10 +661,6 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)
         ff_psy_preprocess_end(s->psypp);
     av_freep(&s->buffer.samples);
     av_freep(&s->cpe);
-    ff_af_queue_close(&s->afq);
-#if FF_API_OLD_ENCODE_AUDIO
-    av_freep(&avctx->coded_frame);
-#endif
     return 0;
 }
 
@@ -692,7 +668,7 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
 {
     int ret = 0;
 
-    ff_dsputil_init(&s->dsp, avctx);
+    dsputil_init(&s->dsp, avctx);
 
     // window init
     ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
@@ -710,19 +686,13 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
 
 static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
 {
-    int ch;
     FF_ALLOCZ_OR_GOTO(avctx, s->buffer.samples, 3 * 1024 * s->channels * sizeof(s->buffer.samples[0]), alloc_fail);
     FF_ALLOCZ_OR_GOTO(avctx, s->cpe, sizeof(ChannelElement) * s->chan_map[0], alloc_fail);
     FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
 
-    for(ch = 0; ch < s->channels; ch++)
+    for(int ch = 0; ch < s->channels; ch++)
         s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
 
-#if FF_API_OLD_ENCODE_AUDIO
-    if (!(avctx->coded_frame = avcodec_alloc_frame()))
-        goto alloc_fail;
-#endif
-
     return 0;
 alloc_fail:
     return AVERROR(ENOMEM);
@@ -784,9 +754,6 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     for (i = 0; i < 428; i++)
         ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));
 
-    avctx->delay = 1024;
-    ff_af_queue_init(avctx, &s->afq);
-
     return 0;
 fail:
     aac_encode_end(avctx);
@@ -816,13 +783,10 @@ AVCodec ff_aac_encoder = {
     .id             = CODEC_ID_AAC,
     .priv_data_size = sizeof(AACEncContext),
     .init           = aac_encode_init,
-    .encode2        = aac_encode_frame,
+    .encode         = aac_encode_frame,
     .close          = aac_encode_end,
-    .supported_samplerates = avpriv_mpeg4audio_sample_rates,
-    .capabilities   = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |
-                      CODEC_CAP_EXPERIMENTAL,
-    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
-                                                     AV_SAMPLE_FMT_NONE },
-    .long_name      = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
-    .priv_class     = &aacenc_class,
+    .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
+    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
+    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
+    .priv_class = &aacenc_class,
 };

libavcodec/aacenc.h

@@ -27,7 +27,7 @@
 #include "dsputil.h"
 
 #include "aac.h"
-#include "audio_frame_queue.h"
+
 #include "psymodel.h"
 
 #define AAC_CODER_NB 4
@@ -74,7 +74,6 @@ typedef struct AACEncContext {
     int cur_channel;
     int last_frame;
     float lambda;
-    AudioFrameQueue afq;
     DECLARE_ALIGNED(16, int,   qcoefs)[96];      ///< quantized coefficients
     DECLARE_ALIGNED(32, float, scoefs)[1024];    ///< scaled coefficients
 

libavcodec/aacps.c

@@ -27,7 +27,6 @@
 #include "aacps.h"
 #include "aacps_tablegen.h"
 #include "aacpsdata.c"
-#include "dsputil.h"
 
 #define PS_BASELINE 0  ///< Operate in Baseline PS mode
                        ///< Baseline implies 10 or 20 stereo bands,
@@ -285,7 +284,7 @@ err:
 
 /** Split one subband into 2 subsubbands with a symmetric real filter.
  * The filter must have its non-center even coefficients equal to zero. */
-static void hybrid2_re(float (*in)[2], float (*out)[32][2], const float filter[8], int len, int reverse)
+static void hybrid2_re(float (*in)[2], float (*out)[32][2], const float filter[7], int len, int reverse)
 {
     int i, j;
     for (i = 0; i < len; i++, in++) {
@@ -305,14 +304,26 @@ 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(float (*in)[2], float (*out)[32][2], const float (*filter)[7][2], int len)
 {
-    int i;
+    int i, j, ssb;
     int N = 8;
-    LOCAL_ALIGNED_16(float, temp, [8], [2]);
+    float temp[8][2];
 
     for (i = 0; i < len; i++, in++) {
-        dsp->hybrid_analysis(temp, in, filter, 1, N);
+        for (ssb = 0; ssb < N; ssb++) {
+            float sum_re = filter[ssb][6][0] * in[6][0], sum_im = filter[ssb][6][0] * in[6][1];
+            for (j = 0; j < 6; j++) {
+                float in0_re = in[j][0];
+                float in0_im = in[j][1];
+                float in1_re = in[12-j][0];
+                float in1_im = in[12-j][1];
+                sum_re += filter[ssb][j][0] * (in0_re + in1_re) - filter[ssb][j][1] * (in0_im - in1_im);
+                sum_im += filter[ssb][j][0] * (in0_im + in1_im) + filter[ssb][j][1] * (in0_re - in1_re);
+            }
+            temp[ssb][0] = sum_re;
+            temp[ssb][1] = sum_im;
+        }
         out[0][i][0] = temp[6][0];
         out[0][i][1] = temp[6][1];
         out[1][i][0] = temp[7][0];
@@ -328,18 +339,28 @@ 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(float (*in)[2], float (*out)[32][2], const float (*filter)[7][2], int N, int len)
 {
-    int i;
+    int i, j, ssb;
 
     for (i = 0; i < len; i++, in++) {
-        dsp->hybrid_analysis(out[0] + i, in, filter, 32, N);
+        for (ssb = 0; ssb < N; ssb++) {
+            float sum_re = filter[ssb][6][0] * in[6][0], sum_im = filter[ssb][6][0] * in[6][1];
+            for (j = 0; j < 6; j++) {
+                float in0_re = in[j][0];
+                float in0_im = in[j][1];
+                float in1_re = in[12-j][0];
+                float in1_im = in[12-j][1];
+                sum_re += filter[ssb][j][0] * (in0_re + in1_re) - filter[ssb][j][1] * (in0_im - in1_im);
+                sum_im += filter[ssb][j][0] * (in0_im + in1_im) + filter[ssb][j][1] * (in0_re - in1_re);
+            }
+            out[ssb][i][0] = sum_re;
+            out[ssb][i][1] = sum_im;
+        }
     }
 }
 
-static void hybrid_analysis(PSDSPContext *dsp, float out[91][32][2],
-                            float in[5][44][2], float L[2][38][64],
-                            int is34, int len)
+static void hybrid_analysis(float out[91][32][2], float in[5][44][2], float L[2][38][64], int is34, int len)
 {
     int i, j;
     for (i = 0; i < 5; i++) {
@@ -349,17 +370,27 @@ static void hybrid_analysis(PSDSPContext *dsp, float out[91][32][2],
         }
     }
     if (is34) {
-        hybrid4_8_12_cx(dsp, in[0], out,    f34_0_12, 12, len);
-        hybrid4_8_12_cx(dsp, in[1], out+12, f34_1_8,   8, len);
-        hybrid4_8_12_cx(dsp, in[2], out+20, f34_2_4,   4, len);
-        hybrid4_8_12_cx(dsp, in[3], out+24, f34_2_4,   4, len);
-        hybrid4_8_12_cx(dsp, in[4], out+28, f34_2_4,   4, len);
-        dsp->hybrid_analysis_ileave(out + 27, L, 5, len);
+        hybrid4_8_12_cx(in[0], out,    f34_0_12, 12, len);
+        hybrid4_8_12_cx(in[1], out+12, f34_1_8,   8, len);
+        hybrid4_8_12_cx(in[2], out+20, f34_2_4,   4, len);
+        hybrid4_8_12_cx(in[3], out+24, f34_2_4,   4, len);
+        hybrid4_8_12_cx(in[4], out+28, f34_2_4,   4, len);
+        for (i = 0; i < 59; i++) {
+            for (j = 0; j < len; j++) {
+                out[i+32][j][0] = L[0][j][i+5];
+                out[i+32][j][1] = L[1][j][i+5];
+            }
+        }
     } else {
-        hybrid6_cx(dsp, in[0], out, f20_0_8, len);
+        hybrid6_cx(in[0], out, f20_0_8, len);
         hybrid2_re(in[1], out+6, g1_Q2, len, 1);
         hybrid2_re(in[2], out+8, g1_Q2, len, 0);
-        dsp->hybrid_analysis_ileave(out + 7, L, 3, len);
+        for (i = 0; i < 61; i++) {
+            for (j = 0; j < len; j++) {
+                out[i+10][j][0] = L[0][j][i+3];
+                out[i+10][j][1] = L[1][j][i+3];
+            }
+        }
     }
     //update in_buf
     for (i = 0; i < 5; i++) {
@@ -367,8 +398,7 @@ static void hybrid_analysis(PSDSPContext *dsp, float out[91][32][2],
     }
 }
 
-static void hybrid_synthesis(PSDSPContext *dsp, float out[2][38][64],
-                             float in[91][32][2], int is34, int len)
+static void hybrid_synthesis(float out[2][38][64], float in[91][32][2], int is34, int len)
 {
     int i, n;
     if (is34) {
@@ -392,7 +422,12 @@ static void hybrid_synthesis(PSDSPContext *dsp, float out[2][38][64],
                 out[1][n][4] += in[28+i][n][1];
             }
         }
-        dsp->hybrid_synthesis_deint(out, in + 27, 5, len);
+        for (i = 0; i < 59; i++) {
+            for (n = 0; n < len; n++) {
+                out[0][n][i+5] = in[i+32][n][0];
+                out[1][n][i+5] = in[i+32][n][1];
+            }
+        }
     } else {
         for (n = 0; n < len; n++) {
             out[0][n][0] = in[0][n][0] + in[1][n][0] + in[2][n][0] +
@@ -404,7 +439,12 @@ static void hybrid_synthesis(PSDSPContext *dsp, float out[2][38][64],
             out[0][n][2] = in[8][n][0] + in[9][n][0];
             out[1][n][2] = in[8][n][1] + in[9][n][1];
         }
-        dsp->hybrid_synthesis_deint(out, in + 7, 3, len);
+        for (i = 0; i < 61; i++) {
+            for (n = 0; n < len; n++) {
+                out[0][n][i+3] = in[i+10][n][0];
+                out[1][n][i+3] = in[i+10][n][1];
+            }
+        }
     }
 }
 
@@ -608,8 +648,8 @@ static void map_val_20_to_34(float par[PS_MAX_NR_IIDICC])
 
 static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[32][2], int is34)
 {
-    LOCAL_ALIGNED_16(float, power, [34], [PS_QMF_TIME_SLOTS]);
-    LOCAL_ALIGNED_16(float, transient_gain, [34], [PS_QMF_TIME_SLOTS]);
+    float power[34][PS_QMF_TIME_SLOTS] = {{0}};
+    float transient_gain[34][PS_QMF_TIME_SLOTS];
     float *peak_decay_nrg = ps->peak_decay_nrg;
     float *power_smooth = ps->power_smooth;
     float *peak_decay_diff_smooth = ps->peak_decay_diff_smooth;
@@ -621,8 +661,10 @@ static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[3
     const float a_smooth          = 0.25f; ///< Smoothing coefficient
     int i, k, m, n;
     int n0 = 0, nL = 32;
-
-    memset(power, 0, 34 * sizeof(*power));
+    static const int link_delay[] = { 3, 4, 5 };
+    static const float a[] = { 0.65143905753106f,
+                               0.56471812200776f,
+                               0.48954165955695f };
 
     if (is34 != ps->is34bands_old) {
         memset(ps->peak_decay_nrg,         0, sizeof(ps->peak_decay_nrg));
@@ -632,9 +674,11 @@ static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[3
         memset(ps->ap_delay,               0, sizeof(ps->ap_delay));
     }
 
-    for (k = 0; k < NR_BANDS[is34]; k++) {
-        int i = k_to_i[k];
-        ps->dsp.add_squares(power[i], s[k], nL - n0);
+    for (n = n0; n < nL; n++) {
+        for (k = 0; k < NR_BANDS[is34]; k++) {
+            int i = k_to_i[k];
+            power[i][n] += s[k][n][0] * s[k][n][0] + s[k][n][1] * s[k][n][1];
+        }
     }
 
