Update for FFmpeg 1.2.7

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

.gitignore

@@ -37,6 +37,7 @@
 /doc/examples/filtering_video
 /doc/examples/metadata
 /doc/examples/muxing
+/doc/examples/pc-uninstalled
 /doc/examples/resampling_audio
 /doc/examples/scaling_video
 /doc/fate.txt
@@ -66,3 +67,4 @@
 /tools/probetest
 /tools/qt-faststart
 /tools/trasher
+/tools/seek_print

CREDITS

@@ -1,55 +1,6 @@
-This file contains the names of some of the people who have contributed to
-FFmpeg. The names are sorted alphabetically by last name.  As this file is
-currently quite outdated and git serves as a much better tool for determining
-authorship, it remains here for historical reasons only.
+See the Git history of the project (git://source.ffmpeg.org/ffmpeg) to
+get the names of people who have contributed to FFmpeg.
 
-Dénes Balatoni
-Michel Bardiaux
-Fabrice Bellard
-Patrice Bensoussan
-Alex Beregszaszi
-BERO
-Thilo Borgmann
-Mario Brito
-Ronald Bultje
-Alex Converse
-Maarten Daniels
-Reimar Doeffinger
-Tim Ferguson
-Brian Foley
-Arpad Gereoffy
-Philip Gladstone
-Vladimir Gneushev
-Roine Gustafsson
-David Hammerton
-Wolfgang Hesseler
-Marc Hoffman
-Falk Hueffner
-Aurélien Jacobs
-Steven Johnson
-Zdenek Kabelac
-Robin Kay
-Todd Kirby
-Nick Kurshev
-Benjamin Larsson
-Loïc Le Loarer
-Daniel Maas
-Mike Melanson
-Loren Merritt
-Jeff Muizelaar
-Michael Niedermayer
-François Revol
-Peter Ross
-Måns Rullgård
-Stefano Sabatini
-Roman Shaposhnik
-Oded Shimon
-Dieter Shirley
-Konstantin Shishkov
-Juan J. Sierralta
-Ewald Snel
-Sascha Sommer
-Leon van Stuivenberg
-Roberto Togni
-Lionel Ulmer
-Reynaldo Verdejo
+To check the log, you can type the command "git log" in the FFmpeg
+source directory, or browse the online repository at
+http://source.ffmpeg.org.

Changelog

@@ -1,7 +1,29 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.
 
-version <next>:
+version 1.2:
+
+- VDPAU hardware acceleration through normal hwaccel
+- SRTP support
+- Error diffusion dither in Swscale
+- Chained Ogg support
+- Theora Midstream reconfiguration support
+- EVRC decoder
+- audio fade filter
+- filtering audio with unknown channel layout
+- allpass, bass, bandpass, bandreject, biquad, equalizer, highpass, lowpass
+  and treble audio filter
+- improved showspectrum filter, with multichannel support and sox-like colors
+- histogram filter
+- tee muxer
+- il filter ported from libmpcodecs
+- support ID3v2 tags in ASF files
+- encrypted TTA stream decoding support
+- RF64 support in WAV muxer
+- noise filter ported from libmpcodecs
+- Subtitles character encoding conversion
+- blend filter
+- stereo3d filter ported from libmpcodecs
 
 
 version 1.1:
@@ -60,7 +82,9 @@ version 1.1:
 - support building on the Plan 9 operating system
 - kerndeint filter ported from MPlayer
 - histeq filter ported from VirtualDub
+- Megalux Frame demuxer
 - 012v decoder
+- Improved AVC Intra decoding support
 
 
 version 1.0:
@@ -349,7 +373,6 @@ easier to use. The changes are:
 - Simple segmenting muxer
 - Indeo 4 decoder
 - SMJPEG demuxer
-- Megalux Frame demuxer
 
 
 version 0.8:

LICENSE

@@ -36,13 +36,20 @@ Specifically, the GPL parts of FFmpeg are
     - vf_hue.c
     - vf_kerndeint.c
     - vf_mp.c
+    - vf_noise.c
     - vf_pp.c
     - vf_smartblur.c
+    - vf_stereo3d.c
     - vf_super2xsai.c
     - vf_tinterlace.c
     - vf_yadif.c
     - vsrc_mptestsrc.c
 
+Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
+the configure parameter --enable-version3 will activate this licensing option
+for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
+COPYING.GPLv3 to learn the exact legal terms that apply in this case.
+
 There are a handful of files under other licensing terms, namely:
 
 * The files libavcodec/jfdctfst.c, libavcodec/jfdctint_template.c and
@@ -52,11 +59,6 @@ There are a handful of files under other licensing terms, namely:
   You must also indicate any changes including additions and deletions to
   those three files in the documentation.
 
-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
-for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
-COPYING.GPLv3 to learn the exact legal terms that apply in this case.
-
 
 external libraries
 ==================

MAINTAINERS

@@ -445,9 +445,9 @@ x86                                     Michael Niedermayer
 Releases
 ========
 
+1.2                                     Michael Niedermayer
 1.1                                     Michael Niedermayer
 1.0                                     Michael Niedermayer
-0.11                                    Michael Niedermayer
 
 
 

RELEASE

@@ -1 +1 @@
-1.1.git
+1.2.7

VERSION

@@ -0,0 +1 @@
+1.2.7

cmdutils.c

@@ -62,10 +62,10 @@
 static int init_report(const char *env);
 
 struct SwsContext *sws_opts;
-SwrContext *swr_opts;
-AVDictionary *format_opts, *codec_opts;
+AVDictionary *swr_opts;
+AVDictionary *format_opts, *codec_opts, *resample_opts;
 
-const int this_year = 2013;
+const int this_year = 2014;
 
 static FILE *report_file;
 
@@ -75,9 +75,6 @@ void init_opts(void)
     if(CONFIG_SWSCALE)
         sws_opts = sws_getContext(16, 16, 0, 16, 16, 0, SWS_BICUBIC,
                               NULL, NULL, NULL);
-
-    if(CONFIG_SWRESAMPLE)
-        swr_opts = swr_alloc();
 }
 
 void uninit_opts(void)
@@ -87,11 +84,10 @@ void uninit_opts(void)
     sws_opts = NULL;
 #endif
 
-    if(CONFIG_SWRESAMPLE)
-        swr_free(&swr_opts);
-
+    av_dict_free(&swr_opts);
     av_dict_free(&format_opts);
     av_dict_free(&codec_opts);
+    av_dict_free(&resample_opts);
 }
 
 void log_callback_help(void *ptr, int level, const char *fmt, va_list vl)
@@ -478,6 +474,9 @@ int opt_default(void *optctx, const char *opt, const char *arg)
     char opt_stripped[128];
     const char *p;
     const AVClass *cc = avcodec_get_class(), *fc = avformat_get_class();
+#if CONFIG_AVRESAMPLE
+    const AVClass *rc = avresample_get_class();
+#endif
     const AVClass *sc, *swr_class;
 
     if (!strcmp(opt, "debug") || !strcmp(opt, "fdebug"))
@@ -516,13 +515,23 @@ int opt_default(void *optctx, const char *opt, const char *arg)
 #endif
 #if CONFIG_SWRESAMPLE
     swr_class = swr_get_class();
-    if (!consumed && 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 (!consumed && (o=av_opt_find(&swr_class, opt, NULL, 0,
+                                    AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
+        struct SwrContext *swr = swr_alloc();
+        int ret = av_opt_set(swr, opt, arg, 0);
+        swr_free(&swr);
         if (ret < 0) {
             av_log(NULL, AV_LOG_ERROR, "Error setting option %s.\n", opt);
             return ret;
         }
+        av_dict_set(&swr_opts, opt, arg, FLAGS);
+        consumed = 1;
+    }
+#endif
+#if CONFIG_AVRESAMPLE
+    if ((o=av_opt_find(&rc, opt, NULL, 0,
+                       AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
+        av_dict_set(&resample_opts, opt, arg, FLAGS);
         consumed = 1;
     }
 #endif
@@ -575,9 +584,11 @@ static void finish_group(OptionParseContext *octx, int group_idx,
     g->swr_opts    = swr_opts;
     g->codec_opts  = codec_opts;
     g->format_opts = format_opts;
+    g->resample_opts = resample_opts;
 
     codec_opts  = NULL;
     format_opts = NULL;
+    resample_opts = NULL;
 #if CONFIG_SWSCALE
     sws_opts    = NULL;
 #endif
@@ -635,11 +646,11 @@ void uninit_parse_context(OptionParseContext *octx)
             av_freep(&l->groups[j].opts);
             av_dict_free(&l->groups[j].codec_opts);
             av_dict_free(&l->groups[j].format_opts);
+            av_dict_free(&l->groups[j].resample_opts);
 #if CONFIG_SWSCALE
             sws_freeContext(l->groups[j].sws_opts);
 #endif
-            if(CONFIG_SWRESAMPLE)
-                swr_free(&l->groups[j].swr_opts);
+            av_dict_free(&l->groups[j].swr_opts);
         }
         av_freep(&l->groups);
     }
@@ -656,6 +667,7 @@ int split_commandline(OptionParseContext *octx, int argc, char *argv[],
                       const OptionGroupDef *groups, int nb_groups)
 {
     int optindex = 1;
+    int dashdash = -2;
 
     /* perform system-dependent conversions for arguments list */
     prepare_app_arguments(&argc, &argv);
@@ -670,8 +682,12 @@ int split_commandline(OptionParseContext *octx, int argc, char *argv[],
 
         av_log(NULL, AV_LOG_DEBUG, "Reading option '%s' ...", opt);
 
+        if (opt[0] == '-' && opt[1] == '-' && !opt[2]) {
+            dashdash = optindex;
+            continue;
+        }
         /* unnamed group separators, e.g. output filename */
-        if (opt[0] != '-' || !opt[1]) {
+        if (opt[0] != '-' || !opt[1] || dashdash+1 == optindex) {
             finish_group(octx, 0, opt);
             av_log(NULL, AV_LOG_DEBUG, " matched as %s.\n", groups[0].name);
             continue;
@@ -743,7 +759,7 @@ do {                                                                           \
         return AVERROR_OPTION_NOT_FOUND;
     }
 
-    if (octx->cur_group.nb_opts || codec_opts || format_opts)
+    if (octx->cur_group.nb_opts || codec_opts || format_opts || resample_opts)
         av_log(NULL, AV_LOG_WARNING, "Trailing options were found on the "
                "commandline.\n");
 
@@ -1612,7 +1628,7 @@ int show_help(void *optctx, const char *opt, const char *arg)
 int read_yesno(void)
 {
     int c = getchar();
-    int yesno = (toupper(c) == 'Y');
+    int yesno = (av_toupper(c) == 'Y');
 
     while (c != '\n' && c != EOF)
         c = getchar();
@@ -1733,10 +1749,8 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
     if (!codec)
         codec            = s->oformat ? avcodec_find_encoder(codec_id)
                                       : avcodec_find_decoder(codec_id);
-    if (!codec)
-        return NULL;
 
-    switch (codec->type) {
+    switch (st->codec->codec_type) {
     case AVMEDIA_TYPE_VIDEO:
         prefix  = 'v';
         flags  |= AV_OPT_FLAG_VIDEO_PARAM;
@@ -1848,12 +1862,6 @@ static int alloc_buffer(FrameBuffer **pool, AVCodecContext *s, FrameBuffer **pbu
         av_log(s, AV_LOG_ERROR, "alloc_buffer: av_image_alloc() failed\n");
         return ret;
     }
-    /* XXX this shouldn't be needed, but some tests break without this line
-     * those decoders are buggy and need to be fixed.
-     * the following tests fail:
-     * cdgraphics, ansi, aasc, fraps-v1, qtrle-1bit
-     */
-    memset(buf->base[0], 128, ret);
 
     avcodec_get_chroma_sub_sample(s->pix_fmt, &h_chroma_shift, &v_chroma_shift);
     for (i = 0; i < FF_ARRAY_ELEMS(buf->data); i++) {

cmdutils.h

@@ -51,8 +51,8 @@ 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;
+extern AVDictionary *swr_opts;
+extern AVDictionary *format_opts, *codec_opts, *resample_opts;
 
 /**
  * Initialize the cmdutils option system, in particular
@@ -123,7 +123,7 @@ double parse_number_or_die(const char *context, const char *numstr, int type,
  * not zero timestr is interpreted as a duration, otherwise as a
  * date
  *
- * @see parse_date()
+ * @see av_parse_time()
  */
 int64_t parse_time_or_die(const char *context, const char *timestr,
                           int is_duration);
@@ -190,13 +190,13 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags,
 void show_help_children(const AVClass *class, int flags);
 
 /**
- * Per-avtool specific help handler. Implemented in each
- * avtool, called by show_help().
+ * Per-fftool specific help handler. Implemented in each
+ * fftool, called by show_help().
  */
 void show_help_default(const char *opt, const char *arg);
 
 /**
- * Generic -h handler common to all avtools.
+ * Generic -h handler common to all fftools.
  */
 int show_help(void *optctx, const char *opt, const char *arg);
 
@@ -253,8 +253,9 @@ typedef struct OptionGroup {
 
     AVDictionary *codec_opts;
     AVDictionary *format_opts;
+    AVDictionary *resample_opts;
     struct SwsContext *sws_opts;
-    struct SwrContext *swr_opts;
+    AVDictionary *swr_opts;
 } OptionGroup;
 
 /**

common.mak

@@ -32,7 +32,7 @@ ASFLAGS    := $(CPPFLAGS) $(ASFLAGS)
 CXXFLAGS   += $(CPPFLAGS) $(CFLAGS)
 YASMFLAGS  += $(IFLAGS:%=%/) -Pconfig.asm
 
-HOSTCCFLAGS = $(IFLAGS) $(HOSTCFLAGS)
+HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
 LDFLAGS    := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)
 
 define COMPILE
@@ -117,12 +117,12 @@ $(HOSTPROGS): %$(HOSTEXESUF): %.o
 	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $< $(HOSTLIBS)
 
 $(OBJS):     | $(sort $(dir $(OBJS)))
+$(HOBJS):    | $(sort $(dir $(HOBJS)))
 $(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
 $(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
-$(HOBJS):    | $(sort $(dir $(HOBJS)))
 $(TOOLOBJS): | tools
 
-OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOSTOBJS) $(TESTOBJS) $(HOBJS))
+OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(TESTOBJS))
 
 CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
 DISTCLEANSUFFIXES = *.pc

compat/strtod.c

@@ -19,7 +19,6 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include <ctype.h>
 #include <limits.h>
 #include <stdlib.h>
 
@@ -49,7 +48,7 @@ double avpriv_strtod(const char *nptr, char **endptr)
     double res;
 
     /* Skip leading spaces */
-    while (isspace(*nptr))
+    while (av_isspace(*nptr))
         nptr++;
 
     if (!av_strncasecmp(nptr, "infinity", 8)) {

doc/APIchanges

@@ -15,7 +15,19 @@ libavutil:     2012-10-22
 
 API changes, most recent first:
 
-2012-12-20 - xxxxxxx - lavfi 3.28.100 - avfilter.h
+2013-03-07 - xxxxxx - lavu 52.18.100 - avstring.h,bprint.h
+  Add av_escape() and av_bprint_escape() API.
+
+2013-02-24 - xxxxxx - lavfi 3.41.100 - buffersink.h
+  Add sample_rates field to AVABufferSinkParams.
+
+2013-01-17 - a1a707f - lavf 54.61.100
+  Add av_codec_get_tag2().
+
+2013-01-01 - 2eb2e17 - lavfi 3.34.100
+  Add avfilter_get_audio_buffer_ref_from_arrays_channels.
+
+2012-12-20 - 34de47aa - lavfi 3.29.100 - avfilter.h
   Add AVFilterLink.channels, avfilter_link_get_channels()
   and avfilter_ref_get_channels().
 