     //Transient detection
@@ -662,31 +706,54 @@ static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[3
     for (k = 0; k < NR_ALLPASS_BANDS[is34]; k++) {
         int b = k_to_i[k];
         float g_decay_slope = 1.f - DECAY_SLOPE * (k - DECAY_CUTOFF[is34]);
+        float ag[PS_AP_LINKS];
         g_decay_slope = av_clipf(g_decay_slope, 0.f, 1.f);
         memcpy(delay[k], delay[k]+nL, PS_MAX_DELAY*sizeof(delay[k][0]));
         memcpy(delay[k]+PS_MAX_DELAY, s[k], numQMFSlots*sizeof(delay[k][0]));
         for (m = 0; m < PS_AP_LINKS; m++) {
             memcpy(ap_delay[k][m],   ap_delay[k][m]+numQMFSlots,           5*sizeof(ap_delay[k][m][0]));
+            ag[m] = a[m] * g_decay_slope;
+        }
+        for (n = n0; n < nL; n++) {
+            float in_re = delay[k][n+PS_MAX_DELAY-2][0] * phi_fract[is34][k][0] -
+                          delay[k][n+PS_MAX_DELAY-2][1] * phi_fract[is34][k][1];
+            float in_im = delay[k][n+PS_MAX_DELAY-2][0] * phi_fract[is34][k][1] +
+                          delay[k][n+PS_MAX_DELAY-2][1] * phi_fract[is34][k][0];
+            for (m = 0; m < PS_AP_LINKS; m++) {
+                float a_re                = ag[m] * in_re;
+                float a_im                = ag[m] * in_im;
+                float link_delay_re       = ap_delay[k][m][n+5-link_delay[m]][0];
+                float link_delay_im       = ap_delay[k][m][n+5-link_delay[m]][1];
+                float fractional_delay_re = Q_fract_allpass[is34][k][m][0];
+                float fractional_delay_im = Q_fract_allpass[is34][k][m][1];
+                ap_delay[k][m][n+5][0] = in_re;
+                ap_delay[k][m][n+5][1] = in_im;
+                in_re = link_delay_re * fractional_delay_re - link_delay_im * fractional_delay_im - a_re;
+                in_im = link_delay_re * fractional_delay_im + link_delay_im * fractional_delay_re - a_im;
+                ap_delay[k][m][n+5][0] += ag[m] * in_re;
+                ap_delay[k][m][n+5][1] += ag[m] * in_im;
+            }
+            out[k][n][0] = transient_gain[b][n] * in_re;
+            out[k][n][1] = transient_gain[b][n] * in_im;
         }
-        ps->dsp.decorrelate(out[k], delay[k] + PS_MAX_DELAY - 2, ap_delay[k],
-                            phi_fract[is34][k], Q_fract_allpass[is34][k],
-                            transient_gain[b], g_decay_slope, nL - n0);
     }
     for (; k < SHORT_DELAY_BAND[is34]; k++) {
-        int i = k_to_i[k];
         memcpy(delay[k], delay[k]+nL, PS_MAX_DELAY*sizeof(delay[k][0]));
         memcpy(delay[k]+PS_MAX_DELAY, s[k], numQMFSlots*sizeof(delay[k][0]));
-        //H = delay 14
-        ps->dsp.mul_pair_single(out[k], delay[k] + PS_MAX_DELAY - 14,
-                                transient_gain[i], nL - n0);
+        for (n = n0; n < nL; n++) {
+            //H = delay 14
+            out[k][n][0] = transient_gain[k_to_i[k]][n] * delay[k][n+PS_MAX_DELAY-14][0];
+            out[k][n][1] = transient_gain[k_to_i[k]][n] * delay[k][n+PS_MAX_DELAY-14][1];
+        }
     }
     for (; k < NR_BANDS[is34]; k++) {
-        int i = k_to_i[k];
         memcpy(delay[k], delay[k]+nL, PS_MAX_DELAY*sizeof(delay[k][0]));
         memcpy(delay[k]+PS_MAX_DELAY, s[k], numQMFSlots*sizeof(delay[k][0]));
-        //H = delay 1
-        ps->dsp.mul_pair_single(out[k], delay[k] + PS_MAX_DELAY - 1,
-                                transient_gain[i], nL - n0);
+        for (n = n0; n < nL; n++) {
+            //H = delay 1
+            out[k][n][0] = transient_gain[k_to_i[k]][n] * delay[k][n+PS_MAX_DELAY-1][0];
+            out[k][n][1] = transient_gain[k_to_i[k]][n] * delay[k][n+PS_MAX_DELAY-1][1];
+        }
     }
 }
 
@@ -730,7 +797,7 @@ static void remap20(int8_t (**p_par_mapped)[PS_MAX_NR_IIDICC],
 
 static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2], int is34)
 {
-    int e, b, k;
+    int e, b, k, n;
 
     float (*H11)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H11;
     float (*H12)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H12;
@@ -842,52 +909,78 @@ static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2
             H22[0][e+1][b] = h22;
         }
         for (k = 0; k < NR_BANDS[is34]; k++) {
-            float h[2][4];
-            float h_step[2][4];
+            float h11r, h12r, h21r, h22r;
+            float h11i, h12i, h21i, h22i;
+            float h11r_step, h12r_step, h21r_step, h22r_step;
+            float h11i_step, h12i_step, h21i_step, h22i_step;
             int start = ps->border_position[e];
             int stop  = ps->border_position[e+1];
             float width = 1.f / (stop - start);
             b = k_to_i[k];
-            h[0][0] = H11[0][e][b];
-            h[0][1] = H12[0][e][b];
-            h[0][2] = H21[0][e][b];
-            h[0][3] = H22[0][e][b];
+            h11r = H11[0][e][b];
+            h12r = H12[0][e][b];
+            h21r = H21[0][e][b];
+            h22r = H22[0][e][b];
             if (!PS_BASELINE && ps->enable_ipdopd) {
             //Is this necessary? ps_04_new seems unchanged
             if ((is34 && k <= 13 && k >= 9) || (!is34 && k <= 1)) {
-                h[1][0] = -H11[1][e][b];
-                h[1][1] = -H12[1][e][b];
-                h[1][2] = -H21[1][e][b];
-                h[1][3] = -H22[1][e][b];
+                h11i = -H11[1][e][b];
+                h12i = -H12[1][e][b];
+                h21i = -H21[1][e][b];
+                h22i = -H22[1][e][b];
             } else {
-                h[1][0] = H11[1][e][b];
-                h[1][1] = H12[1][e][b];
-                h[1][2] = H21[1][e][b];
-                h[1][3] = H22[1][e][b];
+                h11i = H11[1][e][b];
+                h12i = H12[1][e][b];
+                h21i = H21[1][e][b];
+                h22i = H22[1][e][b];
             }
             }
             //Interpolation
-            h_step[0][0] = (H11[0][e+1][b] - h[0][0]) * width;
-            h_step[0][1] = (H12[0][e+1][b] - h[0][1]) * width;
-            h_step[0][2] = (H21[0][e+1][b] - h[0][2]) * width;
-            h_step[0][3] = (H22[0][e+1][b] - h[0][3]) * width;
+            h11r_step = (H11[0][e+1][b] - h11r) * width;
+            h12r_step = (H12[0][e+1][b] - h12r) * width;
+            h21r_step = (H21[0][e+1][b] - h21r) * width;
+            h22r_step = (H22[0][e+1][b] - h22r) * width;
             if (!PS_BASELINE && ps->enable_ipdopd) {
-                h_step[1][0] = (H11[1][e+1][b] - h[1][0]) * width;
-                h_step[1][1] = (H12[1][e+1][b] - h[1][1]) * width;
-                h_step[1][2] = (H21[1][e+1][b] - h[1][2]) * width;
-                h_step[1][3] = (H22[1][e+1][b] - h[1][3]) * width;
+                h11i_step = (H11[1][e+1][b] - h11i) * width;
+                h12i_step = (H12[1][e+1][b] - h12i) * width;
+                h21i_step = (H21[1][e+1][b] - h21i) * width;
+                h22i_step = (H22[1][e+1][b] - h22i) * width;
+            }
+            for (n = start + 1; n <= stop; n++) {
+                //l is s, r is d
+                float l_re = l[k][n][0];
+                float l_im = l[k][n][1];
+                float r_re = r[k][n][0];
+                float r_im = r[k][n][1];
+                h11r += h11r_step;
+                h12r += h12r_step;
+                h21r += h21r_step;
+                h22r += h22r_step;
+                if (!PS_BASELINE && ps->enable_ipdopd) {
+                    h11i += h11i_step;
+                    h12i += h12i_step;
+                    h21i += h21i_step;
+                    h22i += h22i_step;
+
+                    l[k][n][0] = h11r*l_re + h21r*r_re - h11i*l_im - h21i*r_im;
+                    l[k][n][1] = h11r*l_im + h21r*r_im + h11i*l_re + h21i*r_re;
+                    r[k][n][0] = h12r*l_re + h22r*r_re - h12i*l_im - h22i*r_im;
+                    r[k][n][1] = h12r*l_im + h22r*r_im + h12i*l_re + h22i*r_re;
+                } else {
+                    l[k][n][0] = h11r*l_re + h21r*r_re;
+                    l[k][n][1] = h11r*l_im + h21r*r_im;
+                    r[k][n][0] = h12r*l_re + h22r*r_re;
+                    r[k][n][1] = h12r*l_im + h22r*r_im;
+                }
             }
-            ps->dsp.stereo_interpolate[!PS_BASELINE && ps->enable_ipdopd](
-                l[k] + start + 1, r[k] + start + 1,
-                h, h_step, stop - start);
         }
     }
 }
 
 int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float R[2][38][64], int top)
 {
-    LOCAL_ALIGNED_16(float, Lbuf, [91], [32][2]);
-    LOCAL_ALIGNED_16(float, Rbuf, [91], [32][2]);
+    float Lbuf[91][32][2];
+    float Rbuf[91][32][2];
     const int len = 32;
     int is34 = ps->is34bands;
 
@@ -896,11 +989,11 @@ int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float
     if (top < NR_ALLPASS_BANDS[is34])
         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);
+    hybrid_analysis(Lbuf, ps->in_buf, L, is34, len);
     decorrelation(ps, Rbuf, Lbuf, is34);
     stereo_processing(ps, Lbuf, Rbuf, is34);
-    hybrid_synthesis(&ps->dsp, L, Lbuf, is34, len);
-    hybrid_synthesis(&ps->dsp, R, Rbuf, is34, len);
+    hybrid_synthesis(L, Lbuf, is34, len);
+    hybrid_synthesis(R, Rbuf, is34, len);
 
     return 0;
 }
@@ -948,5 +1041,4 @@ av_cold void ff_ps_init(void) {
 
 av_cold void ff_ps_ctx_init(PSContext *ps)
 {
-    ff_psdsp_init(&ps->dsp);
 }

libavcodec/aacps.h

@@ -24,7 +24,6 @@
 
 #include <stdint.h>
 
-#include "aacpsdsp.h"
 #include "avcodec.h"
 #include "get_bits.h"
 
@@ -61,19 +60,18 @@ typedef struct {
     int    is34bands;
     int    is34bands_old;
 
-    DECLARE_ALIGNED(16, float, in_buf)[5][44][2];
-    DECLARE_ALIGNED(16, float, delay)[PS_MAX_SSB][PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2];
-    DECLARE_ALIGNED(16, float, ap_delay)[PS_MAX_AP_BANDS][PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2];
-    DECLARE_ALIGNED(16, float, peak_decay_nrg)[34];
-    DECLARE_ALIGNED(16, float, power_smooth)[34];
-    DECLARE_ALIGNED(16, float, peak_decay_diff_smooth)[34];
-    DECLARE_ALIGNED(16, float, H11)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
-    DECLARE_ALIGNED(16, float, H12)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
-    DECLARE_ALIGNED(16, float, H21)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
-    DECLARE_ALIGNED(16, float, H22)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    float  in_buf[5][44][2];
+    float  delay[PS_MAX_SSB][PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2];
+    float  ap_delay[PS_MAX_AP_BANDS][PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2];
+    float  peak_decay_nrg[34];
+    float  power_smooth[34];
+    float  peak_decay_diff_smooth[34];
+    float  H11[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    float  H12[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    float  H21[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    float  H22[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
     int8_t opd_hist[PS_MAX_NR_IIDICC];
     int8_t ipd_hist[PS_MAX_NR_IIDICC];
-    PSDSPContext dsp;
 } PSContext;
 
 void ff_ps_init(void);

libavcodec/aacps_tablegen.c

@@ -69,23 +69,23 @@ int main(void)
     write_float_3d_array(HB, 46, 8, 4);
     printf("};\n");
 
-    printf("static const DECLARE_ALIGNED(16, float, f20_0_8)[8][8][2] = {\n");
-    write_float_3d_array(f20_0_8, 8, 8, 2);
+    printf("static const float f20_0_8[8][7][2] = {\n");
+    write_float_3d_array(f20_0_8, 8, 7, 2);
     printf("};\n");
-    printf("static const DECLARE_ALIGNED(16, float, f34_0_12)[12][8][2] = {\n");
-    write_float_3d_array(f34_0_12, 12, 8, 2);
+    printf("static const float f34_0_12[12][7][2] = {\n");
+    write_float_3d_array(f34_0_12, 12, 7, 2);
     printf("};\n");
-    printf("static const DECLARE_ALIGNED(16, float, f34_1_8)[8][8][2] = {\n");
-    write_float_3d_array(f34_1_8, 8, 8, 2);
+    printf("static const float f34_1_8[8][7][2] = {\n");
+    write_float_3d_array(f34_1_8, 8, 7, 2);
     printf("};\n");
-    printf("static const DECLARE_ALIGNED(16, float, f34_2_4)[4][8][2] = {\n");
-    write_float_3d_array(f34_2_4, 4, 8, 2);
+    printf("static const float f34_2_4[4][7][2] = {\n");
+    write_float_3d_array(f34_2_4, 4, 7, 2);
     printf("};\n");
 
-    printf("static const DECLARE_ALIGNED(16, float, Q_fract_allpass)[2][50][3][2] = {\n");
+    printf("static const 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");
+    printf("static const float phi_fract[2][50][2] = {\n");
     write_float_3d_array(phi_fract, 2, 50, 2);
     printf("};\n");
 

libavcodec/aacps_tablegen.h

@@ -31,7 +31,6 @@
 #else
 #include "libavutil/common.h"
 #include "libavutil/mathematics.h"
-#include "libavutil/mem.h"
 #define NR_ALLPASS_BANDS20 30
 #define NR_ALLPASS_BANDS34 50
 #define PS_AP_LINKS 3
@@ -39,12 +38,12 @@ static float pd_re_smooth[8*8*8];
 static float pd_im_smooth[8*8*8];
 static float HA[46][8][4];
 static float HB[46][8][4];
-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 DECLARE_ALIGNED(16, float, phi_fract)[2][50][2];
+static float f20_0_8 [ 8][7][2];
+static float f34_0_12[12][7][2];
+static float f34_1_8 [ 8][7][2];
+static float f34_2_4 [ 4][7][2];
+static float Q_fract_allpass[2][50][3][2];
+static float phi_fract[2][50][2];
 
 static const float g0_Q8[] = {
     0.00746082949812f, 0.02270420949825f, 0.04546865930473f, 0.07266113929591f,
@@ -66,7 +65,7 @@ static const float g2_Q4[] = {
      0.16486303567403f,  0.23279856662996f, 0.25f
 };
 