@@ -132,30 +144,54 @@ API changes, most recent first:
 2012-03-26 - a67d9cf - lavfi 2.66.100
   Add avfilter_fill_frame_from_{audio_,}buffer_ref() functions.
 
-2012-xx-xx - xxxxxxx - lavu 52.2.1 - avstring.h
+2013-xx-xx - xxxxxxx - lavfi 3.4.0 - avfiltergraph.h
+  Add resample_lavr_opts to AVFilterGraph for setting libavresample options
+  for auto-inserted resample filters.
+
+2013-xx-xx - xxxxxxx - lavu 52.7.0 - dict.h
+  Add av_dict_parse_string() to set multiple key/value pairs at once from a
+  string.
+
+2013-01-xx - xxxxxxx - lavu 52.6.0 - avstring.h
+  Add av_strnstr()
+
+2013-01-xx - xxxxxxx - lavu 52.5.0 - hmac.h
+  Add AVHMAC.
+
+2013-01-13 - xxxxxxx - lavc 54.87.100 / 54.36.0 - vdpau.h
+  Add AVVDPAUContext struct for VDPAU hardware-accelerated decoding.
+
+2013-01-12 - dae382b / 169fb94 - lavu 52.14.100 / 52.4.0 - pixdesc.h
+  Add AV_PIX_FMT_VDPAU flag.
+
+2013-01-07 - 249fca3 / 074a00d - lavr 1.1.0
+  Add avresample_set_channel_mapping() for input channel reordering,
+  duplication, and silencing.
+
+2012-12-29 - 2ce43b3 / d8fd06c - lavu 52.13.100 / 52.3.0 - avstring.h
   Add av_basename() and av_dirname().
 
-2012-11-10 - 5980f5dd - lavu 52.2.0 - audioconvert.h
+2012-11-11 - 03b0787 / 5980f5d - lavu 52.6.100 / 52.2.0 - audioconvert.h
   Rename audioconvert.h to channel_layout.h. audioconvert.h is now deprecated.
 
-2012-10-26 - dfde8a34 - lavu 52.1.0 - intmath.h
-  Add av_ctz() for trailing zero bit count.
+2012-11-05 - 7d26be6 / dfde8a3 - lavu 52.5.100 / 52.1.0 - intmath.h
+  Add av_ctz() for trailing zero bit count
 
-2012-10-18 - a893655b - lavu 51.45.0 - error.h
-  Add AVERROR_EXPERIMENTAL.
+2012-10-21 - e3a91c5 / a893655 - lavu 51.77.100 / 51.45.0 - error.h
+  Add AVERROR_EXPERIMENTAL
 
-2012-10-12 - d2fcb356 - lavu 51.44.0 - pixdesc.h
+2012-10-12 - a33ed6b / d2fcb35 - lavu 51.76.100 / 51.44.0 - pixdesc.h
   Add functions for accessing pixel format descriptors.
   Accessing the av_pix_fmt_descriptors array directly is now
   deprecated.
 
-2012-10-11 - 9a92aea2 - lavu 51.43.0 - aes.h, md5.h, sha.h, tree.h
+2012-10-11 - f391e40 / 9a92aea - lavu 51.75.100 / 51.43.0 - aes.h, md5.h, sha.h, tree.h
   Add functions for allocating the opaque contexts for the algorithms,
 
-2012-10-10 - b522000e - lavf 54.18.0 - avio.h
+2012-10-10 - de31814 / b522000 - lavf 54.32.100 / 54.18.0 - avio.h
   Add avio_closep to complement avio_close.
 
-2012-10-06 - 78071a14 - lavu 51.42.0 - pixfmt.h
+2012-10-08 - ae77266 / 78071a1 - lavu 51.74.100 / 51.42.0 - pixfmt.h
   Rename PixelFormat to AVPixelFormat and all PIX_FMT_* to AV_PIX_FMT_*.
   To provide backwards compatibility, PixelFormat is now #defined as
   AVPixelFormat.
@@ -163,23 +199,23 @@ API changes, most recent first:
   'PixelFormat' identifier. Such code should either #undef PixelFormat
   or stop using the PixelFormat name.
 
-2012-10-05 - e7ba5b1 - lavr 1.0.0 - avresample.h
+2012-10-05 - 55c49af / e7ba5b1 - lavr 1.0.0 - avresample.h
   Data planes parameters to avresample_convert() and
   avresample_read() are now uint8_t** instead of void**.
   Libavresample is now stable.
 
-2012-09-24 - a42aada - lavc 54.28.0 - avcodec.h
+2012-09-24 - 46a3595 / a42aada - lavc 54.59.100 / 54.28.0 - avcodec.h
   Add avcodec_free_frame(). This function must now
   be used for freeing an AVFrame.
 
-2012-09-12 - 8919fee - lavu 51.41.0 - audioconvert.h
+2012-09-12 - e3e09f2 / 8919fee - lavu 51.73.100 / 51.41.0 - audioconvert.h
   Added AV_CH_LOW_FREQUENCY_2 channel mask value.
 
-2012-09-04 - 686a329 - lavu 51.40.0 - opt.h
+2012-09-04 - b21b5b0 / 686a329 - lavu 51.71.100 / 51.40.0 - opt.h
   Reordered the fields in default_val in AVOption, changed which
   default_val field is used for which AVOptionType.
 
-2012-08-30 - a231832 - lavc 54.26.1 - avcodec.h
+2012-08-30 - 98298eb / a231832 - lavc 54.54.101 / 54.26.1 - avcodec.h
   Add codec descriptor properties AV_CODEC_PROP_LOSSY and
   AV_CODEC_PROP_LOSSLESS.
 
@@ -187,90 +223,90 @@ API changes, most recent first:
   Add codec descriptors for accessing codec properties without having
   to refer to a specific decoder or encoder.
 
-  c223d79 - Add an AVCodecDescriptor struct and functions
+  f5f3684 / c223d79 - Add an AVCodecDescriptor struct and functions
             avcodec_descriptor_get() and avcodec_descriptor_next().
-  51efed1 - Add AVCodecDescriptor.props and AV_CODEC_PROP_INTRA_ONLY.
-  91e59fe - Add avcodec_descriptor_get_by_name().
+  f5f3684 / 51efed1 - Add AVCodecDescriptor.props and AV_CODEC_PROP_INTRA_ONLY.
+  6c180b3 / 91e59fe - Add avcodec_descriptor_get_by_name().
 
-2012-08-08 - 987170c - lavu 51.38 - dict.h
+2012-08-08 - f5f3684 / 987170c - lavu 51.68.100 / 51.38.0 - dict.h
   Add av_dict_count().
 
-2012-08-07 - 104e10f - lavc 54.25 - avcodec.h
+2012-08-07 - 7a72695 / 104e10f - lavc 54.51.100 / 54.25.0 - avcodec.h
   Rename CodecID to AVCodecID and all CODEC_ID_* to AV_CODEC_ID_*.
   To provide backwards compatibility, CodecID is now #defined as AVCodecID.
   Note that this can break user code that includes avcodec.h and uses the
   'CodecID' identifier. Such code should either #undef CodecID or stop using the
   CodecID name.
 
-2012-08-03 - 239fdf1 - lavu 51.37.1 - cpu.h
+2012-08-03 - e776ee8 / 239fdf1 - lavu 51.66.101 / 51.37.1 - cpu.h
                        lsws 2.1.1   - swscale.h
   Rename AV_CPU_FLAG_MMX2  ---> AV_CPU_FLAG_MMXEXT.
   Rename SWS_CPU_CAPS_MMX2 ---> SWS_CPU_CAPS_MMXEXT.
 
-2012-07-29 - 681ed00 - lavf 54.13.0 - avformat.h
+2012-07-29 - 7c26761 / 681ed00 - lavf 54.22.100 / 54.13.0 - avformat.h
   Add AVFMT_FLAG_NOBUFFER for low latency use cases.
 
 2012-07-10 - 5fade8a - lavu 51.37.0
   Add av_malloc_array() and av_mallocz_array()
 
-2012-06-22 - d3d3a32 - lavu 51.34.0
+2012-06-22 - e847f41 / d3d3a32 - lavu 51.61.100 / 51.34.0
   Add av_usleep()
 
-2012-06-20 - ae0a301 - lavu 51.33.0
+2012-06-20 - 4da42eb / ae0a301 - lavu 51.60.100 / 51.33.0
   Move av_gettime() to libavutil, add libavutil/time.h
 
-2012-06-09 - 3971be0 - lavr 0.0.3
+2012-06-09 - 82edf67 / 3971be0 - lavr 0.0.3
   Add a parameter to avresample_build_matrix() for Dolby/DPLII downmixing.
 
-2012-06-12 - 9baeff9 - lavfi 2.23.0 - avfilter.h
+2012-06-12 - c7b9eab / 9baeff9 - lavfi 2.79.100 / 2.23.0 - avfilter.h
   Add AVFilterContext.nb_inputs/outputs. Deprecate
   AVFilterContext.input/output_count.
 
-2012-06-12 - 84b9fbe - lavfi 2.22.0 - avfilter.h
+2012-06-12 - c7b9eab / 84b9fbe - lavfi 2.79.100 / 2.22.0 - avfilter.h
   Add avfilter_pad_get_type() and avfilter_pad_get_name(). Those
   should now be used instead of accessing AVFilterPad members
   directly.
 
-2012-06-12 - b0f0dfc - lavu 51.32.0 - audioconvert.h
+2012-06-12 - 3630a07 / b0f0dfc - lavu 51.57.100 / 51.32.0 - audioconvert.h
   Add av_get_channel_layout_channel_index(), av_get_channel_name()
   and av_channel_layout_extract_channel().
 
-2012-05-25 - 154486f - lavu 51.31.0 - opt.h
+2012-05-25 - 53ce990 / 154486f - lavu 51.55.100 / 51.31.0 - opt.h
   Add av_opt_set_bin()
 
-2012-05-15 - lavfi 2.17.0
+2012-05-15 - lavfi 2.74.100 / 2.17.0
   Add support for audio filters
-  ac71230/a2cd9be - add video/audio buffer sink in a new installed
+  61930bd / ac71230, 1cbf7fb / a2cd9be - add video/audio buffer sink in a new installed
                     header buffersink.h
-  720c6b7 - add av_buffersrc_write_frame(), deprecate
+  1cbf7fb / 720c6b7 - add av_buffersrc_write_frame(), deprecate
             av_vsrc_buffer_add_frame()
-  ab16504 - add avfilter_copy_buf_props()
-  9453c9e - add extended_data to AVFilterBuffer
-  1b8c927 - add avfilter_get_audio_buffer_ref_from_arrays()
+  61930bd / ab16504 - add avfilter_copy_buf_props()
+  61930bd / 9453c9e - add extended_data to AVFilterBuffer
+  61930bd / 1b8c927 - add avfilter_get_audio_buffer_ref_from_arrays()
 
-2012-05-09 - lavu 51.30.0 - samplefmt.h
-  142e740 - add av_samples_copy()
-  6d7f617 - add av_samples_set_silence()
+2012-05-09 - lavu 51.53.100 / 51.30.0 - samplefmt.h
+  61930bd / 142e740 - add av_samples_copy()
+  61930bd / 6d7f617 - add av_samples_set_silence()
 
-2012-05-09 - a5117a2 - lavc 54.13.1
+2012-05-09 - 61930bd / a5117a2 - lavc 54.21.101 / 54.13.1
   For audio formats with fixed frame size, the last frame
   no longer needs to be padded with silence, libavcodec
   will handle this internally (effectively all encoders
   behave as if they had CODEC_CAP_SMALL_LAST_FRAME set).
 
-2012-05-07 - 828bd08 - lavc 54.13.0 - avcodec.h
+2012-05-07 - 653d117 / 828bd08 - lavc 54.20.100 / 54.13.0 - avcodec.h
   Add sample_rate and channel_layout fields to AVFrame.
 
-2012-05-01 - 4010d72 - lavr 0.0.1
+2012-05-01 - 2330eb1 / 4010d72 - lavr 0.0.1
   Change AV_MIX_COEFF_TYPE_Q6 to AV_MIX_COEFF_TYPE_Q8.
 
-2012-04-25 - 3527a73 - lavu 51.29.0 - cpu.h
+2012-04-25 - e890b68 / 3527a73 - lavu 51.48.100 / 51.29.0 - cpu.h
   Add av_parse_cpu_flags()
 
-2012-04-24 - c8af852 - lavr 0.0.0
+2012-04-24 - 3ead79e / c8af852 - lavr 0.0.0
   Add libavresample audio conversion library
 
-2012-04-20 - 0c0d1bc - lavu 51.28.0 - audio_fifo.h
+2012-04-20 - 3194ab7 / 0c0d1bc - lavu 51.47.100 / 51.28.0 - audio_fifo.h
   Add audio FIFO functions:
     av_audio_fifo_free()
     av_audio_fifo_alloc()
@@ -282,10 +318,10 @@ API changes, most recent first:
     av_audio_fifo_size()
     av_audio_fifo_space()
 
-2012-04-14 - lavfi 2.16.0 - avfiltergraph.h
-  d7bcc71 Add avfilter_graph_parse2().
+2012-04-14 - lavfi 2.70.100 / 2.16.0 - avfiltergraph.h
+  7432bcf / d7bcc71 Add avfilter_graph_parse2().
 
-2012-04-08 - 4d693b0 - lavu 51.27.0 - samplefmt.h
+2012-04-08 - 6bfb304 / 4d693b0 - lavu 51.46.100 / 51.27.0 - samplefmt.h
   Add av_get_packed_sample_fmt() and av_get_planar_sample_fmt()
 
 2012-03-21 - b75c67d - lavu 51.43.100
@@ -313,73 +349,73 @@ API changes, most recent first:
 2012-01-24 - 0c3577b - lavfi 2.60.100
   Add avfilter_graph_dump.
 
-2012-03-20 - 3c90cc2 - lavfo 54.2.0
+2012-03-20 - 0ebd836 / 3c90cc2 - lavfo 54.2.0
   Deprecate av_read_packet(), use av_read_frame() with
   AVFMT_FLAG_NOPARSE | AVFMT_FLAG_NOFILLIN in AVFormatContext.flags
 
-2012-03-05 - lavc 54.8.0
-  6699d07 Add av_get_exact_bits_per_sample()
-  9524cf7 Add av_get_audio_frame_duration()
+2012-03-05 - lavc 54.10.100 / 54.8.0
+  f095391 / 6699d07 Add av_get_exact_bits_per_sample()
+  f095391 / 9524cf7 Add av_get_audio_frame_duration()
 
-2012-03-04 - 44fe77b - lavc 54.7.0 - avcodec.h
+2012-03-04 - 2af8f2c / 44fe77b - lavc 54.8.100 / 54.7.0 - avcodec.h
   Add av_codec_is_encoder/decoder().
 
-2012-03-01 - 442c132 - lavc 54.3.0 - avcodec.h
+2012-03-01 - 1eb7f39 / 442c132 - lavc 54.5.100 / 54.3.0 - avcodec.h
   Add av_packet_shrink_side_data.
 
-2012-02-29 - dd2a4bc - lavf 54.2.0 - avformat.h
+2012-02-29 - 79ae084 / dd2a4bc - lavf 54.2.100 / 54.2.0 - avformat.h
   Add AVStream.attached_pic and AV_DISPOSITION_ATTACHED_PIC,
   used for dealing with attached pictures/cover art.
 
-2012-02-25 - c9bca80 - lavu 51.24.0 - error.h
+2012-02-25 - 305e4b3 / c9bca80 - lavu 51.41.100 / 51.24.0 - error.h
   Add AVERROR_UNKNOWN
   NOTE: this was backported to 0.8
 
-2012-02-20 - e9cda85 - lavc 54.2.0
+2012-02-20 - eadd426 / e9cda85 - lavc 54.2.100 / 54.2.0
   Add duration field to AVCodecParserContext
 
-2012-02-20 - 0b42a93 - lavu 51.23.1 - mathematics.h
+2012-02-20 - eadd426 / 0b42a93 - lavu 51.40.100 / 51.23.1 - mathematics.h
   Add av_rescale_q_rnd()
 
-2012-02-08 - 38d5533 - lavu 51.22.1 - pixdesc.h
+2012-02-08 - f2b20b7 / 38d5533 - lavu 51.38.101 / 51.22.1 - pixdesc.h
   Add PIX_FMT_PSEUDOPAL flag.
 
-2012-02-08 - 52f82a1 - lavc 54.01.0
+2012-02-08 - f2b20b7 / 52f82a1 - lavc 54.2.100 / 54.1.0
   Add avcodec_encode_video2() and deprecate avcodec_encode_video().
 
-2012-02-01 - 316fc74 - lavc 54.01.0
+2012-02-01 - 4c677df / 316fc74 - lavc 54.1.0
   Add av_fast_padded_malloc() as alternative for av_realloc() when aligned
   memory is required. The buffer will always have FF_INPUT_BUFFER_PADDING_SIZE
   zero-padded bytes at the end.
 
-2012-01-31 - dd6d3b0 - lavf 54.01.0
+2012-01-31 - a369a6b / dd6d3b0 - lavf 54.1.0
   Add avformat_get_riff_video_tags() and avformat_get_riff_audio_tags().
   NOTE: this was backported to 0.8
 
-2012-01-31 - af08d9a - lavc 54.01.0
+2012-01-31 - a369a6b / af08d9a - lavc 54.1.0
   Add avcodec_is_open() function.
   NOTE: this was backported to 0.8
 
-2012-01-30 - 8b93312 - lavu 51.22.0 - intfloat.h
+2012-01-30 - 151ecc2 / 8b93312 - lavu 51.36.100 / 51.22.0 - intfloat.h
   Add a new installed header libavutil/intfloat.h with int/float punning
   functions.
   NOTE: this was backported to 0.8
 
-2012-01-25 - lavf 53.22.0
-  f1caf01 Allow doing av_write_frame(ctx, NULL) for flushing possible
+2012-01-25 - lavf 53.31.100 / 53.22.0
+  3c5fe5b / f1caf01 Allow doing av_write_frame(ctx, NULL) for flushing possible
           buffered data within a muxer. Added AVFMT_ALLOW_FLUSH for
           muxers supporting it (av_write_frame makes sure it is called
           only for muxers with this flag).
 
-2012-01-15 - lavc 53.34.0
+2012-01-15 - lavc 53.56.105 / 53.34.0
   New audio encoding API:
-  b2c75b6 Add CODEC_CAP_VARIABLE_FRAME_SIZE capability for use by audio
+  67f5650 / b2c75b6 Add CODEC_CAP_VARIABLE_FRAME_SIZE capability for use by audio
           encoders.
-  5ee5fa0 Add avcodec_fill_audio_frame() as a convenience function.
-  b2c75b6 Add avcodec_encode_audio2() and deprecate avcodec_encode_audio().
+  67f5650 / 5ee5fa0 Add avcodec_fill_audio_frame() as a convenience function.
+  67f5650 / b2c75b6 Add avcodec_encode_audio2() and deprecate avcodec_encode_audio().
           Add AVCodec.encode2().
 
-2012-01-12 - 3167dc9 - lavfi 2.15.0
+2012-01-12 - b18e17e / 3167dc9 - lavfi 2.59.100 / 2.15.0
   Add a new installed header -- libavfilter/version.h -- with version macros.
 
 2011-12-08 - a502939 - lavfi 2.52.0
@@ -400,37 +436,37 @@ API changes, most recent first:
 2011-10-20 - b35e9e1 - lavu 51.22.0
   Add av_strtok() to avstring.h.
 
-2011-01-03 - b73ec05 - lavu 51.21.0
+2012-01-03 - ad1c8dd / b73ec05 - lavu 51.34.100 / 51.21.0
   Add av_popcount64
 
-2011-12-18 - 8400b12 - lavc 53.28.1
+2011-12-18 - 7c29313 / 8400b12 - lavc 53.46.1 / 53.28.1
   Deprecate AVFrame.age. The field is unused.
 
-2011-12-12 - 5266045 - lavf 53.17.0
+2011-12-12 - 8bc7fe4 / 5266045 - lavf 53.25.0 / 53.17.0
   Add avformat_close_input().
   Deprecate av_close_input_file() and av_close_input_stream().
 
-2011-12-02 - 0eea212 - lavc 53.25.0
+2011-12-02 - e4de716 / 0eea212 - lavc 53.40.0 / 53.25.0
   Add nb_samples and extended_data fields to AVFrame.
   Deprecate AVCODEC_MAX_AUDIO_FRAME_SIZE.
   Deprecate avcodec_decode_audio3() in favor of avcodec_decode_audio4().
   avcodec_decode_audio4() writes output samples to an AVFrame, which allows
   audio decoders to use get_buffer().
 
-2011-12-04 - 560f773 - lavc 53.24.0
+2011-12-04 - e4de716 / 560f773 - lavc 53.40.0 / 53.24.0
   Change AVFrame.data[4]/base[4]/linesize[4]/error[4] to [8] at next major bump.
   Change AVPicture.data[4]/linesize[4] to [8] at next major bump.
   Change AVCodecContext.error[4] to [8] at next major bump.
   Add AV_NUM_DATA_POINTERS to simplify the bump transition.
 
-2011-11-23 - bbb46f3 - lavu 51.18.0
+2011-11-23 - 8e576d5 / bbb46f3 - lavu 51.27.0 / 51.18.0
   Add av_samples_get_buffer_size(), av_samples_fill_arrays(), and
   av_samples_alloc(), to samplefmt.h.
 
-2011-11-23 - 8889cc4 - lavu 51.17.0
+2011-11-23 - 8e576d5 / 8889cc4 - lavu 51.27.0 / 51.17.0
   Add planar sample formats and av_sample_fmt_is_planar() to samplefmt.h.
 
-2011-11-19 - f3a29b7 - lavc 53.21.0
+2011-11-19 - dbb38bc / f3a29b7 - lavc 53.36.0 / 53.21.0
   Move some AVCodecContext fields to a new private struct, AVCodecInternal,
   which is accessed from a new field, AVCodecContext.internal.
   - fields moved:
@@ -438,55 +474,55 @@ API changes, most recent first:
       AVCodecContext.internal_buffer_count --> AVCodecInternal.buffer_count
       AVCodecContext.is_copy               --> AVCodecInternal.is_copy
 
-2011-11-16 - 6270671 - lavu 51.16.0
+2011-11-16 - 8709ba9 / 6270671 - lavu 51.26.0 / 51.16.0
   Add av_timegm()
 
-2011-11-13 - lavf 53.15.0
+2011-11-13 - lavf 53.21.0 / 53.15.0
   New interrupt callback API, allowing per-AVFormatContext/AVIOContext
   interrupt callbacks.
-  6aa0b98 Add AVIOInterruptCB struct and the interrupt_callback field to
+  5f268ca / 6aa0b98 Add AVIOInterruptCB struct and the interrupt_callback field to
           AVFormatContext.
-  1dee0ac Add avio_open2() with additional parameters. Those are
+  5f268ca / 1dee0ac Add avio_open2() with additional parameters. Those are
           an interrupt callback and an options AVDictionary.
           This will allow passing AVOptions to protocols after lavf
           54.0.
 
-2011-11-06 - ba04ecf - lavu 51.14.0
+2011-11-06 - 13b7781 / ba04ecf - lavu 51.24.0 / 51.14.0
   Add av_strcasecmp() and av_strncasecmp() to avstring.h.
 
-2011-11-06 - 07b172f - lavu 51.13.0
+2011-11-06 - 13b7781 / 07b172f - lavu 51.24.0 / 51.13.0
   Add av_toupper()/av_tolower()
 
-2011-11-05 - b6d08f4 - lavf 53.13.0
+2011-11-05 - d8cab5c / b6d08f4 - lavf 53.19.0 / 53.13.0
   Add avformat_network_init()/avformat_network_deinit()
 
-2011-10-27 - 512557b - lavc 53.15.0
+2011-10-27 - 6faf0a2 / 512557b - lavc 53.24.0 / 53.15.0
   Remove avcodec_parse_frame.
   Deprecate AVCodecContext.parse_only and CODEC_CAP_PARSE_ONLY.
 
-2011-10-19 - 569129a - lavf 53.10.0
+2011-10-19 - d049257 / 569129a - lavf 53.17.0 / 53.10.0
   Add avformat_new_stream(). Deprecate av_new_stream().
 
-2011-10-13 - b631fba - lavf 53.9.0
+2011-10-13 - 91eb1b1 / b631fba - lavf 53.16.0 / 53.9.0
   Add AVFMT_NO_BYTE_SEEK AVInputFormat flag.
 
-2011-10-12 - lavu 51.12.0
+2011-10-12 - lavu 51.21.0 / 51.12.0
   AVOptions API rewrite.
 
-  - 145f741 FF_OPT_TYPE* renamed to AV_OPT_TYPE_*
+  - f884ef0 / 145f741 FF_OPT_TYPE* renamed to AV_OPT_TYPE_*
   - new setting/getting functions with slightly different semantics:
-        dac66da av_set_string3 -> av_opt_set
+        f884ef0 / dac66da av_set_string3 -> av_opt_set
                 av_set_double  -> av_opt_set_double
                 av_set_q       -> av_opt_set_q
                 av_set_int     -> av_opt_set_int
 
-        41d9d51 av_get_string  -> av_opt_get
+        f884ef0 / 41d9d51 av_get_string  -> av_opt_get
                 av_get_double  -> av_opt_get_double
                 av_get_q       -> av_opt_get_q
                 av_get_int     -> av_opt_get_int
 
-  - 8c5dcaa trivial rename av_next_option -> av_opt_next
-  - 641c7af new functions - av_opt_child_next, av_opt_child_class_next
+  - f884ef0 / 8c5dcaa trivial rename av_next_option -> av_opt_next
+  - f884ef0 / 641c7af new functions - av_opt_child_next, av_opt_child_class_next
     and av_opt_find2()
 
 2011-09-22 - a70e787 - lavu 51.17.0
@@ -532,31 +568,31 @@ API changes, most recent first:
 2011-08-20 - 69e2c1a - lavu 51.13.0
   Add av_get_media_type_string().
 
-2011-09-03 - fb4ca26 - lavc 53.13.0
+2011-09-03 - 1889c67 / fb4ca26 - lavc 53.13.0
                        lavf 53.11.0
                        lsws  2.1.0
   Add {avcodec,avformat,sws}_get_class().
 
-2011-08-03 - c11fb82 - lavu 51.15.0
+2011-08-03 - 1889c67 / c11fb82 - lavu 51.15.0
   Add AV_OPT_SEARCH_FAKE_OBJ flag for av_opt_find() function.
 
 2011-08-14 - 323b930 - lavu 51.12.0
   Add av_fifo_peek2(), deprecate av_fifo_peek().
 
-2011-08-26 - lavu 51.9.0
-  - add41de..abc78a5 Do not include intfloat_readwrite.h,
+2011-08-26 - lavu 51.14.0 / 51.9.0
+  - 976a8b2 / add41de..976a8b2 / abc78a5 Do not include intfloat_readwrite.h,
     mathematics.h, rational.h, pixfmt.h, or log.h from avutil.h.
 
-2011-08-16 - 48f9e45 - lavf 53.8.0
+2011-08-16 - 27fbe31 / 48f9e45 - lavf 53.11.0 / 53.8.0
   Add avformat_query_codec().
 
-2011-08-16 - bca06e7 - lavc 53.11.0
+2011-08-16 - 27fbe31 / bca06e7 - lavc 53.11.0
   Add avcodec_get_type().
 
-2011-08-06 - 2f63440 - lavf 53.7.0
+2011-08-06 - 0cb233c / 2f63440 - lavf 53.7.0
   Add error_recognition to AVFormatContext.
 
-2011-08-02 - 9d39cbf - lavc 53.9.1
+2011-08-02 - 1d186e9 / 9d39cbf - lavc 53.9.1
   Add AV_PKT_FLAG_CORRUPT AVPacket flag.
 
 2011-07-16 - b57df29 - lavfi 2.27.0
@@ -567,11 +603,11 @@ API changes, most recent first:
   avfilter_set_common_packing_formats()
   avfilter_all_packing_formats()
 
-2011-07-10 - a67c061 - lavf 53.6.0
+2011-07-10 - 3602ad7 / a67c061 - lavf 53.6.0
   Add avformat_find_stream_info(), deprecate av_find_stream_info().
   NOTE: this was backported to 0.7
 
-2011-07-10 - 0b950fe - lavc 53.8.0
+2011-07-10 - 3602ad7 / 0b950fe - lavc 53.8.0
   Add avcodec_open2(), deprecate avcodec_open().
   NOTE: this was backported to 0.7
 
@@ -614,35 +650,35 @@ API changes, most recent first:
 2011-06-12 - 6119b23 - lavfi 2.16.0 - avfilter_graph_parse()
   Change avfilter_graph_parse() signature.
 