-static void make_filters_from_proto(float (*filter)[8][2], const float *proto, int bands)
+static void make_filters_from_proto(float (*filter)[7][2], const float *proto, int bands)
 {
     int q, n;
     for (q = 0; q < bands; q++) {

libavcodec/aacpsdsp.c

@@ -1,214 +0,0 @@
-/*
- * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
- *
- * This file is part of Libav.
- *
- * Libav is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * Libav is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "config.h"
-#include "libavutil/attributes.h"
-#include "aacpsdsp.h"
-
-static void ps_add_squares_c(float *dst, const float (*src)[2], int n)
-{
-    int i;
-    for (i = 0; i < n; i++)
-        dst[i] += src[i][0] * src[i][0] + src[i][1] * src[i][1];
-}
-
-static void ps_mul_pair_single_c(float (*dst)[2], float (*src0)[2], float *src1,
-                                 int n)
-{
-    int i;
-    for (i = 0; i < n; i++) {
-        dst[i][0] = src0[i][0] * src1[i];
-        dst[i][1] = src0[i][1] * src1[i];
-    }
-}
-
-static void ps_hybrid_analysis_c(float (*out)[2], float (*in)[2],
-                                 const float (*filter)[8][2],
-                                 int stride, int n)
-{
-    int i, j;
-
-    for (i = 0; i < n; i++) {
-        float sum_re = filter[i][6][0] * in[6][0];
-        float sum_im = filter[i][6][0] * in[6][1];
-
-        for (j = 0; j < 6; j++) {
-            float in0_re = in[j][0];
-            float in0_im = in[j][1];
-            float in1_re = in[12-j][0];
-            float in1_im = in[12-j][1];
-            sum_re += filter[i][j][0] * (in0_re + in1_re) -
-                      filter[i][j][1] * (in0_im - in1_im);
-            sum_im += filter[i][j][0] * (in0_im + in1_im) +
-                      filter[i][j][1] * (in0_re - in1_re);
-        }
-        out[i * stride][0] = sum_re;
-        out[i * stride][1] = sum_im;
-    }
-}
-
-static void ps_hybrid_analysis_ileave_c(float (*out)[32][2], float L[2][38][64],
-                                        int i, int len)
-{
-    int j;
-
-    for (; i < 64; i++) {
-        for (j = 0; j < len; j++) {
-            out[i][j][0] = L[0][j][i];
-            out[i][j][1] = L[1][j][i];
-        }
-    }
-}
-
-static void ps_hybrid_synthesis_deint_c(float out[2][38][64],
-                                        float (*in)[32][2],
-                                        int i, int len)
-{
-    int n;
-
-    for (; i < 64; i++) {
-        for (n = 0; n < len; n++) {
-            out[0][n][i] = in[i][n][0];
-            out[1][n][i] = in[i][n][1];
-        }
-    }
-}
-
-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 *transient_gain,
-                             float g_decay_slope,
-                             int len)
-{
-    static const float a[] = { 0.65143905753106f,
-                               0.56471812200776f,
-                               0.48954165955695f };
-    float ag[PS_AP_LINKS];
-    int m, n;
-
-    for (m = 0; m < PS_AP_LINKS; m++)
-        ag[m] = a[m] * g_decay_slope;
-
-    for (n = 0; n < len; n++) {
-        float in_re = delay[n][0] * phi_fract[0] - delay[n][1] * phi_fract[1];
-        float in_im = delay[n][0] * phi_fract[1] + delay[n][1] * phi_fract[0];
-        for (m = 0; m < PS_AP_LINKS; m++) {
-            float a_re                = ag[m] * in_re;
-            float a_im                = ag[m] * in_im;
-            float link_delay_re       = ap_delay[m][n+2-m][0];
-            float link_delay_im       = ap_delay[m][n+2-m][1];
-            float fractional_delay_re = Q_fract[m][0];
-            float fractional_delay_im = Q_fract[m][1];
-            float apd_re = in_re;
-            float apd_im = in_im;
-            in_re = link_delay_re * fractional_delay_re -
-                    link_delay_im * fractional_delay_im - a_re;
-            in_im = link_delay_re * fractional_delay_im +
-                    link_delay_im * fractional_delay_re - a_im;
-            ap_delay[m][n+5][0] = apd_re + ag[m] * in_re;
-            ap_delay[m][n+5][1] = apd_im + ag[m] * in_im;
-        }
-        out[n][0] = transient_gain[n] * in_re;
-        out[n][1] = transient_gain[n] * in_im;
-    }
-}
-
-static void ps_stereo_interpolate_c(float (*l)[2], float (*r)[2],
-                                    float h[2][4], float h_step[2][4],
-                                    int len)
-{
-    float h0 = h[0][0];
-    float h1 = h[0][1];
-    float h2 = h[0][2];
-    float h3 = h[0][3];
-    float hs0 = h_step[0][0];
-    float hs1 = h_step[0][1];
-    float hs2 = h_step[0][2];
-    float hs3 = h_step[0][3];
-    int n;
-
-    for (n = 0; n < len; n++) {
-        //l is s, r is d
-        float l_re = l[n][0];
-        float l_im = l[n][1];
-        float r_re = r[n][0];
-        float r_im = r[n][1];
-        h0 += hs0;
-        h1 += hs1;
-        h2 += hs2;
-        h3 += hs3;
-        l[n][0] = h0 * l_re + h2 * r_re;
-        l[n][1] = h0 * l_im + h2 * r_im;
-        r[n][0] = h1 * l_re + h3 * r_re;
-        r[n][1] = h1 * l_im + h3 * r_im;
-    }
-}
-
-static void ps_stereo_interpolate_ipdopd_c(float (*l)[2], float (*r)[2],
-                                           float h[2][4], float h_step[2][4],
-                                           int len)
-{
-    float h00  = h[0][0],      h10  = h[1][0];
-    float h01  = h[0][1],      h11  = h[1][1];
-    float h02  = h[0][2],      h12  = h[1][2];
-    float h03  = h[0][3],      h13  = h[1][3];
-    float hs00 = h_step[0][0], hs10 = h_step[1][0];
-    float hs01 = h_step[0][1], hs11 = h_step[1][1];
-    float hs02 = h_step[0][2], hs12 = h_step[1][2];
-    float hs03 = h_step[0][3], hs13 = h_step[1][3];
-    int n;
-
-    for (n = 0; n < len; n++) {
-        //l is s, r is d
-        float l_re = l[n][0];
-        float l_im = l[n][1];
-        float r_re = r[n][0];
-        float r_im = r[n][1];
-        h00 += hs00;
-        h01 += hs01;
-        h02 += hs02;
-        h03 += hs03;
-        h10 += hs10;
-        h11 += hs11;
-        h12 += hs12;
-        h13 += hs13;
-
-        l[n][0] = h00 * l_re + h02 * r_re - h10 * l_im - h12 * r_im;
-        l[n][1] = h00 * l_im + h02 * r_im + h10 * l_re + h12 * r_re;
-        r[n][0] = h01 * l_re + h03 * r_re - h11 * l_im - h13 * r_im;
-        r[n][1] = h01 * l_im + h03 * r_im + h11 * l_re + h13 * r_re;
-    }
-}
-
-av_cold void ff_psdsp_init(PSDSPContext *s)
-{
-    s->add_squares            = ps_add_squares_c;
-    s->mul_pair_single        = ps_mul_pair_single_c;
-    s->hybrid_analysis        = ps_hybrid_analysis_c;
-    s->hybrid_analysis_ileave = ps_hybrid_analysis_ileave_c;
-    s->hybrid_synthesis_deint = ps_hybrid_synthesis_deint_c;
-    s->decorrelate            = ps_decorrelate_c;
-    s->stereo_interpolate[0]  = ps_stereo_interpolate_c;
-    s->stereo_interpolate[1]  = ps_stereo_interpolate_ipdopd_c;
-
-    if (ARCH_ARM)
-        ff_psdsp_init_arm(s);
-}

libavcodec/aacpsdsp.h

@@ -1,53 +0,0 @@
-/*
- * Copyright (c) 2012 Mans Rullgard
- *
- * This file is part of Libav.
- *
- * Libav is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * Libav is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef LIBAVCODEC_AACPSDSP_H
-#define LIBAVCODEC_AACPSDSP_H
-
-#define PS_QMF_TIME_SLOTS 32
-#define PS_AP_LINKS 3
-#define PS_MAX_AP_DELAY 5
-
-typedef struct PSDSPContext {
-    void (*add_squares)(float *dst, const float (*src)[2], int n);
-    void (*mul_pair_single)(float (*dst)[2], float (*src0)[2], float *src1,
-                            int n);
-    void (*hybrid_analysis)(float (*out)[2], float (*in)[2],
-                            const float (*filter)[8][2],
-                            int stride, int n);
-    void (*hybrid_analysis_ileave)(float (*out)[32][2], float L[2][38][64],
-                                   int i, int len);
-    void (*hybrid_synthesis_deint)(float out[2][38][64], float (*in)[32][2],
-                                   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 *transient_gain,
-                        float g_decay_slope,
-                        int len);
-    void (*stereo_interpolate[2])(float (*l)[2], float (*r)[2],
-                                  float h[2][4], float h_step[2][4],
-                                  int len);
-} PSDSPContext;
-
-void ff_psdsp_init(PSDSPContext *s);
-void ff_psdsp_init_arm(PSDSPContext *s);
-
-#endif /* LIBAVCODEC_AACPSDSP_H */

libavcodec/aacpsy.c

@@ -389,8 +389,9 @@ static av_unused FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx,
     AacPsyChannel *pch  = &pctx->ch[channel];
     uint8_t grouping     = 0;
     int next_type        = pch->next_window_seq;
-    FFPsyWindowInfo wi  = { { 0 } };
+    FFPsyWindowInfo wi;
 
+    memset(&wi, 0, sizeof(wi));
     if (la) {
         float s[8], v;
         int switch_to_eight = 0;
@@ -784,8 +785,9 @@ static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio,
     int uselongblock = 1;
     int attacks[AAC_NUM_BLOCKS_SHORT + 1] = { 0 };
     int i;
-    FFPsyWindowInfo wi = { { 0 } };
+    FFPsyWindowInfo wi;
 
+    memset(&wi, 0, sizeof(wi));
     if (la) {
         float hpfsmpl[AAC_BLOCK_SIZE_LONG];
         float const *pf = hpfsmpl;

libavcodec/aacsbr.c

@@ -32,7 +32,6 @@
 #include "aacsbrdata.h"
 #include "fft.h"
 #include "aacps.h"
-#include "sbrdsp.h"
 #include "libavutil/libm.h"
 #include "libavutil/avassert.h"
 
@@ -129,24 +128,13 @@ av_cold void ff_aac_sbr_init(void)
     ff_ps_init();
 }
 
-/** Places SBR in pure upsampling mode. */
-static void sbr_turnoff(SpectralBandReplication *sbr) {
-    sbr->start = 0;
-    // Init defults used in pure upsampling mode
-    sbr->kx[1] = 32; //Typo in spec, kx' inits to 32
-    sbr->m[1] = 0;
-    // Reset values for first SBR header
-    sbr->data[0].e_a[1] = sbr->data[1].e_a[1] = -1;
-    memset(&sbr->spectrum_params, -1, sizeof(SpectrumParameters));
-}
-
 av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr)
 {
     float mdct_scale;
     if(sbr->mdct.mdct_bits)
         return;
-    sbr->kx[0] = sbr->kx[1];
-    sbr_turnoff(sbr);
+    sbr->kx[0] = sbr->kx[1] = 32; //Typo in spec, kx' inits to 32
+    sbr->data[0].e_a[1] = sbr->data[1].e_a[1] = -1;
     sbr->data[0].synthesis_filterbank_samples_offset = SBR_SYNTHESIS_BUF_SIZE - (1280 - 128);
     sbr->data[1].synthesis_filterbank_samples_offset = SBR_SYNTHESIS_BUF_SIZE - (1280 - 128);
     /* SBR requires samples to be scaled to +/-32768.0 to work correctly.
@@ -156,7 +144,6 @@ av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr)
     ff_mdct_init(&sbr->mdct,     7, 1, 1.0 / (64 * mdct_scale));
     ff_mdct_init(&sbr->mdct_ana, 7, 1, -2.0 * mdct_scale);
     ff_ps_ctx_init(&sbr->ps);
-    ff_sbrdsp_init(&sbr->dsp);
 }
 
 av_cold void ff_aac_sbr_ctx_close(SpectralBandReplication *sbr)
@@ -555,7 +542,7 @@ static int sbr_hf_calc_npatches(AACContext *ac, SpectralBandReplication *sbr)
             k = sbr->n_master;
     } while (sb != sbr->kx[1] + sbr->m[1]);
 