-2011-06-23 - 67e9ae1 - lavu 51.8.0 - attributes.h
+2011-06-23 - 686959e / 67e9ae1 - lavu 51.10.0 / 51.8.0 - attributes.h
   Add av_printf_format().
 
-2011-06-16 - 05e84c9, 25de595 - lavf 53.2.0 - avformat.h
+2011-06-16 - 2905e3f / 05e84c9, 2905e3f / 25de595 - lavf 53.4.0 / 53.2.0 - avformat.h
   Add avformat_open_input and avformat_write_header().
   Deprecate av_open_input_stream, av_open_input_file,
   AVFormatParameters and av_write_header.
 
-2011-06-16 - 7e83e1c, dc59ec5 - lavu 51.7.0 - opt.h
+2011-06-16 - 2905e3f / 7e83e1c, 2905e3f / dc59ec5 - lavu 51.9.0 / 51.7.0 - opt.h
   Add av_opt_set_dict() and av_opt_find().
   Deprecate av_find_opt().
   Add AV_DICT_APPEND flag.
 
-2011-06-10 - cb7c11c - lavu 51.6.0 - opt.h
+2011-06-10 - 45fb647 / cb7c11c - lavu 51.6.0 - opt.h
   Add av_opt_flag_is_set().
 
 2011-06-10 - c381960 - lavfi 2.15.0 - avfilter_get_audio_buffer_ref_from_arrays
   Add avfilter_get_audio_buffer_ref_from_arrays() to avfilter.h.
 
-2011-06-09 - d9f80ea - lavu 51.8.0 - AVMetadata
+2011-06-09 - f9ecb84 / d9f80ea - lavu 51.8.0 - AVMetadata
   Move AVMetadata from lavf to lavu and rename it to
   AVDictionary -- new installed header dict.h.
   All av_metadata_* functions renamed to av_dict_*.
 
-2011-06-07 - a6703fa - lavu 51.8.0 - av_get_bytes_per_sample()
+2011-06-07 - d552f61 / a6703fa - lavu 51.8.0 - av_get_bytes_per_sample()
   Add av_get_bytes_per_sample() in libavutil/samplefmt.h.
   Deprecate av_get_bits_per_sample_fmt().
 
-2011-06-05 - b39b062 - lavu 51.8.0 - opt.h
+2011-06-05 - f956924 / b39b062 - lavu 51.8.0 - opt.h
   Add av_opt_free convenience function.
 
 2011-06-06 - 95a0242 - lavfi 2.14.0 - AVFilterBufferRefAudioProps
@@ -672,7 +708,7 @@ API changes, most recent first:
   Add av_get_pix_fmt_name() in libavutil/pixdesc.h, and deprecate
   avcodec_get_pix_fmt_name() in libavcodec/avcodec.h in its favor.
 
-2011-05-25 - 30315a8 - lavf 53.3.0 - avformat.h
+2011-05-25 - 39e4206 / 30315a8 - lavf 53.3.0 - avformat.h
   Add fps_probe_size to AVFormatContext.
 
 2011-05-22 - 5ecdfd0 - lavf 53.2.0 - avformat.h
@@ -688,10 +724,10 @@ API changes, most recent first:
 2011-05-14 - 9fdf772 - lavfi 2.6.0 - avcodec.h
   Add avfilter_get_video_buffer_ref_from_frame() to libavfilter/avcodec.h.
 
-2011-05-18 - 64150ff - lavc 53.7.0 - AVCodecContext.request_sample_fmt
+2011-05-18 - 75a37b5 / 64150ff - lavc 53.7.0 - AVCodecContext.request_sample_fmt
   Add request_sample_fmt field to AVCodecContext.
 
-2011-05-10 - 188dea1 - lavc 53.6.0 - avcodec.h
+2011-05-10 - 59eb12f / 188dea1 - lavc 53.6.0 - avcodec.h
   Deprecate AVLPCType and the following fields in
   AVCodecContext: lpc_coeff_precision, prediction_order_method,
   min_partition_order, max_partition_order, lpc_type, lpc_passes.
@@ -721,81 +757,81 @@ API changes, most recent first:
   Add av_dynarray_add function for adding
   an element to a dynamic array.
 
-2011-04-26 - bebe72f - lavu 51.1.0 - avutil.h
+2011-04-26 - d7e5aeb / bebe72f - lavu 51.1.0 - avutil.h
   Add AVPictureType enum and av_get_picture_type_char(), deprecate
   FF_*_TYPE defines and av_get_pict_type_char() defined in
   libavcodec/avcodec.h.
 
-2011-04-26 - 10d3940 - lavfi 2.3.0 - avfilter.h
+2011-04-26 - d7e5aeb / 10d3940 - lavfi 2.3.0 - avfilter.h
   Add pict_type and key_frame fields to AVFilterBufferRefVideo.
 
-2011-04-26 - 7a11c82 - lavfi 2.2.0 - vsrc_buffer
+2011-04-26 - d7e5aeb / 7a11c82 - lavfi 2.2.0 - vsrc_buffer
   Add sample_aspect_ratio fields to vsrc_buffer arguments
 
-2011-04-21 - 94f7451 - lavc 53.1.0 - avcodec.h
+2011-04-21 - 8772156 / 94f7451 - lavc 53.1.0 - avcodec.h
   Add CODEC_CAP_SLICE_THREADS for codecs supporting sliced threading.
 
 2011-04-15 - lavc 52.120.0 - avcodec.h
   AVPacket structure got additional members for passing side information:
-    4de339e introduce side information for AVPacket
-    2d8591c make containers pass palette change in AVPacket
+    c407984 / 4de339e introduce side information for AVPacket
+    c407984 / 2d8591c make containers pass palette change in AVPacket
 
 2011-04-12 - lavf 52.107.0 - avio.h
   Avio cleanup, part II - deprecate the entire URLContext API:
-    175389c add avio_check as a replacement for url_exist
-    ff1ec0c add avio_pause and avio_seek_time as replacements
+    c55780d / 175389c add avio_check as a replacement for url_exist
+    9891004 / ff1ec0c add avio_pause and avio_seek_time as replacements
             for _av_url_read_fseek/fpause
-    cdc6a87 deprecate av_protocol_next(), avio_enum_protocols
+    d4d0932 / cdc6a87 deprecate av_protocol_next(), avio_enum_protocols
             should be used instead.
-    80c6e23 rename url_set_interrupt_cb->avio_set_interrupt_cb.
-    f87b1b3 rename open flags: URL_* -> AVIO_*
-    f8270bb add avio_enum_protocols.
-    5593f03 deprecate URLProtocol.
-    c486dad deprecate URLContext.
-    026e175 deprecate the typedef for URLInterruptCB
-    8e76a19 deprecate av_register_protocol2.
-    b840484 deprecate URL_PROTOCOL_FLAG_NESTED_SCHEME
-    1305d93 deprecate av_url_read_seek
-    fa104e1 deprecate av_url_read_pause
-    727c7aa deprecate url_get_filename().
-    5958df3 deprecate url_max_packet_size().
-    1869ea0 deprecate url_get_file_handle().
-    32a97d4 deprecate url_filesize().
-    e52a914 deprecate url_close().
-    58a48c6 deprecate url_seek().
-    925e908 deprecate url_write().
-    dce3756 deprecate url_read_complete().
-    bc371ac deprecate url_read().
-    0589da0 deprecate url_open().
-    62eaaea deprecate url_connect.
-    5652bb9 deprecate url_alloc.
-    333e894 deprecate url_open_protocol
-    e230705 deprecate url_poll and URLPollEntry
+    c88caa5 / 80c6e23 rename url_set_interrupt_cb->avio_set_interrupt_cb.
+    c88caa5 / f87b1b3 rename open flags: URL_* -> AVIO_*
+    d4d0932 / f8270bb add avio_enum_protocols.
+    d4d0932 / 5593f03 deprecate URLProtocol.
+    d4d0932 / c486dad deprecate URLContext.
+    d4d0932 / 026e175 deprecate the typedef for URLInterruptCB
+    c88caa5 / 8e76a19 deprecate av_register_protocol2.
+    11d7841 / b840484 deprecate URL_PROTOCOL_FLAG_NESTED_SCHEME
+    11d7841 / 1305d93 deprecate av_url_read_seek
+    11d7841 / fa104e1 deprecate av_url_read_pause
+    434f248 / 727c7aa deprecate url_get_filename().
+    434f248 / 5958df3 deprecate url_max_packet_size().
+    434f248 / 1869ea0 deprecate url_get_file_handle().
+    434f248 / 32a97d4 deprecate url_filesize().
+    434f248 / e52a914 deprecate url_close().
+    434f248 / 58a48c6 deprecate url_seek().
+    434f248 / 925e908 deprecate url_write().
+    434f248 / dce3756 deprecate url_read_complete().
+    434f248 / bc371ac deprecate url_read().
+    434f248 / 0589da0 deprecate url_open().
+    434f248 / 62eaaea deprecate url_connect.
+    434f248 / 5652bb9 deprecate url_alloc.
+    434f248 / 333e894 deprecate url_open_protocol
+    434f248 / e230705 deprecate url_poll and URLPollEntry
 
 2011-04-08 - lavf 52.106.0 - avformat.h
   Minor avformat.h cleanup:
-    a9bf9d8 deprecate av_guess_image2_codec
-    c3675df rename avf_sdp_create->av_sdp_create
+    d4d0932 / a9bf9d8 deprecate av_guess_image2_codec
+    d4d0932 / c3675df rename avf_sdp_create->av_sdp_create
 
 2011-04-03 - lavf 52.105.0 - avio.h
   Large-scale renaming/deprecating of AVIOContext-related functions:
-    724f6a0 deprecate url_fdopen
-    403ee83 deprecate url_open_dyn_packet_buf
-    6dc7d80 rename url_close_dyn_buf       -> avio_close_dyn_buf
-    b92c545 rename url_open_dyn_buf        -> avio_open_dyn_buf
-    8978fed introduce an AVIOContext.seekable field as a replacement for
+    2cae980 / 724f6a0 deprecate url_fdopen
+    2cae980 / 403ee83 deprecate url_open_dyn_packet_buf
+    2cae980 / 6dc7d80 rename url_close_dyn_buf       -> avio_close_dyn_buf
+    2cae980 / b92c545 rename url_open_dyn_buf        -> avio_open_dyn_buf
+    2cae980 / 8978fed introduce an AVIOContext.seekable field as a replacement for
             AVIOContext.is_streamed and url_is_streamed()
-    b64030f deprecate get_checksum()
-    4c4427a deprecate init_checksum()
-    4ec153b deprecate udp_set_remote_url/get_local_port
-    933e90a deprecate av_url_read_fseek/fpause
-    8d9769a deprecate url_fileno
-    b7f2fdd rename put_flush_packet -> avio_flush
-    35f1023 deprecate url_close_buf
-    83fddae deprecate url_open_buf
-    d9d86e0 rename url_fprintf -> avio_printf
-    59f65d9 deprecate url_setbufsize
-    3e68b3b deprecate url_ferror
+    1caa412 / b64030f deprecate get_checksum()
+    1caa412 / 4c4427a deprecate init_checksum()
+    2fd41c9 / 4ec153b deprecate udp_set_remote_url/get_local_port
+    4fa0e24 / 933e90a deprecate av_url_read_fseek/fpause
+    4fa0e24 / 8d9769a deprecate url_fileno
+    0fecf26 / b7f2fdd rename put_flush_packet -> avio_flush
+    0fecf26 / 35f1023 deprecate url_close_buf
+    0fecf26 / 83fddae deprecate url_open_buf
+    0fecf26 / d9d86e0 rename url_fprintf -> avio_printf
+    0fecf26 / 59f65d9 deprecate url_setbufsize
+    6947b0c / 3e68b3b deprecate url_ferror
     e8bb2e2 deprecate url_fget_max_packet_size
     76aa876 rename url_fsize -> avio_size
     e519753 deprecate url_fgetc
@@ -816,7 +852,7 @@ API changes, most recent first:
     b3db9ce deprecate get_partial_buffer
     8d9ac96 rename av_alloc_put_byte -> avio_alloc_context
 
-2011-03-25 - 34b47d7 - lavc 52.115.0 - AVCodecContext.audio_service_type
+2011-03-25 - 27ef7b1 / 34b47d7 - lavc 52.115.0 - AVCodecContext.audio_service_type
   Add audio_service_type field to AVCodecContext.
 
 2011-03-17 - e309fdc - lavu 50.40.0 - pixfmt.h
@@ -854,11 +890,11 @@ API changes, most recent first:
 2011-02-10 - 12c14cd - lavf 52.99.0 - AVStream.disposition
   Add AV_DISPOSITION_HEARING_IMPAIRED and AV_DISPOSITION_VISUAL_IMPAIRED.
 
-2011-02-09 - 5592734 - lavc 52.112.0 - avcodec_thread_init()
+2011-02-09 - c0b102c - lavc 52.112.0 - avcodec_thread_init()
   Deprecate avcodec_thread_init()/avcodec_thread_free() use; instead
   set thread_count before calling avcodec_open.
 
-2011-02-09 - 778b08a - lavc 52.111.0 - threading API
+2011-02-09 - 37b00b4 - lavc 52.111.0 - threading API
   Add CODEC_CAP_FRAME_THREADS with new restrictions on get_buffer()/
   release_buffer()/draw_horiz_band() callbacks for appropriate codecs.
   Add thread_type and active_thread_type fields to AVCodecContext.

doc/Doxyfile

@@ -31,7 +31,7 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.
 
-PROJECT_NUMBER         =
+PROJECT_NUMBER         = 1.2.7
 
 # 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

doc/Makefile

@@ -62,7 +62,7 @@ doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
 	$(M)texi2html -I doc -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
 
 doc/%.pod: TAG = POD
-doc/%.pod: doc/%.texi $(GENTEXI)
+doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
 	$(Q)$(TEXIDEP)
 	$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Idoc $< $@
 
@@ -92,9 +92,9 @@ uninstall: uninstall-man
 uninstall-man:
 	$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
 
-docclean: clean
+clean:: docclean
 
-clean::
+docclean:
 	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 $(CLEANSUFFIXES:%=doc/%) doc/avoptions_*.texi
 	$(RM) -r doc/doxy/html
 

doc/RELEASE_NOTES

@@ -1,7 +1,7 @@
 Release Notes
 =============
 
-* 0.10 "Freedom"  January, 2012
+* 1.2 "Magic"  March, 2013
 
 
 General notes
@@ -14,9 +14,3 @@ accepted. If you are experiencing issues with any formally released version of
 FFmpeg, please try git master to check if the issue still exists. If it does,
 make your report against the development code following the usual bug reporting
 guidelines.
-
-Of big interest to our Windows users, FFmpeg now supports building with the MSVC
-compiler. Since MSVC does not support C99 features used extensively by FFmpeg,
-this has been accomplished using a converter that turns C99 code to C89. See the
-platform-specific documentation for more detailed documentation on building
-FFmpeg with MSVC.

doc/decoders.texi

@@ -60,6 +60,78 @@ This decoder generates wave patterns according to predefined sequences. Its
 use is purely internal and the format of the data it accepts is not publicly
 documented.
 
+@section libcelt
+
+libcelt decoder wrapper
+
+libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec.
+Requires the presence of the libcelt headers and library during configuration.
+You need to explicitly configure the build with @code{--enable-libcelt}.
+
+@section libgsm
+
+libgsm decoder wrapper
+
+libgsm allows libavcodec to decode the GSM full rate audio codec. Requires
+the presence of the libgsm headers and library during configuration. You need
+to explicitly configure the build with @code{--enable-libgsm}.
+
+This decoder supports both the ordinary GSM and the Microsoft variant.
+
+@section libilbc
+
+libilbc decoder wrapper
+
+libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC)
+audio codec. Requires the presence of the libilbc headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libilbc}.
+
+@subsection Options
+
+The following option is supported by the libilbc wrapper.
+
+@table @option
+@item enhance
+
+Enable the enhancement of the decoded audio when set to 1. The default
+value is 0 (disabled).
+
+@end table
+
+@section libopencore-amrnb
+
+libopencore-amrnb decoder wrapper
+
+libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate
+Narrowband audio codec. Using it requires the presence of the
+libopencore-amrnb headers and library during configuration. You need to
+explicitly configure the build with @code{--enable-libopencore-amrnb}.
+
+An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB
+without this library.
+
+@section libopencore-amrwb
+
+libopencore-amrwb decoder wrapper.
+
+libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate
+Wideband audio codec. Using it requires the presence of the
+libopencore-amrwb headers and library during configuration. You need to
+explicitly configure the build with @code{--enable-libopencore-amrwb}.
+
+An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB
+without this library.
+
+@section libopus
+
+libopus decoder wrapper.
+
+libopus allows libavcodec to decode the Opus Interactive Audio Codec.
+Requires the presence of the libopus headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libopus}.
+
 @c man end AUDIO DECODERS
 
 @chapter Subtitles Decoders

doc/demuxers.texi

@@ -6,18 +6,100 @@ multimedia streams from a particular type of file.
 
 When you configure your FFmpeg build, all the supported demuxers
 are enabled by default. You can list all available ones using the
-configure option "--list-demuxers".
+configure option @code{--list-demuxers}.
 
 You can disable all the demuxers using the configure option
-"--disable-demuxers", and selectively enable a single demuxer with
-the option "--enable-demuxer=@var{DEMUXER}", or disable it
-with the option "--disable-demuxer=@var{DEMUXER}".
+@code{--disable-demuxers}, and selectively enable a single demuxer with
+the option @code{--enable-demuxer=@var{DEMUXER}}, or disable it
+with the option @code{--disable-demuxer=@var{DEMUXER}}.
 
-The option "-formats" of the ff* tools will display the list of
+The option @code{-formats} of the ff* tools will display the list of
 enabled demuxers.
 
 The description of some of the currently available demuxers follows.
 
+@section applehttp
+
+Apple HTTP Live Streaming demuxer.
+
+This demuxer presents all AVStreams from all variant streams.
+The id field is set to the bitrate variant index number. By setting
+the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
+the caller can decide which variant streams to actually receive.
+The total bitrate of the variant that the stream belongs to is
+available in a metadata key named "variant_bitrate".
+
+@anchor{concat}
+@section concat
+
+Virtual concatenation script demuxer.
+
+This demuxer reads a list of files and other directives from a text file and
+demuxes them one after the other, as if all their packet had been muxed
+together.
+
+The timestamps in the files are adjusted so that the first file starts at 0
+and each next file starts where the previous one finishes. Note that it is
+done globally and may cause gaps if all streams do not have exactly the same
+length.
+
+All files must have the same streams (same codecs, same time base, etc.).
+
+The duration of each file is used to adjust the timestamps of the next file:
+if the duration is incorrect (because it was computed using the bit-rate or
+because the file is truncated, for example), it can cause artifacts. The
+@code{duration} directive can be used to override the duration stored in
+each file.
+
+@subsection Syntax
+
+The script is a text file in extended-ASCII, with one directive per line.
+Empty lines, leading spaces and lines starting with '#' are ignored. The
+following directive is recognized:
+
+@table @option
+
+@item @code{file @var{path}}
+Path to a file to read; special characters and spaces must be escaped with
+backslash or single quotes.
+
+All subsequent directives apply to that file.
+
+@item @code{ffconcat version 1.0}
+Identify the script type and version. It also sets the @option{safe} option
+to 1 if it was to its default -1.
+
+To make FFmpeg recognize the format automatically, this directive must
+appears exactly as is (no extra space or byte-order-mark) on the very first
+line of the script.
+
+@item @code{duration @var{dur}}
+Duration of the file. This information can be specified from the file;
+specifying it here may be more efficient or help if the information from the
+file is not available or accurate.
+
+@end table
+
+@subsection Options
+
+This demuxer accepts the following option:
+
+@table @option
+
+@item safe
+If set to 1, reject unsafe file paths. A file path is considered safe if it
+does not contain a protocol specification and is relative and all components
+only contain characters from the portable character set (letters, digits,
+period, underscore and hyphen) and have no period at the beginning of a
+component.
+
+If set to 0, any file name is accepted.
+
+The default is -1, it is equivalent to 1 if the format was automatically
+probed and 0 otherwise.
+
+@end table
+
 @section image2
 
 Image file demuxer.
@@ -143,16 +225,34 @@ ffmpeg -pattern_type glob -i "*.png" -r 10 out.mkv
 @end example
 @end itemize
 
-@section applehttp
+@section rawvideo
 
-Apple HTTP Live Streaming demuxer.
+Raw video demuxer.
 
-This demuxer presents all AVStreams from all variant streams.
-The id field is set to the bitrate variant index number. By setting
-the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
-the caller can decide which variant streams to actually receive.
-The total bitrate of the variant that the stream belongs to is
-available in a metadata key named "variant_bitrate".
+This demuxer allows to read raw video data. Since there is no header
+specifying the assumed video parameters, the user must specify them
+in order to be able to decode the data correctly.
+
+This demuxer accepts the following options:
+@table @option
+
+@item framerate
+Set input video frame rate. Default value is 25.
+
+@item pixel_format
+Set the input video pixel format. Default value is @code{yuv420p}.
+
+@item video_size
+Set the input video size. This value must be specified explicitly.
+@end table
+
+For example to read a rawvideo file @file{input.raw} with
+@command{ffplay}, assuming a pixel format of @code{rgb24}, a video
+size of @code{320x240}, and a frame rate of 10 images per second, use
+the command:
+@example
+ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw
+@end example
 
 @section sbg
 
@@ -184,37 +284,6 @@ the script is directly played, the actual times will match the absolute
 timestamps up to the sound controller's clock accuracy, but if the user
 somehow pauses the playback or seeks, all times will be shifted accordingly.
 
-@section concat
-
-Virtual concatenation script demuxer.
-
-This demuxer reads a list of files and other directives from a text file and
-demuxes them one after the other, as if all their packet had been muxed
-together.
-
-The timestamps in the files are adjusted so that the first file starts at 0
-and each next file starts where the previous one finishes. Note that it is
-done globally and may cause gaps if all streams do not have exactly the same
-length.
-
-All files must have the same streams (same codecs, same time base, etc.).
-
-This script format can currently not be probed, it must be specified explicitly.
-
-@subsection Syntax
-
-The script is a text file in extended-ASCII, with one directive per line.
-Empty lines, leading spaces and lines starting with '#' are ignored. The
-following directive is recognized:
-
-@table @option
-
-@item @code{file @var{path}}
-Path to a file to read; special characters and spaces must be escaped with
-backslash or single quotes.
-
-@end table
-
 @section tedcaptions
 
 JSON captions used for @url{http://www.ted.com/, TED Talks}.
@@ -236,4 +305,4 @@ Example: convert the captions to a format most players understand:
 ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt
 @end example
 
-@c man end INPUT DEVICES
+@c man end DEMUXERS

doc/developer.texi

@@ -147,29 +147,41 @@ GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
 @end itemize
 
 @subsection Naming conventions
-All names are using underscores (_), not CamelCase. For example, @samp{avfilter_get_video_buffer} is
-a valid function name and @samp{AVFilterGetVideo} is not. The exception from this are type names, like
+All names should be composed with underscores (_), not CamelCase. For example,
+@samp{avfilter_get_video_buffer} is an acceptable function name and
+@samp{AVFilterGetVideo} is not. The exception from this are type names, like
 for example structs and enums; they should always be in the CamelCase
 
-
-There are following conventions for naming variables and functions:
+There are the following conventions for naming variables and functions:
 @itemize @bullet
 @item
 For local variables no prefix is required.
 @item
-For variables and functions declared as @code{static} no prefixes are required.
+For variables and functions declared as @code{static} no prefix is required.
 @item
-For variables and functions used internally by the library, @code{ff_} prefix
-should be used.
-For example, @samp{ff_w64_demuxer}.
+For variables and functions used internally by a library an @code{ff_}
+prefix should be used, e.g. @samp{ff_w64_demuxer}.
 @item
 For variables and functions used internally across multiple libraries, use
 @code{avpriv_}. For example, @samp{avpriv_aac_parse_header}.
 @item
-For exported names, each library has its own prefixes. Just check the existing
-code and name accordingly.
+Each library has its own prefix for public symbols, in addition to the
+commonly used @code{av_} (@code{avformat_} for libavformat,
+@code{avcodec_} for libavcodec, @code{swr_} for libswresample, etc).
+Check the existing code and choose names accordingly.
+Note that some symbols without these prefixes are also exported for
+retro-compatibility reasons. These exceptions are declared in the
+@code{lib<name>/lib<name>.v} files.
 @end itemize
 
+Furthermore, name space reserved for the system should not be invaded.
+Identifiers ending in @code{_t} are reserved by
+@url{http://pubs.opengroup.org/onlinepubs/007904975/functions/xsh_chap02_02.html#tag_02_02_02, POSIX}.
+Also avoid names starting with @code{__} or @code{_} followed by an uppercase
+letter as they are reserved by the C standard. Names starting with @code{_}
+are reserved at the file level and may not be used for externally visible
+symbols. If in doubt, just avoid names starting with @code{_} altogether.
+
 @subsection Miscellaneous conventions
 @itemize @bullet
 @item
@@ -190,8 +202,8 @@ 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/
@@ -217,8 +229,13 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
 
 @enumerate
 @item
-   Contributions should be licensed under the LGPL 2.1, including an
-   "or any later version" clause, or the MIT license.  GPL 2 including
+   Contributions should be licensed under the
+   @uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
+   including an "or any later version" clause, or, if you prefer
+   a gift-style license, the
+   @uref{http://www.isc.org/software/license/, ISC} or
+   @uref{http://mit-license.org/, MIT} license.
+   @uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
    an "or any later version" clause is also acceptable, but LGPL is
    preferred.
 @item
@@ -228,6 +245,13 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
    You can commit unfinished stuff (for testing etc), but it must be disabled
    (#ifdef etc) by default so it does not interfere with other developers'
    work.
+@item
+   The commit message should have a short first line in the form of
+   a @samp{topic: short description} as a header, separated by a newline
+   from the body consisting of an explanation of why the change is necessary.
+   If the commit fixes a known bug on the bug tracker, the commit message
+   should include its bug ID. Referring to the issue on the bug tracker does
+   not exempt you from writing an excerpt of the bug in the commit message.
 @item
    You do not have to over-test things. If it works for you, and you think it
    should work for others, then commit. If your code has problems
@@ -334,8 +358,6 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
 
 We think our rules are not too hard. If you have comments, contact us.
 
-Note, these rules are mostly borrowed from the MPlayer project.
-
 @anchor{Submitting patches}
 @section Submitting patches
 
@@ -358,11 +380,6 @@ The tool is located in the tools directory.
 Run the @ref{Regression tests} before submitting a patch in order to verify
 it does not cause unexpected problems.
 
-Patches should be posted as base64 encoded attachments (or any other
-encoding which ensures that the patch will not be trashed during
-transmission) to the ffmpeg-devel mailing list, see
-@url{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel}
-
 It also helps quite a bit if you tell us what the patch does (for example
 'replaces lrint by lrintf'), and why (for example '*BSD isn't C99 compliant
 and has no lrint()')
@@ -370,6 +387,13 @@ and has no lrint()')
 Also please if you send several patches, send each patch as a separate mail,
 do not attach several unrelated patches to the same mail.
 
+Patches should be posted to the
+@uref{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
+mailing list. Use @code{git send-email} when possible since it will properly
+send patches without requiring extra care. If you cannot, then send patches
+as base64-encoded attachments, so your patch is not trashed during
+transmission.
+
 Your patch will be reviewed on the mailing list. You will likely be asked
 to make some changes and are expected to send in an improved version that
 incorporates the requests from the review. This process may go through
@@ -398,7 +422,7 @@ send a reminder by email. Your patch should eventually be dealt with.
     When adding new codec IDs, also add an entry to the codec descriptor
     list in @file{libavcodec/codec_desc.c}.
 @item
-    If it has a fourCC, did you add it to @file{libavformat/riff.c},
+    If it has a FourCC, did you add it to @file{libavformat/riff.c},
     even if it is only a decoder?
 @item
     Did you add a rule to compile the appropriate files in the Makefile?
@@ -451,8 +475,10 @@ send a reminder by email. Your patch should eventually be dealt with.
     other security issues?
 @item
     Did you test your decoder or demuxer against damaged data? If no, see
-    tools/trasher and the noise bitstream filter. Your decoder or demuxer
-    should not crash or end in a (near) infinite loop when fed damaged data.
+    tools/trasher, the noise bitstream filter, and
+    @uref{http://caca.zoy.org/wiki/zzuf, zzuf}. Your decoder or demuxer
+    should not crash, end in a (near) infinite loop, or allocate ridiculous
+    amounts of memory when fed damaged data.
 @item
     Does the patch not mix functional and cosmetic changes?
 @item
@@ -496,6 +522,9 @@ send a reminder by email. Your patch should eventually be dealt with.
     Make sure you check the return values of function and return appropriate
     error codes. Especially memory allocation functions like @code{av_malloc()}
     are notoriously left unchecked, which is a serious problem.
+@item
+    Test your code with valgrind and or Address Sanitizer to ensure it's free
+    of leaks, out of array accesses, etc.
 @end enumerate
 
 @section Patch review process

doc/encoders.texi

@@ -25,6 +25,95 @@ enabled encoders.
 A description of some of the currently available audio encoders
 follows.
 