-    if (sbr->patch_num_subbands[sbr->num_patches-1] < 3 && sbr->num_patches > 1)
+    if (sbr->num_patches > 1 && sbr->patch_num_subbands[sbr->num_patches-1] < 3)
         sbr->num_patches--;
 
     return 0;
@@ -918,7 +905,7 @@ static void read_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
 {
     switch (bs_extension_id) {
     case EXTENSION_ID_PS:
-        if (!ac->oc[1].m4ac.ps) {
+        if (!ac->m4ac.ps) {
             av_log(ac->avctx, AV_LOG_ERROR, "Parametric Stereo signaled to be not-present but was found in the bitstream.\n");
             skip_bits_long(gb, *num_bits_left); // bs_fill_bits
             *num_bits_left = 0;
@@ -933,9 +920,7 @@ static void read_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
         }
         break;
     default:
-        // some files contain 0-padding
-        if (bs_extension_id || *num_bits_left > 16 || show_bits(gb, *num_bits_left))
-            av_log_missing_feature(ac->avctx, "Reserved SBR extensions are", 1);
+        av_log_missing_feature(ac->avctx, "Reserved SBR extensions are", 1);
         skip_bits_long(gb, *num_bits_left); // bs_fill_bits
         *num_bits_left = 0;
         break;
@@ -1011,18 +996,18 @@ static unsigned int read_sbr_data(AACContext *ac, SpectralBandReplication *sbr,
 
     if (id_aac == TYPE_SCE || id_aac == TYPE_CCE) {
         if (read_sbr_single_channel_element(ac, sbr, gb)) {
-            sbr_turnoff(sbr);
+            sbr->start = 0;
             return get_bits_count(gb) - cnt;
         }
     } else if (id_aac == TYPE_CPE) {
         if (read_sbr_channel_pair_element(ac, sbr, gb)) {
-            sbr_turnoff(sbr);
+            sbr->start = 0;
             return get_bits_count(gb) - cnt;
         }
     } else {
         av_log(ac->avctx, AV_LOG_ERROR,
             "Invalid bitstream - cannot apply SBR to element type %d\n", id_aac);
-        sbr_turnoff(sbr);
+        sbr->start = 0;
         return get_bits_count(gb) - cnt;
     }
     if (get_bits1(gb)) { // bs_extended_data
@@ -1054,7 +1039,7 @@ static void sbr_reset(AACContext *ac, SpectralBandReplication *sbr)
     if (err < 0) {
         av_log(ac->avctx, AV_LOG_ERROR,
                "SBR reset failed. Switching SBR to pure upsampling mode.\n");
-        sbr_turnoff(sbr);
+        sbr->start = 0;
     }
 }
 
@@ -1077,9 +1062,9 @@ int ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
     sbr->reset = 0;
 
     if (!sbr->sample_rate)
-        sbr->sample_rate = 2 * ac->oc[1].m4ac.sample_rate; //TODO use the nominal sample rate for arbitrary sample rate support
-    if (!ac->oc[1].m4ac.ext_sample_rate)
-        ac->oc[1].m4ac.ext_sample_rate = 2 * ac->oc[1].m4ac.sample_rate;
+        sbr->sample_rate = 2 * ac->m4ac.sample_rate; //TODO use the nominal sample rate for arbitrary sample rate support
+    if (!ac->m4ac.ext_sample_rate)
+        ac->m4ac.ext_sample_rate = 2 * ac->m4ac.sample_rate;
 
     if (crc) {
         skip_bits(gb, 10); // bs_sbr_crc_bits; TODO - implement CRC check
@@ -1089,7 +1074,6 @@ int ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
     //Save some state from the previous frame.
     sbr->kx[0] = sbr->kx[1];
     sbr->m[0] = sbr->m[1];
-    sbr->kx_and_m_pushed = 1;
 
     num_sbr_bits++;
     if (get_bits1(gb)) // bs_header_flag
@@ -1159,21 +1143,33 @@ static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
  * @param   x       pointer to the beginning of the first sample window
  * @param   W       array of complex-valued samples split into subbands
  */
-static void sbr_qmf_analysis(DSPContext *dsp, FFTContext *mdct,
-                             SBRDSPContext *sbrdsp, const float *in, float *x,
+static void sbr_qmf_analysis(DSPContext *dsp, FFTContext *mdct, const float *in, float *x,
                              float z[320], float W[2][32][32][2])
 {
-    int i;
+    int i, k;
     memcpy(W[0], W[1], sizeof(W[0]));
     memcpy(x    , x+1024, (320-32)*sizeof(x[0]));
     memcpy(x+288, in,         1024*sizeof(x[0]));
     for (i = 0; i < 32; i++) { // numTimeSlots*RATE = 16*2 as 960 sample frames
                                // are not supported
         dsp->vector_fmul_reverse(z, sbr_qmf_window_ds, x, 320);
-        sbrdsp->sum64x5(z);
-        sbrdsp->qmf_pre_shuffle(z);
+        for (k = 0; k < 64; k++) {
+            float f = z[k] + z[k + 64] + z[k + 128] + z[k + 192] + z[k + 256];
+            z[k] = f;
+        }
+        //Shuffle to IMDCT
+        z[64] = z[0];
+        for (k = 1; k < 32; k++) {
+            z[64+2*k-1] =  z[   k];
+            z[64+2*k  ] = -z[64-k];
+        }
+        z[64+63] = z[32];
+
         mdct->imdct_half(mdct, z, z+64);
-        sbrdsp->qmf_post_shuffle(W[1][i], z);
+        for (k = 0; k < 32; k++) {
+            W[1][i][k][0] = -z[63-k];
+            W[1][i][k][1] = z[k];
+        }
         x += 32;
     }
 }
@@ -1183,7 +1179,6 @@ static void sbr_qmf_analysis(DSPContext *dsp, FFTContext *mdct,
  * (14496-3 sp04 p206)
  */
 static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
-                              SBRDSPContext *sbrdsp,
                               float *out, float X[2][38][64],
                               float mdct_buf[2][64],
                               float *v0, int *v_off, const unsigned int div)
@@ -1207,12 +1202,20 @@ static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
                 X[0][i][32+n] =  X[1][i][31-n];
             }
             mdct->imdct_half(mdct, mdct_buf[0], X[0][i]);
-            sbrdsp->qmf_deint_neg(v, mdct_buf[0]);
+            for (n = 0; n < 32; n++) {
+                v[     n] =  mdct_buf[0][63 - 2*n];
+                v[63 - n] = -mdct_buf[0][62 - 2*n];
+            }
         } else {
-            sbrdsp->neg_odd_64(X[1][i]);
+            for (n = 1; n < 64; n+=2) {
+                X[1][i][n] = -X[1][i][n];
+            }
             mdct->imdct_half(mdct, mdct_buf[0], X[0][i]);
             mdct->imdct_half(mdct, mdct_buf[1], X[1][i]);
-            sbrdsp->qmf_deint_bfly(v, mdct_buf[1], mdct_buf[0]);
+            for (n = 0; n < 64; n++) {
+                v[      n] = -mdct_buf[0][63 -   n] + mdct_buf[1][  n    ];
+                v[127 - n] =  mdct_buf[0][63 -   n] + mdct_buf[1][  n    ];
+            }
         }
         dsp->vector_fmul_add(out, v                , sbr_qmf_window               , zero64, 64 >> div);
         dsp->vector_fmul_add(out, v + ( 192 >> div), sbr_qmf_window + ( 64 >> div), out   , 64 >> div);
@@ -1228,20 +1231,45 @@ static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
     }
 }
 
+static void autocorrelate(const float x[40][2], float phi[3][2][2], int lag)
+{
+    int i;
+    float real_sum = 0.0f;
+    float imag_sum = 0.0f;
+    if (lag) {
+        for (i = 1; i < 38; i++) {
+            real_sum += x[i][0] * x[i+lag][0] + x[i][1] * x[i+lag][1];
+            imag_sum += x[i][0] * x[i+lag][1] - x[i][1] * x[i+lag][0];
+        }
+        phi[2-lag][1][0] = real_sum + x[ 0][0] * x[lag][0] + x[ 0][1] * x[lag][1];
+        phi[2-lag][1][1] = imag_sum + x[ 0][0] * x[lag][1] - x[ 0][1] * x[lag][0];
+        if (lag == 1) {
+            phi[0][0][0] = real_sum + x[38][0] * x[39][0] + x[38][1] * x[39][1];
+            phi[0][0][1] = imag_sum + x[38][0] * x[39][1] - x[38][1] * x[39][0];
+        }
+    } else {
+        for (i = 1; i < 38; i++) {
+            real_sum += x[i][0] * x[i][0] + x[i][1] * x[i][1];
+        }
+        phi[2][1][0] = real_sum + x[ 0][0] * x[ 0][0] + x[ 0][1] * x[ 0][1];
+        phi[1][0][0] = real_sum + x[38][0] * x[38][0] + x[38][1] * x[38][1];
+    }
+}
+
 /** High Frequency Generation (14496-3 sp04 p214+) and Inverse Filtering
  * (14496-3 sp04 p214)
  * Warning: This routine does not seem numerically stable.
  */
-static void sbr_hf_inverse_filter(SBRDSPContext *dsp,
-                                  float (*alpha0)[2], float (*alpha1)[2],
+static void sbr_hf_inverse_filter(float (*alpha0)[2], float (*alpha1)[2],
                                   const float X_low[32][40][2], int k0)
 {
     int k;
     for (k = 0; k < k0; k++) {
-        LOCAL_ALIGNED_16(float, phi, [3], [2][2]);
-        float dk;
+        float phi[3][2][2], dk;
 
-        dsp->autocorrelate(X_low[k], phi);
+        autocorrelate(X_low[k], phi, 0);
+        autocorrelate(X_low[k], phi, 1);
+        autocorrelate(X_low[k], phi, 2);
 
         dk =  phi[2][1][0] * phi[1][0][0] -
              (phi[1][1][0] * phi[1][1][0] + phi[1][1][1] * phi[1][1][1]) / 1.000001f;
@@ -1337,11 +1365,12 @@ static int sbr_hf_gen(AACContext *ac, SpectralBandReplication *sbr,
                       const float bw_array[5], const uint8_t *t_env,
                       int bs_num_env)
 {
-    int j, x;
+    int i, j, x;
     int g = 0;
     int k = sbr->kx[1];
     for (j = 0; j < sbr->num_patches; j++) {
         for (x = 0; x < sbr->patch_num_subbands[j]; x++, k++) {
+            float alpha[4];
             const int p = sbr->patch_start_subband[j] + x;
             while (g <= sbr->n_q && k >= sbr->f_tablenoise[g])
                 g++;
@@ -1353,10 +1382,26 @@ static int sbr_hf_gen(AACContext *ac, SpectralBandReplication *sbr,
                 return -1;
             }
 
-            sbr->dsp.hf_gen(X_high[k] + ENVELOPE_ADJUSTMENT_OFFSET,
-                            X_low[p]  + ENVELOPE_ADJUSTMENT_OFFSET,
-                            alpha0[p], alpha1[p], bw_array[g],
-                            2 * t_env[0], 2 * t_env[bs_num_env]);
+            alpha[0] = alpha1[p][0] * bw_array[g] * bw_array[g];
+            alpha[1] = alpha1[p][1] * bw_array[g] * bw_array[g];
+            alpha[2] = alpha0[p][0] * bw_array[g];
+            alpha[3] = alpha0[p][1] * bw_array[g];
+
+            for (i = 2 * t_env[0]; i < 2 * t_env[bs_num_env]; i++) {
+                const int idx = i + ENVELOPE_ADJUSTMENT_OFFSET;
+                X_high[k][idx][0] =
+                    X_low[p][idx - 2][0] * alpha[0] -
+                    X_low[p][idx - 2][1] * alpha[1] +
+                    X_low[p][idx - 1][0] * alpha[2] -
+                    X_low[p][idx - 1][1] * alpha[3] +
+                    X_low[p][idx][0];
+                X_high[k][idx][1] =
+                    X_low[p][idx - 2][1] * alpha[0] +
+                    X_low[p][idx - 2][0] * alpha[1] +
+                    X_low[p][idx - 1][1] * alpha[2] +
+                    X_low[p][idx - 1][0] * alpha[3] +
+                    X_low[p][idx][1];
+            }
         }
     }
     if (k < sbr->m[1] + sbr->kx[1])
@@ -1367,8 +1412,8 @@ static int sbr_hf_gen(AACContext *ac, SpectralBandReplication *sbr,
 
 /// Generate the subband filtered lowband
 static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][38][64],
-                     const float Y0[38][64][2], const float Y1[38][64][2],
-                     const float X_low[32][40][2], int ch)
+                     const float X_low[32][40][2], const float Y[2][38][64][2],
+                     int ch)
 {
     int k, i;
     const int i_f = 32;
@@ -1382,8 +1427,8 @@ static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][38][64],
     }
     for (; k < sbr->kx[0] + sbr->m[0]; k++) {
         for (i = 0; i < i_Temp; i++) {
-            X[0][i][k] = Y0[i + i_f][k][0];
-            X[1][i][k] = Y0[i + i_f][k][1];
+            X[0][i][k] = Y[0][i + i_f][k][0];
+            X[1][i][k] = Y[0][i + i_f][k][1];
         }
     }
 
@@ -1395,8 +1440,8 @@ static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][38][64],
     }
     for (; k < sbr->kx[1] + sbr->m[1]; k++) {
         for (i = i_Temp; i < i_f; i++) {
-            X[0][i][k] = Y1[i][k][0];
-            X[1][i][k] = Y1[i][k][1];
+            X[0][i][k] = Y[1][i][k][0];
+            X[1][i][k] = Y[1][i][k][1];
         }
     }
     return 0;
@@ -1405,7 +1450,7 @@ static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][38][64],
 /** High Frequency Adjustment (14496-3 sp04 p217) and Mapping
  * (14496-3 sp04 p217)
  */
-static int sbr_mapping(AACContext *ac, SpectralBandReplication *sbr,
+static void sbr_mapping(AACContext *ac, SpectralBandReplication *sbr,
                         SBRData *ch_data, int e_a[2])
 {
     int e, i, m;
@@ -1416,12 +1461,7 @@ static int sbr_mapping(AACContext *ac, SpectralBandReplication *sbr,
         uint16_t *table = ch_data->bs_freq_res[e + 1] ? sbr->f_tablehigh : sbr->f_tablelow;
         int k;
 
-        if (sbr->kx[1] != table[0]) {
-            av_log(ac->avctx, AV_LOG_ERROR, "kx != f_table{high,low}[0]. "
-                   "Derived frequency tables were not regenerated.\n");
-            sbr_turnoff(sbr);
-            return AVERROR_BUG;
-        }
+        av_assert0(sbr->kx[1] <= table[0]);
         for (i = 0; i < ilim; i++)
             for (m = table[i]; m < table[i + 1]; m++)
                 sbr->e_origmapped[e][m - sbr->kx[1]] = ch_data->env_facs[e+1][i];
@@ -1456,15 +1496,13 @@ static int sbr_mapping(AACContext *ac, SpectralBandReplication *sbr,
     }
 
     memcpy(ch_data->s_indexmapped[0], ch_data->s_indexmapped[ch_data->bs_num_env], sizeof(ch_data->s_indexmapped[0]));
-    return 0;
 }
 
 /// Estimation of current envelope (14496-3 sp04 p218)
 static void sbr_env_estimate(float (*e_curr)[48], float X_high[64][40][2],
                              SpectralBandReplication *sbr, SBRData *ch_data)
 {
-    int e, m;
-    int kx1 = sbr->kx[1];
+    int e, i, m;
 
     if (sbr->bs_interpol_freq) {
         for (e = 0; e < ch_data->bs_num_env; e++) {
@@ -1473,7 +1511,12 @@ static void sbr_env_estimate(float (*e_curr)[48], float X_high[64][40][2],
             int iub = ch_data->t_env[e + 1] * 2 + ENVELOPE_ADJUSTMENT_OFFSET;
 
             for (m = 0; m < sbr->m[1]; m++) {
-                float sum = sbr->dsp.sum_square(X_high[m+kx1] + ilb, iub - ilb);
+                float sum = 0.0f;
+
+                for (i = ilb; i < iub; i++) {
+                    sum += X_high[m + sbr->kx[1]][i][0] * X_high[m + sbr->kx[1]][i][0] +
+                           X_high[m + sbr->kx[1]][i][1] * X_high[m + sbr->kx[1]][i][1];
+                }
                 e_curr[e][m] = sum * recip_env_size;
             }
         }
@@ -1491,11 +1534,14 @@ static void sbr_env_estimate(float (*e_curr)[48], float X_high[64][40][2],
                 const int den = env_size * (table[p + 1] - table[p]);
 
                 for (k = table[p]; k < table[p + 1]; k++) {
-                    sum += sbr->dsp.sum_square(X_high[k] + ilb, iub - ilb);
+                    for (i = ilb; i < iub; i++) {
+                        sum += X_high[k][i][0] * X_high[k][i][0] +
+                               X_high[k][i][1] * X_high[k][i][1];
+                    }
                 }
                 sum /= den;
                 for (k = table[p]; k < table[p + 1]; k++) {
-                    e_curr[e][k - kx1] = sum;
+                    e_curr[e][k - sbr->kx[1]] = sum;
                 }
             }
         }
@@ -1562,8 +1608,7 @@ static void sbr_gain_calc(AACContext *ac, SpectralBandReplication *sbr,
 }
 
 /// Assembling HF Signals (14496-3 sp04 p220)
-static void sbr_hf_assemble(float Y1[38][64][2],
-                            const float X_high[64][40][2],
+static void sbr_hf_assemble(float Y[2][38][64][2], const float X_high[64][40][2],
                             SpectralBandReplication *sbr, SBRData *ch_data,
                             const int e_a[2])
 {
@@ -1585,6 +1630,7 @@ static void sbr_hf_assemble(float Y1[38][64][2],
     float (*g_temp)[48] = ch_data->g_temp, (*q_temp)[48] = ch_data->q_temp;
     int indexnoise = ch_data->f_indexnoise;
     int indexsine  = ch_data->f_indexsine;
+    memcpy(Y[0], Y[1], sizeof(Y[0]));
 
     if (sbr->reset) {
         for (i = 0; i < h_SL; i++) {
@@ -1606,44 +1652,63 @@ static void sbr_hf_assemble(float Y1[38][64][2],
     for (e = 0; e < ch_data->bs_num_env; e++) {
         for (i = 2 * ch_data->t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
             int phi_sign = (1 - 2*(kx & 1));
-            LOCAL_ALIGNED_16(float, g_filt_tab, [48]);
-            LOCAL_ALIGNED_16(float, q_filt_tab, [48]);
-            float *g_filt, *q_filt;
 
             if (h_SL && e != e_a[0] && e != e_a[1]) {
-                g_filt = g_filt_tab;
-                q_filt = q_filt_tab;
                 for (m = 0; m < m_max; m++) {
                     const int idx1 = i + h_SL;
-                    g_filt[m] = 0.0f;
-                    q_filt[m] = 0.0f;
-                    for (j = 0; j <= h_SL; j++) {
-                        g_filt[m] += g_temp[idx1 - j][m] * h_smooth[j];
-                        q_filt[m] += q_temp[idx1 - j][m] * h_smooth[j];
-                    }
+                    float g_filt = 0.0f;
+                    for (j = 0; j <= h_SL; j++)
+                        g_filt += g_temp[idx1 - j][m] * h_smooth[j];
+                    Y[1][i][m + kx][0] =
+                        X_high[m + kx][i + ENVELOPE_ADJUSTMENT_OFFSET][0] * g_filt;
+                    Y[1][i][m + kx][1] =
+                        X_high[m + kx][i + ENVELOPE_ADJUSTMENT_OFFSET][1] * g_filt;
                 }
             } else {
-                g_filt = g_temp[i + h_SL];
-                q_filt = q_temp[i];
+                for (m = 0; m < m_max; m++) {
+                    const float g_filt = g_temp[i + h_SL][m];
+                    Y[1][i][m + kx][0] =
+                        X_high[m + kx][i + ENVELOPE_ADJUSTMENT_OFFSET][0] * g_filt;
+                    Y[1][i][m + kx][1] =
+                        X_high[m + kx][i + ENVELOPE_ADJUSTMENT_OFFSET][1] * g_filt;
+                }
             }
 
-            sbr->dsp.hf_g_filt(Y1[i] + kx, X_high + kx, g_filt, m_max,
-                               i + ENVELOPE_ADJUSTMENT_OFFSET);
-
             if (e != e_a[0] && e != e_a[1]) {
-                sbr->dsp.hf_apply_noise[indexsine](Y1[i] + kx, sbr->s_m[e],
-                                                   q_filt, indexnoise,
-                                                   kx, m_max);
-            } else {
                 for (m = 0; m < m_max; m++) {
-                    Y1[i][m + kx][0] +=
+                    indexnoise = (indexnoise + 1) & 0x1ff;
+                    if (sbr->s_m[e][m]) {
+                        Y[1][i][m + kx][0] +=
+                            sbr->s_m[e][m] * phi[0][indexsine];
+                        Y[1][i][m + kx][1] +=
+                            sbr->s_m[e][m] * (phi[1][indexsine] * phi_sign);
+                    } else {
+                        float q_filt;
+                        if (h_SL) {
+                            const int idx1 = i + h_SL;
+                            q_filt = 0.0f;
+                            for (j = 0; j <= h_SL; j++)
+                                q_filt += q_temp[idx1 - j][m] * h_smooth[j];
+                        } else {
+                            q_filt = q_temp[i][m];
+                        }
+                        Y[1][i][m + kx][0] +=
+                            q_filt * sbr_noise_table[indexnoise][0];
+                        Y[1][i][m + kx][1] +=
+                            q_filt * sbr_noise_table[indexnoise][1];
+                    }
+                    phi_sign = -phi_sign;
+                }
+            } else {
+                indexnoise = (indexnoise + m_max) & 0x1ff;
+                for (m = 0; m < m_max; m++) {
+                    Y[1][i][m + kx][0] +=
                         sbr->s_m[e][m] * phi[0][indexsine];
-                    Y1[i][m + kx][1] +=
+                    Y[1][i][m + kx][1] +=
                         sbr->s_m[e][m] * (phi[1][indexsine] * phi_sign);
                     phi_sign = -phi_sign;
                 }
             }
-            indexnoise = (indexnoise + m_max) & 0x1ff;
             indexsine = (indexsine + 1) & 3;
         }
     }
@@ -1654,54 +1719,39 @@ static void sbr_hf_assemble(float Y1[38][64][2],
 void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
                   float* L, float* R)
 {
-    int downsampled = ac->oc[1].m4ac.ext_sample_rate < sbr->sample_rate;
+    int downsampled = ac->m4ac.ext_sample_rate < sbr->sample_rate;
     int ch;
     int nch = (id_aac == TYPE_CPE) ? 2 : 1;
-    int err;
-
-    if (!sbr->kx_and_m_pushed) {
-        sbr->kx[0] = sbr->kx[1];
-        sbr->m[0] = sbr->m[1];
-    } else {
-        sbr->kx_and_m_pushed = 0;
-    }
 
     if (sbr->start) {
         sbr_dequant(sbr, id_aac);
     }
     for (ch = 0; ch < nch; ch++) {
         /* decode channel */
-        sbr_qmf_analysis(&ac->dsp, &sbr->mdct_ana, &sbr->dsp, ch ? R : L, sbr->data[ch].analysis_filterbank_samples,
+        sbr_qmf_analysis(&ac->dsp, &sbr->mdct_ana, ch ? R : L, sbr->data[ch].analysis_filterbank_samples,
                          (float*)sbr->qmf_filter_scratch,
                          sbr->data[ch].W);
         sbr_lf_gen(ac, sbr, sbr->X_low, sbr->data[ch].W);
-        sbr->data[ch].Ypos ^= 1;
         if (sbr->start) {
-            sbr_hf_inverse_filter(&sbr->dsp, sbr->alpha0, sbr->alpha1, sbr->X_low, sbr->k[0]);
+            sbr_hf_inverse_filter(sbr->alpha0, sbr->alpha1, 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->data[ch].bw_array, sbr->data[ch].t_env,
                        sbr->data[ch].bs_num_env);
 
             // hf_adj
-            err = sbr_mapping(ac, sbr, &sbr->data[ch], sbr->data[ch].e_a);
-            if (!err) {
-                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_hf_assemble(sbr->data[ch].Y[sbr->data[ch].Ypos],
-                                sbr->X_high, sbr, &sbr->data[ch],
-                                sbr->data[ch].e_a);
-            }
+            sbr_mapping(ac, sbr, &sbr->data[ch], sbr->data[ch].e_a);
+            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_hf_assemble(sbr->data[ch].Y, sbr->X_high, sbr, &sbr->data[ch],
+                            sbr->data[ch].e_a);
         }
 
         /* synthesis */
-        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);
+        sbr_x_gen(sbr, sbr->X[ch], sbr->X_low, sbr->data[ch].Y, ch);
     }
 
-    if (ac->oc[1].m4ac.ps == 1) {
+    if (ac->m4ac.ps == 1) {
         if (sbr->ps.start) {
             ff_ps_apply(ac->avctx, &sbr->ps, sbr->X[0], sbr->X[1], sbr->kx[1] + sbr->m[1]);
         } else {
@@ -1710,12 +1760,12 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
         nch = 2;
     }
 
-    sbr_qmf_synthesis(&ac->dsp, &sbr->mdct, &sbr->dsp, L, sbr->X[0], sbr->qmf_filter_scratch,
+    sbr_qmf_synthesis(&ac->dsp, &sbr->mdct, L, sbr->X[0], sbr->qmf_filter_scratch,
                       sbr->data[0].synthesis_filterbank_samples,
                       &sbr->data[0].synthesis_filterbank_samples_offset,
                       downsampled);
     if (nch == 2)
-        sbr_qmf_synthesis(&ac->dsp, &sbr->mdct, &sbr->dsp, R, sbr->X[1], sbr->qmf_filter_scratch,
+        sbr_qmf_synthesis(&ac->dsp, &sbr->mdct, R, sbr->X[1], sbr->qmf_filter_scratch,
                           sbr->data[1].synthesis_filterbank_samples,
                           &sbr->data[1].synthesis_filterbank_samples_offset,
                           downsampled);

libavcodec/aacsbrdata.h

@@ -267,8 +267,8 @@ static const int8_t sbr_offset[6][16] = {
 };
 
 ///< window coefficients for analysis/synthesis QMF banks
-static DECLARE_ALIGNED(32, float, sbr_qmf_window_ds)[320];
-static DECLARE_ALIGNED(32, float, sbr_qmf_window_us)[640] = {
+static DECLARE_ALIGNED(16, float, sbr_qmf_window_ds)[320];
+static DECLARE_ALIGNED(16, float, sbr_qmf_window_us)[640] = {
      0.0000000000, -0.0005525286, -0.0005617692, -0.0004947518,
     -0.0004875227, -0.0004893791, -0.0005040714, -0.0005226564,
     -0.0005466565, -0.0005677802, -0.0005870930, -0.0006132747,
@@ -352,8 +352,7 @@ static DECLARE_ALIGNED(32, float, sbr_qmf_window_us)[640] = {
      0.8537385600,
 };
 