+@anchor{aacenc}
+@section aac
+
+Advanced Audio Coding (AAC) encoder.
+
+This encoder is an experimental FFmpeg-native AAC encoder. Currently only the
+low complexity (AAC-LC) profile is supported. To use this encoder, you must set
+@option{strict} option to @samp{experimental} or lower.
+
+As this encoder is experimental, unexpected behavior may exist from time to
+time. For a more stable AAC encoder, see @ref{libvo-aacenc}. However, be warned
+that it has a worse quality reported by some users.
+
+@c Comment this out until somebody writes the respective documentation.
+@c See also @ref{libfaac}, @ref{libaacplus}, and @ref{libfdk-aac-enc}.
+
+@subsection Options
+
+@table @option
+@item b
+Set bit rate in bits/s. Setting this automatically activates constant bit rate
+(CBR) mode.
+
+@item q
+Set quality for variable bit rate (VBR) mode. This option is valid only using
+the @command{ffmpeg} command-line tool. For library interface users, use
+@option{global_quality}.
+
+@item stereo_mode
+Set stereo encoding mode. Possible values:
+
+@table @samp
+@item auto
+Automatically selected by the encoder.
+
+@item ms_off
+Disable middle/side encoding. This is the default.
+
+@item ms_force
+Force middle/side encoding.
+@end table
+
+@item aac_coder
+Set AAC encoder coding method. Possible values:
+
+@table @samp
+@item 0
+FAAC-inspired method.
+
+This method is a simplified reimplementation of the method used in FAAC, which
+sets thresholds proportional to the band energies, and then decreases all the
+thresholds with quantizer steps to find the appropriate quantization with
+distortion below threshold band by band.
+
+The quality of this method is comparable to the two loop searching method
+descibed below, but somewhat a little better and slower.
+
+@item 1
+Average noise to mask ratio (ANMR) trellis-based solution.
+
+This has a theoretic best quality out of all the coding methods, but at the
+cost of the slowest speed.
+
+@item 2
+Two loop searching (TLS) method.
+
+This method first sets quantizers depending on band thresholds and then tries
+to find an optimal combination by adding or subtracting a specific value from
+all quantizers and adjusting some individual quantizer a little.
+
+This method produces similar quality with the FAAC method and is the default.
+
+@item 3
+Constant quantizer method.
+
+This method sets a constant quantizer for all bands. This is the fastest of all
+the methods, yet produces the worst quality.
+
+@end table
+
+@end table
+
+@subsection Tips and Tricks
+
+According to some reports
+(e.g. @url{http://d.hatena.ne.jp/kamedo2/20120729/1343545890}), setting the
+@option{cutoff} option to 15000 Hz greatly improves the quality of the output
+quality. As a result, we encourage you to do the same.
+
 @section ac3 and ac3_fixed
 
 AC-3 audio encoders.
@@ -412,6 +501,279 @@ Selected by Encoder (default)
 
 @end table
 
+@section libmp3lame
+
+LAME (Lame Ain't an MP3 Encoder) MP3 encoder wrapper
+
+Requires the presence of the libmp3lame headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libmp3lame}.
+
+@subsection Options
+
+The following options are supported by the libmp3lame wrapper. The
+@command{lame}-equivalent of the options are listed in parentheses.
+
+@table @option
+@item b (@emph{-b})
+Set bitrate expressed in bits/s for CBR. LAME @code{bitrate} is
+expressed in kilobits/s.
+
+@item q (@emph{-V})
+Set constant quality setting for VBR. This option is valid only
+using the @command{ffmpeg} command-line tool. For library interface
+users, use @option{global_quality}.
+
+@item compression_level (@emph{-q})
+Set algorithm quality. Valid arguments are integers in the 0-9 range,
+with 0 meaning highest quality but slowest, and 9 meaning fastest
+while producing the worst quality.
+
+@item reservoir
+Enable use of bit reservoir when set to 1. Default value is 1. LAME
+has this enabled by default, but can be overriden by use
+@option{--nores} option.
+
+@end table
+
+@section libopencore-amrnb
+
+OpenCORE Adaptive Multi-Rate Narrowband encoder.
+
+Requires the presence of the libopencore-amrnb headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libopencore-amrnb --enable-version3}.
+
+This is a mono-only encoder. Officially it only supports 8000Hz sample rate,
+but you can override it by setting @option{strict} to @samp{unofficial} or
+lower.
+
+@subsection Options
+
+@table @option
+
+@item b
+Set bitrate in bits per second. Only the following bitrates are supported,
+otherwise libavcodec will round to the nearest valid bitrate.
+
+@table @option
+@item 4750
+@item 5150
+@item 5900
+@item 6700
+@item 7400
+@item 7950
+@item 10200
+@item 12200
+@end table
+
+@item dtx
+Allow discontinuous transmission (generate comfort noise) when set to 1. The
+default value is 0 (disabled).
+
+@end table
+
+@section libtwolame
+
+TwoLAME MP2 encoder wrapper
+
+Requires the presence of the libtwolame headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libtwolame}.
+
+@subsection Options
+
+The following options are supported by the libtwolame wrapper. The
+@command{twolame}-equivalent options follow the FFmpeg ones and are in
+parentheses.
+
+@table @option
+@item b (@emph{-b})
+Set bitrate expressed in bits/s for CBR. @command{twolame} @option{b}
+option is expressed in kilobits/s. Default value is 128k.
+
+@item q (@emph{-V})
+Set quality for experimental VBR support. Maximum value range is
+from -50 to 50, useful range is from -10 to 10. The higher the
+value, the better the quality. This option is valid only using the
+@command{ffmpeg} command-line tool. For library interface users,
+use @option{global_quality}.
+
+@item mode (@emph{--mode})
+Set the mode of the resulting audio. Possible values:
+
+@table @samp
+@item auto
+Choose mode automatically based on the input. This is the default.
+@item stereo
+Stereo
+@item joint_stereo
+Joint stereo
+@item dual_channel
+Dual channel
+@item mono
+Mono
+@end table
+
+@item psymodel (@emph{--psyc-mode})
+Set psychoacoustic model to use in encoding. The argument must be
+an integer between -1 and 4, inclusive. The higher the value, the
+better the quality. The default value is 3.
+
+@item energy_levels (@emph{--energy})
+Enable energy levels extensions when set to 1. The default value is
+0 (disabled).
+
+@item error_protection (@emph{--protect})
+Enable CRC error protection when set to 1. The default value is 0
+(disabled).
+
+@item copyright (@emph{--copyright})
+Set MPEG audio copyright flag when set to 1. The default value is 0
+(disabled).
+
+@item original (@emph{--original})
+Set MPEG audio original flag when set to 1. The default value is 0
+(disabled).
+
+@end table
+
+@anchor{libvo-aacenc}
+@section libvo-aacenc
+
+VisualOn AAC encoder
+
+Requires the presence of the libvo-aacenc headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libvo-aacenc --enable-version3}.
+
+This encoder is considered to be worse than the
+@ref{aacenc,,native experimental FFmpeg AAC encoder}, according to
+multiple sources.
+
+@subsection Options
+
+The VisualOn AAC encoder only support encoding AAC-LC and up to 2
+channels. It is also CBR-only.
+
+@table @option
+
+@item b
+Set bit rate in bits/s.
+
+@end table
+
+@section libvo-amrwbenc
+
+VisualOn Adaptive Multi-Rate Wideband encoder
+
+Requires the presence of the libvo-amrwbenc headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libvo-amrwbenc --enable-version3}.
+
+This is a mono-only encoder. Officially it only supports 16000Hz sample
+rate, but you can override it by setting @option{strict} to
+@samp{unofficial} or lower.
+
+@subsection Options
+
+@table @option
+
+@item b
+Set bitrate in bits/s. Only the following bitrates are supported, otherwise
+libavcodec will round to the nearest valid bitrate.
+
+@table @samp
+@item 6600
+@item 8850
+@item 12650
+@item 14250
+@item 15850
+@item 18250
+@item 19850
+@item 23050
+@item 23850
+@end table
+
+@item dtx
+Allow discontinuous transmission (generate comfort noise) when set to 1. The
+default value is 0 (disabled).
+
+@end table
+
+@section libopus
+
+libopus Opus Interactive Audio Codec encoder wrapper.
+
+Requires the presence of the libopus headers and library during
+configuration. You need to explicitly configure the build with
+@code{--enable-libopus}.
+
+@subsection Option Mapping
+
+Most libopus options are modeled after the @command{opusenc} utility from
+opus-tools. The following is an option mapping chart describing options
+supported by the libopus wrapper, and their @command{opusenc}-equivalent
+in parentheses.
+
+@table @option
+
+@item b (@emph{bitrate})
+Set the bit rate in bits/s.  FFmpeg's @option{b} option is
+expressed in bits/s, while @command{opusenc}'s @option{bitrate} in
+kilobits/s.
+
+@item vbr (@emph{vbr}, @emph{hard-cbr}, and @emph{cvbr})
+Set VBR mode. The FFmpeg @option{vbr} option has the following
+valid arguments, with the their @command{opusenc} equivalent options
+in parentheses:
+
+@table @samp
+@item off (@emph{hard-cbr})
+Use constant bit rate encoding.
+
+@item on (@emph{vbr})
+Use variable bit rate encoding (the default).
+
+@item constrained (@emph{cvbr})
+Use constrained variable bit rate encoding.
+@end table
+
+@item compression_level (@emph{comp})
+Set encoding algorithm complexity. Valid options are integers in
+the 0-10 range. 0 gives the fastest encodes but lower quality, while 10
+gives the highest quality but slowest encoding. The default is 10.
+
+@item frame_duration (@emph{framesize})
+Set maximum frame size, or duration of a frame in milliseconds. The
+argument must be exactly the following: 2.5, 5, 10, 20, 40, 60. Smaller
+frame sizes achieve lower latency but less quality at a given bitrate.
+Sizes greater than 20ms are only interesting at fairly low bitrates.
+The default is 20ms.
+
+@item packet_loss (@emph{expect-loss})
+Set expected packet loss percentage. The default is 0.
+
+@item application (N.A.)
+Set intended application type. Valid options are listed below:
+
+@table @samp
+@item voip
+Favor improved speech intelligibility.
+@item audio
+Favor faithfulness to the input (the default).
+@item lowdelay
+Restrict to only the lowest delay modes.
+@end table
+
+@item cutoff (N.A.)
+Set cutoff bandwidth in Hz. The argument must be exactly one of the
+following: 4000, 6000, 8000, 12000, or 20000, corresponding to
+narrowband, mediumband, wideband, super wideband, and fullband
+respectively. The default is 0 (cutoff disabled).
+
+@end table
+
 @c man end AUDIO ENCODERS
 
 @chapter Video Encoders
@@ -581,56 +943,458 @@ For more information about libvpx see:
 
 @section libx264
 
-H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 format supported through
-libx264.
+x264 H.264/MPEG-4 AVC encoder wrapper
 
-Requires the presence of the libx264 headers and library during
-configuration. You need to explicitly configure the build with
+This encoder requires the presence of the libx264 headers and library
+during configuration. You need to explicitly configure the build with
 @code{--enable-libx264}.
 
+libx264 supports an impressive number of features, including 8x8 and
+4x4 adaptive spatial transform, adaptive B-frame placement, CAVLC/CABAC
+entropy coding, interlacing (MBAFF), lossless mode, psy optimizations
+for detail retention (adaptive quantization, psy-RD, psy-trellis).
+
+Many libx264 encoder options are mapped to FFmpeg global codec
+options, while unique encoder options are provided through private
+options. Additionally the @option{x264opts} and @option{x264-params}
+private options allows to pass a list of key=value tuples as accepted
+by the libx264 @code{x264_param_parse} function.
+
+The x264 project website is at
+@url{http://www.videolan.org/developers/x264.html}.
+
 @subsection Options
 
-@table @option
+The following options are supported by the libx264 wrapper. The
+@command{x264}-equivalent options or values are listed in parentheses
+for easy migration.
 
-@item preset @var{preset_name}
+To reduce the duplication of documentation, only the private options
+and some others requiring special attention are documented here. For
+the documentation of the undocumented generic options, see
+@ref{codec-options,,the Codec Options chapter}.
+
+To get a more accurate and extensive documentation of the libx264
+options, invoke the command @command{x264 --full-help} or consult
+the libx264 documentation.
+
+@table @option
+@item b (@emph{bitrate})
+Set bitrate in bits/s. Note that FFmpeg's @option{b} option is
+expressed in bits/s, while @command{x264}'s @option{bitrate} is in
+kilobits/s.
+
+@item bf (@emph{bframes})
+
+@item g (@emph{keyint})
+
+@item qmax (@emph{qpmax})
+
+@item qmin (@emph{qpmin})
+
+@item qdiff (@emph{qpstep})
+
+@item qblur (@emph{qblur})
+
+@item qcomp (@emph{qcomp})
+
+@item refs (@emph{ref})
+
+@item sc_threshold (@emph{scenecut})
+
+@item trellis (@emph{trellis})
+
+@item nr  (@emph{nr})
+
+@item me_range (@emph{merange})
+
+@item me_method (@emph{me})
+Set motion estimation method. Possible values in the decreasing order
+of speed:
+
+@table @samp
+@item dia (@emph{dia})
+@item epzs (@emph{dia})
+Diamond search with radius 1 (fastest). @samp{epzs} is an alias for
+@samp{dia}.
+@item hex (@emph{hex})
+Hexagonal search with radius 2.
+@item umh (@emph{umh})
+Uneven multi-hexagon search.
+@item esa (@emph{esa})
+Exhaustive search.
+@item tesa (@emph{tesa})
+Hadamard exhaustive search (slowest).
+@end table
+
+@item subq (@emph{subme})
+
+@item b_strategy (@emph{b-adapt})
+
+@item keyint_min (@emph{min-keyint})
+
+@item coder
+Set entropy encoder. Possible values:
+
+@table @samp
+@item ac
+Enable CABAC.
+
+@item vlc
+Enable CAVLC and disable CABAC. It generates the same effect as
+@command{x264}'s @option{--no-cabac} option.
+@end table
+
+@item cmp
+Set full pixel motion estimation comparation algorithm. Possible values:
+
+@table @samp
+@item chroma
+Enable chroma in motion estimation.
+
+@item sad
+Ignore chroma in motion estimation. It generates the same effect as
+@command{x264}'s @option{--no-chroma-me} option.
+@end table
+
+@item threads (@emph{threads})
+
+@item thread_type
+Set multithreading technique. Possible values:
+
+@table @samp
+@item slice
+Slice-based multithreading. It generates the same effect as
+@command{x264}'s @option{--sliced-threads} option.
+@item frame
+Frame-based multithreading.
+@end table
+
+@item flags
+Set encoding flags. It can be used to disable closed GOP and enable
+open GOP by setting it to @code{-cgop}. The result is similar to
+the behavior of @command{x264}'s @option{--open-gop} option.
+
+@item rc_init_occupancy (@emph{vbv-init})
+
+@item preset (@emph{preset})
 Set the encoding preset.
 
-@item tune @var{tune_name}
-Tune the encoding params.
+@item tune (@emph{tune})
+Set tuning of the encoding params.
 
-@item fastfirstpass @var{bool}
-Use fast settings when encoding first pass, default value is 1.
-
-@item profile @var{profile_name}
+@item profile (@emph{profile})
 Set profile restrictions.
 