-/* First two entries repeated at end to simplify SIMD implementations. */
-const DECLARE_ALIGNED(16, float, ff_sbr_noise_table)[][2] = {
+static const float sbr_noise_table[512][2] = {
 {-0.99948153278296, -0.59483417516607}, { 0.97113454393991, -0.67528515225647},
 { 0.14130051758487, -0.95090983575689}, {-0.47005496701697, -0.37340549728647},
 { 0.80705063769351,  0.29653668284408}, {-0.38981478896926,  0.89572605717087},
@@ -610,7 +609,6 @@ const DECLARE_ALIGNED(16, float, ff_sbr_noise_table)[][2] = {
 {-0.93412041758744,  0.41374052024363}, { 0.96063943315511,  0.93116709541280},
 { 0.97534253457837,  0.86150930812689}, { 0.99642466504163,  0.70190043427512},
 {-0.94705089665984, -0.29580042814306}, { 0.91599807087376, -0.98147830385781},
-{-0.99948153278296, -0.59483417516607}, { 0.97113454393991, -0.67528515225647},
 };
 
 #endif /* AVCODEC_AACSBRDATA_H */

libavcodec/aactab.c

@@ -33,8 +33,8 @@
 
 #include <stdint.h>
 
-DECLARE_ALIGNED(32, float,  ff_aac_kbd_long_1024)[1024];
-DECLARE_ALIGNED(32, float,  ff_aac_kbd_short_128)[128];
+DECLARE_ALIGNED(16, float,  ff_aac_kbd_long_1024)[1024];
+DECLARE_ALIGNED(16, float,  ff_aac_kbd_short_128)[128];
 
 const uint8_t ff_aac_num_swb_1024[] = {
     41, 41, 47, 49, 49, 51, 47, 47, 43, 43, 43, 40, 40

libavcodec/aactab.h

@@ -44,8 +44,8 @@
 /* @name window coefficients
  * @{
  */
-DECLARE_ALIGNED(32, extern float,  ff_aac_kbd_long_1024)[1024];
-DECLARE_ALIGNED(32, extern float,  ff_aac_kbd_short_128)[128];
+DECLARE_ALIGNED(16, extern float,  ff_aac_kbd_long_1024)[1024];
+DECLARE_ALIGNED(16, extern float,  ff_aac_kbd_short_128)[128];
 // @}
 
 /* @name number of scalefactor window bands for long and short transform windows respectively

libavcodec/aasc.c

@@ -43,17 +43,7 @@ static av_cold int aasc_decode_init(AVCodecContext *avctx)
     AascContext *s = avctx->priv_data;
 
     s->avctx = avctx;
-    switch (avctx->bits_per_coded_sample) {
-    case 16:
-        avctx->pix_fmt = PIX_FMT_RGB555;
-        break;
-    case 24:
-        avctx->pix_fmt = PIX_FMT_BGR24;
-        break;
-    default:
-        av_log(avctx, AV_LOG_ERROR, "Unsupported bit depth: %d\n", avctx->bits_per_coded_sample);
-        return -1;
-    }
+    avctx->pix_fmt = PIX_FMT_BGR24;
     avcodec_get_frame_defaults(&s->frame);
 
     return 0;
@@ -78,12 +68,6 @@ static int aasc_decode_frame(AVCodecContext *avctx,
     compr = AV_RL32(buf);
     buf += 4;
     buf_size -= 4;
-    switch (avctx->codec_tag) {
-    case MKTAG('A', 'A', 'S', '4'):
-        bytestream2_init(&s->gb, buf - 4, buf_size + 4);
-        ff_msrle_decode(avctx, (AVPicture*)&s->frame, 8, &s->gb);
-        break;
-    case MKTAG('A', 'A', 'S', 'C'):
     switch(compr){
     case 0:
         stride = (avctx->width * 3 + 3) & ~3;
@@ -98,18 +82,13 @@ static int aasc_decode_frame(AVCodecContext *avctx,
         }
         break;
     case 1:
-        bytestream2_init(&s->gb, buf, buf_size);
+        bytestream2_init(&s->gb, buf - 4, buf_size + 4);
         ff_msrle_decode(avctx, (AVPicture*)&s->frame, 8, &s->gb);
         break;
     default:
         av_log(avctx, AV_LOG_ERROR, "Unknown compression type %d\n", compr);
         return -1;
     }
-        break;
-    default:
-        av_log(avctx, AV_LOG_ERROR, "Unknown FourCC: %X\n", avctx->codec_tag);
-        return -1;
-    }
 
     *data_size = sizeof(AVFrame);
     *(AVFrame*)data = s->frame;
@@ -138,5 +117,5 @@ AVCodec ff_aasc_decoder = {
     .close          = aasc_decode_end,
     .decode         = aasc_decode_frame,
     .capabilities   = CODEC_CAP_DR1,
-    .long_name      = NULL_IF_CONFIG_SMALL("Autodesk RLE"),
+    .long_name = NULL_IF_CONFIG_SMALL("Autodesk RLE"),
 };

libavcodec/ac3dec.c

@@ -162,12 +162,17 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
     AC3DecodeContext *s = avctx->priv_data;
     s->avctx = avctx;
 
+#if FF_API_DRC_SCALE
+    if (avctx->drc_scale)
+        s->drc_scale = avctx->drc_scale;
+#endif
+
     ff_ac3_common_init();
     ac3_tables_init();
     ff_mdct_init(&s->imdct_256, 8, 1, 1.0);
     ff_mdct_init(&s->imdct_512, 9, 1, 1.0);
     ff_kbd_window_init(s->window, 5.0, 256);
-    ff_dsputil_init(&s->dsp, avctx);
+    dsputil_init(&s->dsp, avctx);
     ff_ac3dsp_init(&s->ac3dsp, avctx->flags & CODEC_FLAG_BITEXACT);
     ff_fmt_convert_init(&s->fmt_conv, avctx);
     av_lfg_init(&s->dith_state, 0);
@@ -753,7 +758,9 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
     int downmix_output;
     int cpl_in_use;
     GetBitContext *gbc = &s->gbc;
-    uint8_t bit_alloc_stages[AC3_MAX_CHANNELS] = { 0 };
+    uint8_t bit_alloc_stages[AC3_MAX_CHANNELS];
+
+    memset(bit_alloc_stages, 0, AC3_MAX_CHANNELS);
 
     /* block switch flags */
     different_transforms = 0;
@@ -1395,16 +1402,13 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
         if (s->out_channels < s->channels)
             s->output_mode  = s->out_channels == 1 ? AC3_CHMODE_MONO : AC3_CHMODE_STEREO;
     }
-    if (avctx->channels != s->out_channels) {
-        av_log(avctx, AV_LOG_ERROR, "channel number mismatching on damaged frame\n");
-        return AVERROR_INVALIDDATA;
-    }
     /* set audio service type based on bitstream mode for AC-3 */
     avctx->audio_service_type = s->bitstream_mode;
     if (s->bitstream_mode == 0x7 && s->channels > 1)
         avctx->audio_service_type = AV_AUDIO_SERVICE_TYPE_KARAOKE;
 
     /* get output buffer */
+    avctx->channels = s->out_channels;
     s->frame.nb_samples = s->num_blocks * 256;
     if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");

libavcodec/ac3enc.c

@@ -1383,7 +1383,8 @@ static void ac3_output_frame_header(AC3EncodeContext *s)
  */
 static void output_audio_block(AC3EncodeContext *s, int blk)
 {
-    int ch, i, baie, bnd, got_cpl, ch0;
+    int ch, i, baie, bnd, got_cpl;
+    int av_uninit(ch0);
     AC3Block *block = &s->blocks[blk];
 
     /* block switching */
@@ -2050,9 +2051,7 @@ av_cold int ff_ac3_encode_close(AVCodecContext *avctx)
 
     s->mdct_end(s);
 
-#if FF_API_OLD_ENCODE_AUDIO
     av_freep(&avctx->coded_frame);
-#endif
     return 0;
 }
 
@@ -2071,7 +2070,7 @@ static av_cold int set_channel_info(AC3EncodeContext *s, int channels,
         return AVERROR(EINVAL);
     ch_layout = *channel_layout;
     if (!ch_layout)
-        ch_layout = av_get_default_channel_layout(channels);
+        ch_layout = avcodec_guess_channel_layout(channels, CODEC_ID_AC3, NULL);
 
     s->lfe_on       = !!(ch_layout & AV_CH_LOW_FREQUENCY);
     s->channels     = channels;
@@ -2140,17 +2139,6 @@ static av_cold int validate_options(AC3EncodeContext *s)
     s->bit_alloc.sr_code  = i % 3;
     s->bitstream_id       = s->eac3 ? 16 : 8 + s->bit_alloc.sr_shift;
 
-    /* select a default bit rate if not set by the user */
-    if (!avctx->bit_rate) {
-        switch (s->fbw_channels) {
-        case 1: avctx->bit_rate =  96000; break;
-        case 2: avctx->bit_rate = 192000; break;
-        case 3: avctx->bit_rate = 320000; break;
-        case 4: avctx->bit_rate = 384000; break;
-        case 5: avctx->bit_rate = 448000; break;
-        }
-    }
-
     /* validate bit rate */
     if (s->eac3) {
         int max_br, min_br, wpf, min_br_dist, min_br_code;
@@ -2199,20 +2187,15 @@ static av_cold int validate_options(AC3EncodeContext *s)
             wpf--;
         s->frame_size_min = 2 * wpf;
     } else {
-        int best_br = 0, best_code = 0, best_diff = INT_MAX;
         for (i = 0; i < 19; i++) {
-            int br   = (ff_ac3_bitrate_tab[i] >> s->bit_alloc.sr_shift) * 1000;
-            int diff = abs(br - avctx->bit_rate);
-            if (diff < best_diff) {
-                best_br   = br;
-                best_code = i;
-                best_diff = diff;
-            }
-            if (!best_diff)
+            if ((ff_ac3_bitrate_tab[i] >> s->bit_alloc.sr_shift)*1000 == avctx->bit_rate)
                 break;
         }
-        avctx->bit_rate    = best_br;
-        s->frame_size_code = best_code << 1;
+        if (i == 19) {
+            av_log(avctx, AV_LOG_ERROR, "invalid bit rate\n");
+            return AVERROR(EINVAL);
+        }
+        s->frame_size_code = i << 1;
         s->frame_size_min  = 2 * ff_ac3_frame_size_tab[s->frame_size_code][s->bit_alloc.sr_code];
         s->num_blks_code   = 0x3;
         s->num_blocks      = 6;
@@ -2250,7 +2233,8 @@ static av_cold int validate_options(AC3EncodeContext *s)
  */
 static av_cold void set_bandwidth(AC3EncodeContext *s)
 {
-    int blk, ch, cpl_start;
+    int blk, ch;
+    int av_uninit(cpl_start);
 
     if (s->cutoff) {
         /* calculate bandwidth based on user-specified cutoff frequency */
@@ -2436,7 +2420,6 @@ av_cold int ff_ac3_encode_init(AVCodecContext *avctx)
         return ret;
 
     avctx->frame_size = AC3_BLOCK_SIZE * s->num_blocks;
-    avctx->delay      = AC3_BLOCK_SIZE;
 
     s->bitstream_mode = avctx->audio_service_type;
     if (s->bitstream_mode == AV_AUDIO_SERVICE_TYPE_KARAOKE)
@@ -2482,15 +2465,9 @@ av_cold int ff_ac3_encode_init(AVCodecContext *avctx)
     if (ret)
         goto init_fail;
 
-#if FF_API_OLD_ENCODE_AUDIO
     avctx->coded_frame= avcodec_alloc_frame();
-    if (!avctx->coded_frame) {
-        ret = AVERROR(ENOMEM);
-        goto init_fail;
-    }
-#endif
 
-    ff_dsputil_init(&s->dsp, avctx);
+    dsputil_init(&s->dsp, avctx);
     ff_ac3dsp_init(&s->ac3dsp, avctx->flags & CODEC_FLAG_BITEXACT);
 
     dprint_options(s);

libavcodec/ac3enc.h

@@ -297,9 +297,9 @@ int ff_ac3_float_mdct_init(AC3EncodeContext *s);
 int ff_ac3_fixed_allocate_sample_buffers(AC3EncodeContext *s);
 int ff_ac3_float_allocate_sample_buffers(AC3EncodeContext *s);
 
-int ff_ac3_fixed_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
-                              const AVFrame *frame, int *got_packet_ptr);
-int ff_ac3_float_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
-                              const AVFrame *frame, int *got_packet_ptr);
+int ff_ac3_fixed_encode_frame(AVCodecContext *avctx, unsigned char *frame,
+                              int buf_size, void *data);
+int ff_ac3_float_encode_frame(AVCodecContext *avctx, unsigned char *frame,
+                              int buf_size, void *data);
 
 #endif /* AVCODEC_AC3ENC_H */

libavcodec/ac3enc_fixed.c

@@ -28,7 +28,6 @@
 
 #define CONFIG_FFT_FLOAT 0
 #undef CONFIG_AC3ENC_FLOAT
-#include "internal.h"
 #include "ac3enc.h"
 #include "eac3enc.h"
 
@@ -147,17 +146,15 @@ static av_cold int ac3_fixed_encode_init(AVCodecContext *avctx)
 