-@item level @var{level}
-Specify level (as defined by Annex A).
-Deprecated in favor of @var{x264opts}.
+@item fastfirstpass
+Enable fast settings when encoding first pass, when set to 1. When set
+to 0, it has the same effect of @command{x264}'s
+@option{--slow-firstpass} option.
 
-@item passlogfile @var{filename}
-Specify filename for 2 pass stats.
-Deprecated in favor of @var{x264opts} (see @var{stats} libx264 option).
+@item crf (@emph{crf})
+Set the quality for constant quality mode.
 
-@item wpredp @var{wpred_type}
-Specify Weighted prediction for P-frames.
-Deprecated in favor of @var{x264opts} (see @var{weightp} libx264 option).
+@item crf_max (@emph{crf-max})
+In CRF mode, prevents VBV from lowering quality beyond this point.
 
-@item x264opts @var{options}
-Allow to set any x264 option, see @code{x264 --fullhelp} for a list.
+@item qp (@emph{qp})
+Set constant quantization rate control method parameter.
 
-@var{options} is a list of @var{key}=@var{value} couples separated by
+@item aq-mode (@emph{aq-mode})
+Set AQ method. Possible values:
+
+@table @samp
+@item none (@emph{0})
+Disabled.
+
+@item variance (@emph{1})
+Variance AQ (complexity mask).
+
+@item autovariance (@emph{2})
+Auto-variance AQ (experimental).
+@end table
+
+@item aq-strength (@emph{aq-strength})
+Set AQ strength, reduce blocking and blurring in flat and textured areas.
+
+@item psy
+Use psychovisual optimizations when set to 1. When set to 0, it has the
+same effect as @command{x264}'s @option{--no-psy} option.
+
+@item psy-rd  (@emph{psy-rd})
+Set strength of psychovisual optimization, in
+@var{psy-rd}:@var{psy-trellis} format.
+
+@item rc-lookahead (@emph{rc-lookahead})
+Set number of frames to look ahead for frametype and ratecontrol.
+
+@item weightb
+Enable weighted prediction for B-frames when set to 1. When set to 0,
+it has the same effect as @command{x264}'s @option{--no-weightb} option.
+
+@item weightp (@emph{weightp})
+Set weighted prediction method for P-frames. Possible values:
+
+@table @samp
+@item none (@emph{0})
+Disabled
+@item simple (@emph{1})
+Enable only weighted refs
+@item smart (@emph{2})
+Enable both weighted refs and duplicates
+@end table
+
+@item ssim (@emph{ssim})
+Enable calculation and printing SSIM stats after the encoding.
+
+@item intra-refresh (@emph{intra-refresh})
+Enable the use of Periodic Intra Refresh instead of IDR frames when set
+to 1.
+
+@item b-bias (@emph{b-bias})
+Set the influence on how often B-frames are used.
+
+@item b-pyramid (@emph{b-pyramid})
+Set method for keeping of some B-frames as references. Possible values:
+
+@table @samp
+@item none (@emph{none})
+Disabled.
+@item strict (@emph{strict})
+Strictly hierarchical pyramid.
+@item normal (@emph{normal})
+Non-strict (not Blu-ray compatible).
+@end table
+
+@item mixed-refs
+Enable the use of one reference per partition, as opposed to one
+reference per macroblock when set to 1. When set to 0, it has the
+same effect as @command{x264}'s @option{--no-mixed-refs} option.
+
+@item 8x8dct
+Enable adaptive spatial transform (high profile 8x8 transform)
+when set to 1. When set to 0, it has the same effect as
+@command{x264}'s @option{--no-8x8dct} option.
+
+@item fast-pskip
+Enable early SKIP detection on P-frames when set to 1. When set
+to 0, it has the same effect as @command{x264}'s
+@option{--no-fast-pskip} option.
+
+@item aud (@emph{aud})
+Enable use of access unit delimiters when set to 1.
+
+@item mbtree
+Enable use macroblock tree ratecontrol when set to 1. When set
+to 0, it has the same effect as @command{x264}'s
+@option{--no-mbtree} option.
+
+@item deblock (@emph{deblock})
+Set loop filter parameters, in @var{alpha}:@var{beta} form.
+
+@item cplxblur (@emph{cplxblur})
+Set fluctuations reduction in QP (before curve compression).
+
+@item partitions (@emph{partitions})
+Set partitions to consider as a comma-separated list of. Possible
+values in the list:
+
+@table @samp
+@item p8x8
+8x8 P-frame partition.
+@item p4x4
+4x4 P-frame partition.
+@item b8x8
+4x4 B-frame partition.
+@item i8x8
+8x8 I-frame partition.
+@item i4x4
+4x4 I-frame partition.
+(Enabling @samp{p4x4} requires @samp{p8x8} to be enabled. Enabling
+@samp{i8x8} requires adaptive spatial transform (@option{8x8dct}
+option) to be enabled.)
+@item none (@emph{none})
+Do not consider any partitions.
+@item all (@emph{all})
+Consider every partition.
+@end table
+
+@item direct-pred (@emph{direct})
+Set direct MV prediction mode. Possible values:
+
+@table @samp
+@item none (@emph{none})
+Disable MV prediction.
+@item spatial (@emph{spatial})
+Enable spatial predicting.
+@item temporal (@emph{temporal})
+Enable temporal predicting.
+@item auto (@emph{auto})
+Automatically decided.
+@end table
+
+@item slice-max-size (@emph{slice-max-size})
+Set the limit of the size of each slice in bytes. If not specified
+but RTP payload size (@option{ps}) is specified, that is used.
+
+@item stats (@emph{stats})
+Set the file name for multi-pass stats.
+
+@item nal-hrd (@emph{nal-hrd})
+Set signal HRD information (requires @option{vbv-bufsize} to be set).
+Possible values:
+
+@table @samp
+@item none (@emph{none})
+Disable HRD information signaling.
+@item vbr (@emph{vbr})
+Variable bit rate.
+@item cbr (@emph{cbr})
+Constant bit rate (not allowed in MP4 container).
+@end table
+
+@item x264opts (N.A.)
+Set any x264 option, see @command{x264 --fullhelp} for a list.
+
+Argument is a list of @var{key}=@var{value} couples separated by
 ":". In @var{filter} and @var{psy-rd} options that use ":" as a separator
 themselves, use "," instead. They accept it as well since long ago but this
 is kept undocumented for some reason.
-@end table
 
 For example to specify libx264 encoding options with @command{ffmpeg}:
 @example
 ffmpeg -i foo.mpg -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv
 @end example
 
-For more information about libx264 and the supported options see:
-@url{http://www.videolan.org/developers/x264.html}
+@item x264-params (N.A.)
+Override the x264 configuration using a :-separated list of key=value
+parameters.
+
+This option is functionally the same as the @option{x264opts}, but is
+duplicated for compability with the Libav fork.
+
+For example to specify libx264 encoding options with @command{ffmpeg}:
+@example
+ffmpeg -i INPUT -c:v libx264 -x264-params level=30:bframes=0:weightp=0:\
+cabac=0:ref=1:vbv-maxrate=768:vbv-bufsize=2000:analyse=all:me=umh:\
+no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 OUTPUT
+@end example
+@end table
+
+Encoding ffpresets for common usages are provided so they can be used with the
+general presets system (e.g. passing the @option{pre} option).
+
+@section libxvid
+
+Xvid MPEG-4 Part 2 encoder wrapper.
+
+This encoder requires the presence of the libxvidcore headers and library
+during configuration. You need to explicitly configure the build with
+@code{--enable-libxvid --enable-gpl}.
+
+The native @code{mpeg4} encoder supports the MPEG-4 Part 2 format, so
+users can encode to this format without this library.
+
+@subsection Options
+
+The following options are supported by the libxvid wrapper. Some of
+the following options are listed but are not documented, and
+correspond to shared codec options. See @ref{codec-options,,the Codec
+Options chapter} for their documentation. The other shared options
+which are not listed have no effect for the libxvid encoder.
+
+@table @option
+@item b
+
+@item g
+
+@item qmin
+
+@item qmax
+
+@item mpeg_quant
+
+@item threads
+
+@item bf
+
+@item b_qfactor
+
+@item b_qoffset
+
+@item flags
+Set specific encoding flags. Possible values:
+
+@table @samp
+
+@item mv4
+Use four motion vector by macroblock.
+
+@item aic
+Enable high quality AC prediction.
+
+@item gray
+Only encode grayscale.
+
+@item gmc
+Enable the use of global motion compensation (GMC).
+
+@item qpel
+Enable quarter-pixel motion compensation.
+
+@item cgop
+Enable closed GOP.
+
+@item global_header
+Place global headers in extradata instead of every keyframe.
+
+@end table
+
+@item trellis
+
+@item me_method
+Set motion estimation method. Possible values in decreasing order of
+speed and increasing order of quality:
+
+@table @samp
+@item zero
+Use no motion estimation (default).
+
+@item phods
+@item x1
+@item log
+Enable advanced diamond zonal search for 16x16 blocks and half-pixel
+refinement for 16x16 blocks. @samp{x1} and @samp{log} are aliases for
+@samp{phods}.
+
+@item epzs
+Enable all of the things described above, plus advanced diamond zonal
+search for 8x8 blocks, half-pixel refinement for 8x8 blocks, and motion
+estimation on chroma planes.
+
+@item full
+Enable all of the things described above, plus extended 16x16 and 8x8
+blocks search.
+@end table
+
+@item mbd
+Set macroblock decision algorithm. Possible values in the increasing
+order of quality:
+
+@table @samp
+@item simple
+Use macroblock comparing function algorithm (default).
+
+@item bits
+Enable rate distortion-based half pixel and quarter pixel refinement for
+16x16 blocks.
+
+@item rd
+Enable all of the things described above, plus rate distortion-based
+half pixel and quarter pixel refinement for 8x8 blocks, and rate
+distortion-based search using square pattern.
+@end table
+
+@end table
 
 @c man end VIDEO ENCODERS

doc/eval.texi

@@ -20,57 +20,93 @@ The following unary operators are available: @code{+}, @code{-}.
 
 The following functions are available:
 @table @option
-@item sinh(x)
-Compute hyperbolic sine of @var{x}.
-
-@item cosh(x)
-Compute hyperbolic cosine of @var{x}.
-
-@item tanh(x)
-Compute hyperbolic tangent of @var{x}.
-
-@item sin(x)
-Compute sine of @var{x}.
-
-@item cos(x)
-Compute cosine of @var{x}.
-
-@item tan(x)
-Compute tangent of @var{x}.
-
-@item atan(x)
-Compute arctangent of @var{x}.
-
-@item asin(x)
-Compute arcsine of @var{x}.
+@item abs(x)
+Compute absolute value of @var{x}.
 
 @item acos(x)
 Compute arccosine of @var{x}.
 
+@item asin(x)
+Compute arcsine of @var{x}.
+
+@item atan(x)
+Compute arctangent of @var{x}.
+
+@item ceil(expr)
+Round the value of expression @var{expr} upwards to the nearest
+integer. For example, "ceil(1.5)" is "2.0".
+
+@item cos(x)
+Compute cosine of @var{x}.
+
+@item cosh(x)
+Compute hyperbolic cosine of @var{x}.
+
+@item eq(x, y)
+Return 1 if @var{x} and @var{y} are equivalent, 0 otherwise.
+
 @item exp(x)
 Compute exponential of @var{x} (with base @code{e}, the Euler's number).
 
-@item log(x)
-Compute natural logarithm of @var{x}.
-
-@item abs(x)
-Compute absolute value of @var{x}.
-
-@item squish(x)
-Compute expression @code{1/(1 + exp(4*x))}.
+@item floor(expr)
+Round the value of expression @var{expr} downwards to the nearest
+integer. For example, "floor(-1.5)" is "-2.0".
 
 @item gauss(x)
 Compute Gauss function of @var{x}, corresponding to
 @code{exp(-x*x/2) / sqrt(2*PI)}.
 
+@item gcd(x, y)
+Return the greatest common divisor of @var{x} and @var{y}. If both @var{x} and
+@var{y} are 0 or either or both are less than zero then behavior is undefined.
+
+@item gt(x, y)
+Return 1 if @var{x} is greater than @var{y}, 0 otherwise.
+
+@item gte(x, y)
+Return 1 if @var{x} is greater than or equal to @var{y}, 0 otherwise.
+
+@item hypot(x, y)
+This function is similar to the C function with the same name; it returns
+"sqrt(@var{x}*@var{x} + @var{y}*@var{y})", the length of the hypotenuse of a
+right triangle with sides of length @var{x} and @var{y}, or the distance of the
+point (@var{x}, @var{y}) from the origin.
+
+@item if(x, y)
+Evaluate @var{x}, and if the result is non-zero return the result of
+the evaluation of @var{y}, return 0 otherwise.
+
+@item if(x, y, z)
+Evaluate @var{x}, and if the result is non-zero return the evaluation
+result of @var{y}, otherwise the evaluation result of @var{z}.
+
+@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 ifnot(x, y, z)
+Evaluate @var{x}, and if the result is zero return the evaluation
+result of @var{y}, otherwise the evaluation result of @var{z}.
+
 @item isinf(x)
 Return 1.0 if @var{x} is +/-INFINITY, 0.0 otherwise.
 
 @item isnan(x)
 Return 1.0 if @var{x} is NAN, 0.0 otherwise.
 
-@item mod(x, y)
-Compute the remainder of division of @var{x} by @var{y}.
+@item ld(var)
+Allow to load the value of the internal variable with number
+@var{var}, which was previously stored with st(@var{var}, @var{expr}).
+The function returns the loaded value.
+
+@item log(x)
+Compute natural logarithm of @var{x}.
+
+@item lt(x, y)
+Return 1 if @var{x} is lesser than @var{y}, 0 otherwise.
+
+@item lte(x, y)
+Return 1 if @var{x} is lesser than or equal to @var{y}, 0 otherwise.
 
 @item max(x, y)
 Return the maximum between @var{x} and @var{y}.
@@ -78,53 +114,8 @@ Return the maximum between @var{x} and @var{y}.
 @item min(x, y)
 Return the maximum between @var{x} and @var{y}.
 