 
 AVCodec ff_ac3_fixed_encoder = {
-    .name            = "ac3_fixed",
-    .type            = AVMEDIA_TYPE_AUDIO,
-    .id              = CODEC_ID_AC3,
-    .priv_data_size  = sizeof(AC3EncodeContext),
-    .init            = ac3_fixed_encode_init,
-    .encode2         = ff_ac3_fixed_encode_frame,
-    .close           = ff_ac3_encode_close,
-    .sample_fmts     = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
-                                                      AV_SAMPLE_FMT_NONE },
-    .long_name       = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
-    .priv_class      = &ac3enc_class,
+    .name           = "ac3_fixed",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = CODEC_ID_AC3,
+    .priv_data_size = sizeof(AC3EncodeContext),
+    .init           = ac3_fixed_encode_init,
+    .encode         = ff_ac3_fixed_encode_frame,
+    .close          = ff_ac3_encode_close,
+    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
+    .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
+    .priv_class = &ac3enc_class,
     .channel_layouts = ff_ac3_channel_layouts,
-    .defaults        = ac3_defaults,
 };

libavcodec/ac3enc_float.c

@@ -27,7 +27,6 @@
  */
 
 #define CONFIG_AC3ENC_FLOAT 1
-#include "internal.h"
 #include "ac3enc.h"
 #include "eac3enc.h"
 #include "kbdwin.h"
@@ -145,18 +144,16 @@ static CoefType calc_cpl_coord(CoefSumType energy_ch, CoefSumType energy_cpl)
 
 #if CONFIG_AC3_ENCODER
 AVCodec ff_ac3_encoder = {
-    .name            = "ac3",
-    .type            = AVMEDIA_TYPE_AUDIO,
-    .id              = CODEC_ID_AC3,
-    .priv_data_size  = sizeof(AC3EncodeContext),
-    .init            = ff_ac3_encode_init,
-    .encode2         = ff_ac3_float_encode_frame,
-    .close           = ff_ac3_encode_close,
-    .sample_fmts     = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
-                                                      AV_SAMPLE_FMT_NONE },
-    .long_name       = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
-    .priv_class      = &ac3enc_class,
+    .name           = "ac3",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = CODEC_ID_AC3,
+    .priv_data_size = sizeof(AC3EncodeContext),
+    .init           = ff_ac3_encode_init,
+    .encode         = ff_ac3_float_encode_frame,
+    .close          = ff_ac3_encode_close,
+    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
+    .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
+    .priv_class = &ac3enc_class,
     .channel_layouts = ff_ac3_channel_layouts,
-    .defaults        = ac3_defaults,
 };
 #endif

libavcodec/ac3enc_opts_template.c

@@ -20,7 +20,6 @@
  */
 
 #include "libavutil/opt.h"
-#include "internal.h"
 #include "ac3.h"
 
 #if AC3ENC_TYPE == AC3ENC_TYPE_AC3_FIXED
@@ -79,8 +78,3 @@ static const AVOption eac3_options[] = {
     {"auto", "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.dbl = AC3ENC_OPT_AUTO }, INT_MIN, INT_MAX, AC3ENC_PARAM, "cpl_start_band"},
 {NULL}
 };
-
-static const AVCodecDefault ac3_defaults[] = {
-    { "b",  "0" },
-    { NULL }
-};

libavcodec/ac3enc_template.c

@@ -338,7 +338,7 @@ static void compute_rematrixing_strategy(AC3EncodeContext *s)
 {
     int nb_coefs;
     int blk, bnd;
-    AC3Block *block, *block0;
+    AC3Block *block, *av_uninit(block0);
 
     if (s->channel_mode != AC3_CHMODE_STEREO)
         return;
@@ -386,11 +386,11 @@ static void compute_rematrixing_strategy(AC3EncodeContext *s)
 }
 
 
-int AC3_NAME(encode_frame)(AVCodecContext *avctx, AVPacket *avpkt,
-                           const AVFrame *frame, int *got_packet_ptr)
+int AC3_NAME(encode_frame)(AVCodecContext *avctx, unsigned char *frame,
+                           int buf_size, void *data)
 {
     AC3EncodeContext *s = avctx->priv_data;
-    const SampleType *samples = (const SampleType *)frame->data[0];
+    const SampleType *samples = data;
     int ret;
 
     if (s->options.allow_per_frame_metadata) {
@@ -437,13 +437,7 @@ int AC3_NAME(encode_frame)(AVCodecContext *avctx, AVPacket *avpkt,
 
     ff_ac3_quantize_mantissas(s);
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, s->frame_size)))
-        return ret;
-    ff_ac3_output_frame(s, avpkt->data);
+    ff_ac3_output_frame(s, frame);
 
-    if (frame->pts != AV_NOPTS_VALUE)
-        avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
-
-    *got_packet_ptr = 1;
-    return 0;
+    return s->frame_size;
 }

libavcodec/acelp_filters.c

@@ -142,3 +142,4 @@ void ff_tilt_compensation(float *mem, float tilt, float *samples, int size)
     samples[0] -= tilt * *mem;
     *mem = new_tilt_mem;
 }
+

libavcodec/acelp_pitch_delay.c

@@ -116,7 +116,7 @@ int16_t ff_acelp_decode_gain_code(
            );
 #else
     mr_energy = gain_corr_factor * exp(M_LN10 / (20 << 23) * mr_energy) /
-                sqrt(dsp->scalarproduct_int16(fc_v, fc_v, subframe_size));
+                sqrt(dsp->scalarproduct_int16(fc_v, fc_v, subframe_size, 0));
     return mr_energy >> 12;
 #endif
 }

libavcodec/adpcm.h

@@ -26,20 +26,18 @@
 #ifndef AVCODEC_ADPCM_H
 #define AVCODEC_ADPCM_H
 
-#include <stdint.h>
-
 #define BLKSIZE 1024
 
 typedef struct ADPCMChannelStatus {
     int predictor;
-    int16_t step_index;
+    short int step_index;
     int step;
     /* for encoding */
     int prev_sample;
 
     /* MS version */
-    int16_t sample1;
-    int16_t sample2;
+    short sample1;
+    short sample2;
     int coeff1;
     int coeff2;
     int idelta;

libavcodec/adpcmenc.c

@@ -24,7 +24,6 @@
 #include "bytestream.h"
 #include "adpcm.h"
 #include "adpcm_data.h"
-#include "internal.h"
 
 /**
  * @file
@@ -59,23 +58,17 @@ typedef struct ADPCMEncodeContext {
 
 #define FREEZE_INTERVAL 128
 
-static av_cold int adpcm_encode_close(AVCodecContext *avctx);
-
 static av_cold int adpcm_encode_init(AVCodecContext *avctx)
 {
     ADPCMEncodeContext *s = avctx->priv_data;
     uint8_t *extradata;
     int i;
-    int ret = AVERROR(ENOMEM);
-
-    if (avctx->channels > 2) {
-        av_log(avctx, AV_LOG_ERROR, "only stereo or mono is supported\n");
-        return AVERROR(EINVAL);
-    }
+    if (avctx->channels > 2)
+        return -1; /* only stereo or mono =) */
 
     if (avctx->trellis && (unsigned)avctx->trellis > 16U) {
         av_log(avctx, AV_LOG_ERROR, "invalid trellis size\n");
-        return AVERROR(EINVAL);
+        return -1;
     }
 
     if (avctx->trellis) {
@@ -112,12 +105,12 @@ static av_cold int adpcm_encode_init(AVCodecContext *avctx)
         /* each 16 bits sample gives one nibble
            and we have 7 bytes per channel overhead */
         avctx->frame_size = (BLKSIZE - 7 * avctx->channels) * 2 / avctx->channels + 2;
+        avctx->block_align = BLKSIZE;
         avctx->bits_per_coded_sample = 4;
-        avctx->block_align    = BLKSIZE;
-        if (!(avctx->extradata = av_malloc(32 + FF_INPUT_BUFFER_PADDING_SIZE)))
-            goto error;
         avctx->extradata_size = 32;
-        extradata = avctx->extradata;
+        extradata = avctx->extradata = av_malloc(avctx->extradata_size);
+        if (!extradata)
+            return AVERROR(ENOMEM);
         bytestream_put_le16(&extradata, avctx->frame_size);
         bytestream_put_le16(&extradata, 7); /* wNumCoef */
         for (i = 0; i < 7; i++) {
@@ -126,7 +119,7 @@ static av_cold int adpcm_encode_init(AVCodecContext *avctx)
         }
         break;
     case CODEC_ID_ADPCM_YAMAHA:
-        avctx->frame_size  = BLKSIZE * 2 / avctx->channels;
+        avctx->frame_size  = BLKSIZE * avctx->channels;
         avctx->block_align = BLKSIZE;
         break;
     case CODEC_ID_ADPCM_SWF:
@@ -135,33 +128,30 @@ static av_cold int adpcm_encode_init(AVCodecContext *avctx)
             avctx->sample_rate != 44100) {
             av_log(avctx, AV_LOG_ERROR, "Sample rate must be 11025, "
                    "22050 or 44100\n");
-            ret = AVERROR(EINVAL);
             goto error;
         }
         avctx->frame_size = 512 * (avctx->sample_rate / 11025);
         break;
     default:
-        ret = AVERROR(EINVAL);
         goto error;
     }
 
-#if FF_API_OLD_ENCODE_AUDIO
-    if (!(avctx->coded_frame = avcodec_alloc_frame()))
-        goto error;
-#endif
+    avctx->coded_frame = avcodec_alloc_frame();
+    avctx->coded_frame->key_frame= 1;
 
     return 0;
 error:
-    adpcm_encode_close(avctx);
-    return ret;
+    av_freep(&s->paths);
+    av_freep(&s->node_buf);
+    av_freep(&s->nodep_buf);
+    av_freep(&s->trellis_hash);
+    return -1;
 }
 
 static av_cold int adpcm_encode_close(AVCodecContext *avctx)
 {
     ADPCMEncodeContext *s = avctx->priv_data;
-#if FF_API_OLD_ENCODE_AUDIO
     av_freep(&avctx->coded_frame);
-#endif
     av_freep(&s->paths);
     av_freep(&s->node_buf);
     av_freep(&s->nodep_buf);
@@ -171,8 +161,8 @@ static av_cold int adpcm_encode_close(AVCodecContext *avctx)
 }
 
 
-static inline uint8_t adpcm_ima_compress_sample(ADPCMChannelStatus *c,
-                                                int16_t sample)
+static inline unsigned char adpcm_ima_compress_sample(ADPCMChannelStatus *c,
+                                                      short sample)
 {
     int delta  = sample - c->prev_sample;
     int nibble = FFMIN(7, abs(delta) * 4 /
@@ -184,8 +174,8 @@ static inline uint8_t adpcm_ima_compress_sample(ADPCMChannelStatus *c,
     return nibble;
 }
 
-static inline uint8_t adpcm_ima_qt_compress_sample(ADPCMChannelStatus *c,
-                                                   int16_t sample)
+static inline unsigned char adpcm_ima_qt_compress_sample(ADPCMChannelStatus *c,
+                                                         short sample)
 {
     int delta  = sample - c->prev_sample;
     int diff, step = ff_adpcm_step_table[c->step_index];
@@ -221,8 +211,8 @@ static inline uint8_t adpcm_ima_qt_compress_sample(ADPCMChannelStatus *c,
     return nibble;
 }
 
-static inline uint8_t adpcm_ms_compress_sample(ADPCMChannelStatus *c,
-                                               int16_t sample)
+static inline unsigned char adpcm_ms_compress_sample(ADPCMChannelStatus *c,
+                                                     short sample)
 {
     int predictor, nibble, bias;
 
@@ -238,20 +228,20 @@ static inline uint8_t adpcm_ms_compress_sample(ADPCMChannelStatus *c,
     nibble = (nibble + bias) / c->idelta;
     nibble = av_clip(nibble, -8, 7) & 0x0F;
 
-    predictor += ((nibble & 0x08) ? (nibble - 0x10) : nibble) * c->idelta;
+    predictor += (signed)((nibble & 0x08) ? (nibble - 0x10) : nibble) * c->idelta;
 
     c->sample2 = c->sample1;
     c->sample1 = av_clip_int16(predictor);
 
-    c->idelta = (ff_adpcm_AdaptationTable[nibble] * c->idelta) >> 8;
+    c->idelta = (ff_adpcm_AdaptationTable[(int)nibble] * c->idelta) >> 8;
     if (c->idelta < 16)
         c->idelta = 16;
 
     return nibble;
 }
 
-static inline uint8_t adpcm_yamaha_compress_sample(ADPCMChannelStatus *c,
-                                                   int16_t sample)
+static inline unsigned char adpcm_yamaha_compress_sample(ADPCMChannelStatus *c,
+                                                         short sample)
 {
     int nibble, delta;
 
@@ -272,9 +262,8 @@ static inline uint8_t adpcm_yamaha_compress_sample(ADPCMChannelStatus *c,
     return nibble;
 }
 
-static void adpcm_compress_trellis(AVCodecContext *avctx,
-                                   const int16_t *samples, uint8_t *dst,
-                                   ADPCMChannelStatus *c, int n)
+static void adpcm_compress_trellis(AVCodecContext *avctx, const short *samples,
+                                   uint8_t *dst, ADPCMChannelStatus *c, int n)
 {
     //FIXME 6% faster if frontier is a compile-time constant
     ADPCMEncodeContext *s = avctx->priv_data;
@@ -478,41 +467,35 @@ static void adpcm_compress_trellis(AVCodecContext *avctx,
     c->idelta     = nodes[0]->step;
 }
 
-static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
-                              const AVFrame *frame, int *got_packet_ptr)
+static int adpcm_encode_frame(AVCodecContext *avctx,
+                              unsigned char *frame, int buf_size, void *data)
 {
-    int n, i, st, pkt_size, ret;
-    const int16_t *samples;
-    uint8_t *dst;
+    int n, i, st;
+    short *samples;
+    unsigned char *dst;
     ADPCMEncodeContext *c = avctx->priv_data;
     uint8_t *buf;
 
-    samples = (const int16_t *)frame->data[0];
+    dst = frame;
+    samples = (short *)data;
     st = avctx->channels == 2;
-
-    if (avctx->codec_id == CODEC_ID_ADPCM_SWF)
-        pkt_size = (2 + avctx->channels * (22 + 4 * (frame->nb_samples - 1)) + 7) / 8;
-    else
-        pkt_size = avctx->block_align;
-    if ((ret = ff_alloc_packet2(avctx, avpkt, pkt_size)))
-        return ret;
-    dst = avpkt->data;
+    /* n = (BLKSIZE - 4 * avctx->channels) / (2 * 8 * avctx->channels); */
 
     switch(avctx->codec->id) {
     case CODEC_ID_ADPCM_IMA_WAV:
-        n = frame->nb_samples / 8;
-        c->status[0].prev_sample = samples[0];
+        n = avctx->frame_size / 8;
+        c->status[0].prev_sample = (signed short)samples[0]; /* XXX */
         /* c->status[0].step_index = 0;
         XXX: not sure how to init the state machine */
         bytestream_put_le16(&dst, c->status[0].prev_sample);
-        *dst++ = c->status[0].step_index;
+        *dst++ = (unsigned char)c->status[0].step_index;
         *dst++ = 0; /* unknown */
         samples++;
         if (avctx->channels == 2) {
-            c->status[1].prev_sample = samples[0];
+            c->status[1].prev_sample = (signed short)samples[0];
             /* c->status[1].step_index = 0; */
             bytestream_put_le16(&dst, c->status[1].prev_sample);
-            *dst++ = c->status[1].step_index;
+            *dst++ = (unsigned char)c->status[1].step_index;
             *dst++ = 0;
             samples++;
         }
@@ -568,10 +551,10 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     {
         int ch, i;
         PutBitContext pb;
-        init_put_bits(&pb, dst, pkt_size * 8);
+        init_put_bits(&pb, dst, buf_size * 8);
 
         for (ch = 0; ch < avctx->channels; ch++) {
-            put_bits(&pb, 9, (c->status[ch].prev_sample & 0xFFFF) >> 7);
+            put_bits(&pb, 9, (c->status[ch].prev_sample + 0x10000) >> 7);
             put_bits(&pb, 7,  c->status[ch].step_index);
             if (avctx->trellis > 0) {
                 uint8_t buf[64];
@@ -592,15 +575,16 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         }
 
         flush_put_bits(&pb);
+        dst += put_bits_count(&pb) >> 3;
         break;
     }
     case CODEC_ID_ADPCM_SWF:
     {
         int i;
         PutBitContext pb;
-        init_put_bits(&pb, dst, pkt_size * 8);
+        init_put_bits(&pb, dst, buf_size * 8);
 
-        n = frame->nb_samples - 1;
+        n = avctx->frame_size - 1;
 
         // store AdpcmCodeSize
         put_bits(&pb, 2, 2);    // set 4-bit flash adpcm format
@@ -611,7 +595,7 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
             c->status[i].step_index = av_clip(c->status[i].step_index, 0, 63);
             put_sbits(&pb, 16, samples[i]);
             put_bits(&pb, 6, c->status[i].step_index);
-            c->status[i].prev_sample = samples[i];
+            c->status[i].prev_sample = (signed short)samples[i];
         }
 
         if (avctx->trellis > 0) {
@@ -627,7 +611,7 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
             }
             av_free(buf);
         } else {
-            for (i = 1; i < frame->nb_samples; i++) {
+            for (i = 1; i < avctx->frame_size; i++) {
                 put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0],
                          samples[avctx->channels * i]));
                 if (avctx->channels == 2)
@@ -636,6 +620,7 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
             }
         }
         flush_put_bits(&pb);
+        dst += put_bits_count(&pb) >> 3;
         break;
     }
     case CODEC_ID_ADPCM_MS:
@@ -683,7 +668,7 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         }
         break;
     case CODEC_ID_ADPCM_YAMAHA:
-        n = frame->nb_samples / 2;
+        n = avctx->frame_size / 2;
         if (avctx->trellis > 0) {
             FF_ALLOC_OR_GOTO(avctx, buf, 2 * n * 2, error);
             n *= 2;
@@ -707,14 +692,10 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
             }
         break;
     default:
-        return AVERROR(EINVAL);
+    error:
+        return -1;
     }
-
-    avpkt->size = pkt_size;
-    *got_packet_ptr = 1;
-    return 0;
-error:
-    return AVERROR(ENOMEM);
+    return dst - frame;
 }
 
 
@@ -725,10 +706,10 @@ AVCodec ff_ ## name_ ## _encoder = {                        \
     .id             = id_,                                  \
     .priv_data_size = sizeof(ADPCMEncodeContext),           \
     .init           = adpcm_encode_init,                    \
-    .encode2        = adpcm_encode_frame,                   \
+    .encode         = adpcm_encode_frame,                   \
     .close          = adpcm_encode_close,                   \
-    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,    \
-                                                      AV_SAMPLE_FMT_NONE }, \
+    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,   \
+                                                      AV_SAMPLE_FMT_NONE}, \
     .long_name      = NULL_IF_CONFIG_SMALL(long_name_),     \
 }
 

libavcodec/adx_parser.c

@@ -80,9 +80,6 @@ static int adx_parse(AVCodecParserContext *s1,
         *poutbuf_size = 0;
         return buf_size;
     }
-
-    s1->duration = BLOCK_SAMPLES;
-
     *poutbuf = buf;
     *poutbuf_size = buf_size;
     return next;

libavcodec/adxdec.c

@@ -100,7 +100,6 @@ static int adx_decode_frame(AVCodecContext *avctx, void *data,
     ADXContext *c       = avctx->priv_data;
     int16_t *samples;
     const uint8_t *buf  = avpkt->data;
-    const uint8_t *buf_end = buf + avpkt->size;
     int num_blocks, ch, ret;
 
     if (c->eof) {
@@ -149,7 +148,7 @@ static int adx_decode_frame(AVCodecContext *avctx, void *data,
 
     while (num_blocks--) {
         for (ch = 0; ch < c->channels; ch++) {
-            if (buf_end - buf < BLOCK_SIZE || adx_decode(c, samples + ch, buf, ch)) {
+            if (adx_decode(c, samples + ch, buf, ch)) {
                 c->eof = 1;
                 buf = avpkt->data + avpkt->size;
                 break;

libavcodec/adxenc.c

@@ -22,7 +22,6 @@
 #include "avcodec.h"
 #include "adx.h"
 #include "bytestream.h"
-#include "internal.h"
 #include "put_bits.h"
 
 /**
@@ -88,6 +87,9 @@ static int adx_encode_header(AVCodecContext *avctx, uint8_t *buf, int bufsize)
 {
     ADXContext *c = avctx->priv_data;
 
+    if (bufsize < HEADER_SIZE)
+        return AVERROR(EINVAL);
+
     bytestream_put_be16(&buf, 0x8000);              /* header signature */
     bytestream_put_be16(&buf, HEADER_SIZE - 4);     /* copyright offset */
     bytestream_put_byte(&buf, 3);                   /* encoding */
@@ -117,10 +119,7 @@ static av_cold int adx_encode_init(AVCodecContext *avctx)
     }
     avctx->frame_size = BLOCK_SAMPLES;
 
-#if FF_API_OLD_ENCODE_AUDIO
-    avcodec_get_frame_defaults(&c->frame);
-    avctx->coded_frame = &c->frame;
-#endif
+    avctx->coded_frame = avcodec_alloc_frame();
 
     /* the cutoff can be adjusted, but this seems to work pretty well */
     c->cutoff = 500;
@@ -129,36 +128,40 @@ static av_cold int adx_encode_init(AVCodecContext *avctx)
     return 0;
 }
 
-static int adx_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
-                            const AVFrame *frame, int *got_packet_ptr)
+static av_cold int adx_encode_close(AVCodecContext *avctx)
+{
+    av_freep(&avctx->coded_frame);
+    return 0;
+}
+
+static int adx_encode_frame(AVCodecContext *avctx, uint8_t *frame,
+                            int buf_size, void *data)
 {
     ADXContext *c          = avctx->priv_data;
-    const int16_t *samples = (const int16_t *)frame->data[0];
-    uint8_t *dst;
-    int ch, out_size, ret;
-
-    out_size = BLOCK_SIZE * avctx->channels + !c->header_parsed * HEADER_SIZE;
-    if ((ret = ff_alloc_packet2(avctx, avpkt, out_size)) < 0)
-        return ret;
-    dst = avpkt->data;
+    const int16_t *samples = data;
+    uint8_t *dst           = frame;
+    int ch;
 
     if (!c->header_parsed) {
         int hdrsize;
-        if ((hdrsize = adx_encode_header(avctx, dst, avpkt->size)) < 0) {
+        if ((hdrsize = adx_encode_header(avctx, dst, buf_size)) < 0) {
             av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
             return AVERROR(EINVAL);
         }
         dst      += hdrsize;
+        buf_size -= hdrsize;
         c->header_parsed = 1;
     }
+    if (buf_size < BLOCK_SIZE * avctx->channels) {
+        av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
+        return AVERROR(EINVAL);
+    }
 
     for (ch = 0; ch < avctx->channels; ch++) {
         adx_encode(c, dst, samples + ch, &c->prev[ch], avctx->channels);
         dst += BLOCK_SIZE;
     }
-
-    *got_packet_ptr = 1;
-    return 0;
+    return dst - frame;
 }
 
 AVCodec ff_adpcm_adx_encoder = {
@@ -167,7 +170,8 @@ AVCodec ff_adpcm_adx_encoder = {
     .id             = CODEC_ID_ADPCM_ADX,
     .priv_data_size = sizeof(ADXContext),
     .init           = adx_encode_init,
-    .encode2        = adx_encode_frame,
+    .encode         = adx_encode_frame,
+    .close          = adx_encode_close,
     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
                                                       AV_SAMPLE_FMT_NONE },
     .long_name      = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),

libavcodec/alac.c

@@ -602,30 +602,29 @@ buf_alloc_fail:
 
 static int alac_set_info(ALACContext *alac)
 {
-    GetByteContext gb;
+    const unsigned char *ptr = alac->avctx->extradata;
 
-    bytestream2_init(&gb, alac->avctx->extradata,
-                     alac->avctx->extradata_size);
+    ptr += 4; /* size */
+    ptr += 4; /* alac */
+    ptr += 4; /* version */
 
-    bytestream2_skipu(&gb, 12); // size:4, alac:4, version:4
+    if(AV_RB32(ptr) >= UINT_MAX/4){
+        av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n");
+        return -1;
+    }
 
     /* buffer size / 2 ? */
-    alac->setinfo_max_samples_per_frame = bytestream2_get_be32u(&gb);
-    if (alac->setinfo_max_samples_per_frame >= UINT_MAX/4){
-        av_log(alac->avctx, AV_LOG_ERROR,
-               "setinfo_max_samples_per_frame too large\n");
-        return AVERROR_INVALIDDATA;
-    }
-    bytestream2_skipu(&gb, 1);  // compatible version
-    alac->setinfo_sample_size           = bytestream2_get_byteu(&gb);
-    alac->setinfo_rice_historymult      = bytestream2_get_byteu(&gb);
-    alac->setinfo_rice_initialhistory   = bytestream2_get_byteu(&gb);
-    alac->setinfo_rice_kmodifier        = bytestream2_get_byteu(&gb);
-    alac->numchannels                   = bytestream2_get_byteu(&gb);
-    bytestream2_get_be16u(&gb); // maxRun
-    bytestream2_get_be32u(&gb); // max coded frame size
-    bytestream2_get_be32u(&gb); // average bitrate
-    bytestream2_get_be32u(&gb); // samplerate
+    alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr);
+    ptr++;                          /* compatible version */
+    alac->setinfo_sample_size           = *ptr++;
+    alac->setinfo_rice_historymult      = *ptr++;
+    alac->setinfo_rice_initialhistory   = *ptr++;
+    alac->setinfo_rice_kmodifier        = *ptr++;
+    alac->numchannels                   = *ptr++;
+    bytestream_get_be16(&ptr);      /* maxRun */
+    bytestream_get_be32(&ptr);      /* max coded frame size */
+    bytestream_get_be32(&ptr);      /* average bitrate */
+    bytestream_get_be32(&ptr);      /* samplerate */
 
     return 0;
 }
@@ -637,10 +636,9 @@ static av_cold int alac_decode_init(AVCodecContext * avctx)
     alac->avctx = avctx;
 
     /* initialize from the extradata */
-    if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) {
-        av_log(avctx, AV_LOG_ERROR, "alac: expected %d extradata bytes\n",
-            ALAC_EXTRADATA_SIZE);
-        return -1;
+    if (alac->avctx->extradata_size < ALAC_EXTRADATA_SIZE) {
+        av_log(avctx, AV_LOG_ERROR, "alac: extradata is too small\n");
+        return AVERROR_INVALIDDATA;
     }
     if (alac_set_info(alac)) {
         av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n");
@@ -693,5 +691,5 @@ AVCodec ff_alac_decoder = {
     .close          = alac_decode_close,
     .decode         = alac_decode_frame,
     .capabilities   = CODEC_CAP_DR1,
-    .long_name      = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
+    .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
 };