-@item eq(x, y)
-Return 1 if @var{x} and @var{y} are equivalent, 0 otherwise.
-
-@item gte(x, y)
-Return 1 if @var{x} is greater than or equal to @var{y}, 0 otherwise.
-
-@item gt(x, y)
-Return 1 if @var{x} is greater than @var{y}, 0 otherwise.
-
-@item lte(x, y)
-Return 1 if @var{x} is lesser than or equal to @var{y}, 0 otherwise.
-
-@item lt(x, y)
-Return 1 if @var{x} is lesser than @var{y}, 0 otherwise.
-
-@item st(var, expr)
-Allow to store the value of the expression @var{expr} in an internal
-variable. @var{var} specifies the number of the variable where to
-store the value, and it is a value ranging from 0 to 9. The function
-returns the value stored in the internal variable.
-Note, Variables are currently not shared between expressions.
-
-@item ld(var)
-Allow to load the value of the internal variable with number
-@var{var}, which was previously stored with st(@var{var}, @var{expr}).
-The function returns the loaded value.
-
-@item while(cond, expr)
-Evaluate expression @var{expr} while the expression @var{cond} is
-non-zero, and returns the value of the last @var{expr} evaluation, or
-NAN if @var{cond} was always false.
-
-@item ceil(expr)
-Round the value of expression @var{expr} upwards to the nearest
-integer. For example, "ceil(1.5)" is "2.0".
-
-@item floor(expr)
-Round the value of expression @var{expr} downwards to the nearest
-integer. For example, "floor(-1.5)" is "-2.0".
-
-@item trunc(expr)
-Round the value of expression @var{expr} towards zero to the nearest
-integer. For example, "trunc(-1.5)" is "-1.0".
-
-@item sqrt(expr)
-Compute the square root of @var{expr}. This is equivalent to
-"(@var{expr})^.5".
+@item mod(x, y)
+Compute the remainder of division of @var{x} by @var{y}.
 
 @item not(expr)
 Return 1.0 if @var{expr} is zero, 0.0 otherwise.
@@ -133,39 +124,83 @@ Return 1.0 if @var{expr} is zero, 0.0 otherwise.
 Compute the power of @var{x} elevated @var{y}, it is equivalent to
 "(@var{x})^(@var{y})".
 
+@item print(t)
+@item print(t, l)
+Print the value of expression @var{t} with loglevel @var{l}. If
+@var{l} is not specified then a default log level is used.
+Returns the value of the expression printed.
+
+Prints t with loglevel l
+
 @item random(x)
 Return a pseudo random value between 0.0 and 1.0. @var{x} is the index of the
 internal variable which will be used to save the seed/state.
 
-@item hypot(x, y)
-This function is similar to the C function with the same name; it returns
-"sqrt(@var{x}*@var{x} + @var{y}*@var{y})", the length of the hypotenuse of a
-right triangle with sides of length @var{x} and @var{y}, or the distance of the
-point (@var{x}, @var{y}) from the origin.
-
-@item gcd(x, y)
-Return the greatest common divisor of @var{x} and @var{y}. If both @var{x} and
-@var{y} are 0 or either or both are less than zero then behavior is undefined.
-
-@item if(x, y)
-Evaluate @var{x}, and if the result is non-zero return the result of
-the evaluation of @var{y}, return 0 otherwise.
-
-@item ifnot(x, y)
-Evaluate @var{x}, and if the result is zero return the result of the
-evaluation of @var{y}, return 0 otherwise.
-
-@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.
+Find an input value for which the function represented by @var{expr}
+with argument @var{ld(0)} is 0 in the interval 0..@var{max}.
+
+The expression in @var{expr} must denote a continuous function or the
+result is undefined.
+
+@var{ld(0)} is used to represent the function input value, which means
+that the given expression will be evaluated multiple times with
+various input values that the expression can access through
+@code{ld(0)}. When the expression evaluates to 0 then the
+corresponding input value will be returned.
+
+@item sin(x)
+Compute sine of @var{x}.
+
+@item sinh(x)
+Compute hyperbolic sine of @var{x}.
+
+@item sqrt(expr)
+Compute the square root of @var{expr}. This is equivalent to
+"(@var{expr})^.5".
+
+@item squish(x)
+Compute expression @code{1/(1 + exp(4*x))}.
+
+@item st(var, expr)
+Allow to store the value of the expression @var{expr} in an internal
+variable. @var{var} specifies the number of the variable where to
+store the value, and it is a value ranging from 0 to 9. The function
+returns the value stored in the internal variable.
+Note, Variables are currently not shared between expressions.
+
+@item tan(x)
+Compute tangent of @var{x}.
+
+@item tanh(x)
+Compute hyperbolic tangent of @var{x}.
+
+@item taylor(expr, x)
+@item taylor(expr, x, id)
+Evaluate a Taylor series at @var{x}, given an expression representing
+the @code{ld(id)}-th derivative of a function at 0.
+
+When the series does not converge the result is undefined.
+
+@var{ld(id)} is used to represent the derivative order in @var{expr},
+which means that the given expression will be evaluated multiple times
+with various input values that the expression can access through
+@code{ld(id)}. If @var{id} is not specified then 0 is assumed.
+
+Note, when you have the derivatives at y instead of 0,
+@code{taylor(expr, x-y)} can be used.
+
+@item time(0)
+Return the current (wallclock) time in seconds.
+
+@item trunc(expr)
+Round the value of expression @var{expr} towards zero to the nearest
+integer. For example, "trunc(-1.5)" is "-1.0".
+
+@item while(cond, expr)
+Evaluate expression @var{expr} while the expression @var{cond} is
+non-zero, and returns the value of the last @var{expr} evaluation, or
+NAN if @var{cond} was always false.
 @end table
 
 The following constants are available:
@@ -185,68 +220,69 @@ value, note that:
 
 @code{+} works like OR
 
-and the construct:
+For example the construct:
 @example
-if A then B else C
+if (A AND B) then C
 @end example
-is equivalent to
+is equivalent to:
 @example
-if(A,B) + ifnot(A,C)
+if(A*B, C)
 @end example
 
 In your C code, you can extend the list of unary and binary functions,
 and define recognized constants, so that they are available for your
 expressions.
 
-The evaluator also recognizes the International System number
-postfixes. If 'i' is appended after the postfix, powers of 2 are used
-instead of powers of 10. The 'B' postfix multiplies the value for 8,
-and can be appended after another postfix or used alone. This allows
-using for example 'KB', 'MiB', 'G' and 'B' as postfix.
+The evaluator also recognizes the International System unit prefixes.
+If 'i' is appended after the prefix, binary prefixes are used, which
+are based on powers of 1024 instead of powers of 1000.
+The 'B' postfix multiplies the value by 8, and can be appended after a
+unit prefix or used alone. This allows using for example 'KB', 'MiB',
+'G' and 'B' as number postfix.
 
-Follows the list of available International System postfixes, with
+The list of available International System prefixes follows, with
 indication of the corresponding powers of 10 and of 2.
 @table @option
 @item y
--24 / -80
+10^-24 / 2^-80
 @item z
--21 / -70
+10^-21 / 2^-70
 @item a
--18 / -60
+10^-18 / 2^-60
 @item f
--15 / -50
+10^-15 / 2^-50
 @item p
--12 / -40
+10^-12 / 2^-40
 @item n
--9 / -30
+10^-9 / 2^-30
 @item u
--6 / -20
+10^-6 / 2^-20
 @item m
--3 / -10
+10^-3 / 2^-10
 @item c
--2
+10^-2
 @item d
--1
+10^-1
 @item h
-2
+10^2
 @item k
-3 / 10
+10^3 / 2^10
 @item K
-3 / 10
+10^3 / 2^10
 @item M
-6 / 20
+10^6 / 2^20
 @item G
-9 / 30
+10^9 / 2^30
 @item T
-12 / 40
+10^12 / 2^40
 @item P
-15 / 40
+10^15 / 2^40
 @item E
-18 / 50
+10^18 / 2^50
 @item Z
-21 / 60
+10^21 / 2^60
 @item Y
-24 / 70
+10^24 / 2^70
 @end table
 
 @c man end

doc/examples/demuxing.c

@@ -98,7 +98,7 @@ static int decode_packet(int *got_frame, int cached)
                    audio_frame_count++, frame->nb_samples,
                    av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
 
-            ret = av_samples_alloc(audio_dst_data, &audio_dst_linesize, frame->channels,
+            ret = av_samples_alloc(audio_dst_data, &audio_dst_linesize, av_frame_get_channels(frame),
                                    frame->nb_samples, frame->format, 1);
             if (ret < 0) {
                 fprintf(stderr, "Could not allocate audio buffer\n");
@@ -107,13 +107,13 @@ static int decode_packet(int *got_frame, int cached)
 
             /* TODO: extend return code of the av_samples_* functions so that this call is not needed */
             audio_dst_bufsize =
-                av_samples_get_buffer_size(NULL, frame->channels,
+                av_samples_get_buffer_size(NULL, av_frame_get_channels(frame),
                                            frame->nb_samples, frame->format, 1);
 
             /* copy audio data to destination buffer:
              * this is required since rawaudio expects non aligned data */
             av_samples_copy(audio_dst_data, frame->data, 0, 0,
-                            frame->nb_samples, frame->channels, frame->format);
+                            frame->nb_samples, av_frame_get_channels(frame), frame->format);
 
             /* write to rawaudio file */
             fwrite(audio_dst_data[0], 1, audio_dst_bufsize, audio_dst_file);
@@ -292,8 +292,10 @@ int main (int argc, char **argv)
         printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);
 
     /* read frames from the file */
-    while (av_read_frame(fmt_ctx, &pkt) >= 0)
+    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
         decode_packet(&got_frame, 0);
+        av_free_packet(&pkt);
+    }
 
     /* flush cached frames */
     pkt.data = NULL;
@@ -314,7 +316,7 @@ int main (int argc, char **argv)
     if (audio_stream) {
         const char *fmt;
 
-        if ((ret = get_format_from_sample_fmt(&fmt, audio_dec_ctx->sample_fmt) < 0))
+        if ((ret = get_format_from_sample_fmt(&fmt, audio_dec_ctx->sample_fmt)) < 0)
             goto end;
         printf("Play the output audio file with the command:\n"
                "ffplay -f %s -ac %d -ar %d %s\n",

doc/examples/filtering_audio.c

@@ -169,9 +169,13 @@ int main(int argc, char **argv)
 {
     int ret;
     AVPacket packet;
-    AVFrame frame;
+    AVFrame *frame = avcodec_alloc_frame();
     int got_frame;
 
+    if (!frame) {
+        perror("Could not allocate frame");
+        exit(1);
+    }
     if (argc != 2) {
         fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
         exit(1);
@@ -193,9 +197,9 @@ int main(int argc, char **argv)
             break;
 
         if (packet.stream_index == audio_stream_index) {
-            avcodec_get_frame_defaults(&frame);
+            avcodec_get_frame_defaults(frame);
             got_frame = 0;
-            ret = avcodec_decode_audio4(dec_ctx, &frame, &got_frame, &packet);
+            ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
             if (ret < 0) {
                 av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
                 continue;
@@ -203,7 +207,7 @@ int main(int argc, char **argv)
 
             if (got_frame) {
                 /* push the audio data from decoded frame into the filtergraph */
-                if (av_buffersrc_add_frame(buffersrc_ctx, &frame, 0) < 0) {
+                if (av_buffersrc_add_frame(buffersrc_ctx, frame, 0) < 0) {
                     av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
                     break;
                 }
@@ -229,6 +233,7 @@ end:
     if (dec_ctx)
         avcodec_close(dec_ctx);
     avformat_close_input(&fmt_ctx);
+    av_freep(&frame);
 
     if (ret < 0 && ret != AVERROR_EOF) {
         char buf[1024];

doc/examples/filtering_video.c

@@ -173,9 +173,13 @@ int main(int argc, char **argv)
 {
     int ret;
     AVPacket packet;
-    AVFrame frame;
+    AVFrame *frame = avcodec_alloc_frame();
     int got_frame;
 
+    if (!frame) {
+        perror("Could not allocate frame");
+        exit(1);
+    }
     if (argc != 2) {
         fprintf(stderr, "Usage: %s file\n", argv[0]);
         exit(1);
@@ -197,19 +201,19 @@ int main(int argc, char **argv)
             break;
 
         if (packet.stream_index == video_stream_index) {
-            avcodec_get_frame_defaults(&frame);
+            avcodec_get_frame_defaults(frame);
             got_frame = 0;
-            ret = avcodec_decode_video2(dec_ctx, &frame, &got_frame, &packet);
+            ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet);
             if (ret < 0) {
                 av_log(NULL, AV_LOG_ERROR, "Error decoding video\n");
                 break;
             }
 
             if (got_frame) {
-                frame.pts = av_frame_get_best_effort_timestamp(&frame);
+                frame->pts = av_frame_get_best_effort_timestamp(frame);
 
                 /* push the decoded frame into the filtergraph */
-                if (av_buffersrc_add_frame(buffersrc_ctx, &frame, 0) < 0) {
+                if (av_buffersrc_add_frame(buffersrc_ctx, frame, 0) < 0) {
                     av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
                     break;
                 }
@@ -236,6 +240,7 @@ end:
     if (dec_ctx)
         avcodec_close(dec_ctx);
     avformat_close_input(&fmt_ctx);
+    av_freep(&frame);
 
     if (ret < 0 && ret != AVERROR_EOF) {
         char buf[1024];

doc/examples/muxing.c

@@ -86,7 +86,6 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
         break;
 
     case AVMEDIA_TYPE_VIDEO:
-        avcodec_get_context_defaults3(c, *codec);
         c->codec_id = codec_id;
 
         c->bit_rate = 400000;
@@ -396,7 +395,7 @@ int main(int argc, char **argv)
     AVStream *audio_st, *video_st;
     AVCodec *audio_codec, *video_codec;
     double audio_pts, video_pts;
-    int ret, i;
+    int ret;
 
     /* Initialize libavcodec, and register all codecs and formats. */
     av_register_all();
@@ -504,18 +503,12 @@ int main(int argc, char **argv)
     if (audio_st)
         close_audio(oc, audio_st);
 
-    /* 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. */
         avio_close(oc->pb);
 
     /* free the stream */
-    av_free(oc);
+    avformat_free_context(oc);
 
     return 0;
 }

doc/examples/resampling_audio.c

@@ -200,7 +200,7 @@ int main(int argc, char **argv)
         fwrite(dst_data[0], 1, dst_bufsize, dst_file);
     } while (t < 10);
 
-    if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt) < 0))
+    if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
         goto end;
     fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
             "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",

doc/examples/scaling_video.c

@@ -107,7 +107,7 @@ int main(int argc, char **argv)
         goto end;
     }
 
-    /* buffer is going to be written to rawvideo file, no alignmnet */
+    /* buffer is going to be written to rawvideo file, no alignment */
     if ((ret = av_image_alloc(dst_data, dst_linesize,
                               dst_w, dst_h, dst_pix_fmt, 1)) < 0) {
         fprintf(stderr, "Could not allocate destination image\n");

doc/faq.texi

@@ -294,8 +294,12 @@ your format doesn't support file level concatenation.
 
 @subsection Concatenating using the concat @emph{protocol} (file level)
 
+FFmpeg has a @url{http://ffmpeg.org/ffmpeg-protocols.html#concat,
+@code{concat}} protocol designed specifically for that, with examples in the
+documentation.
+
 A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate
-video by merely concatenating the files them.
+video by merely concatenating the files containing them.
 
 Hence you may concatenate your multimedia files by first transcoding them to
 these privileged formats, then using the humble @code{cat} command (or the

doc/ffmpeg-codecs.texi

@@ -17,6 +17,7 @@ the libavcodec library.
 
 @c man end DESCRIPTION
 
+@anchor{codec-options}
 @chapter Codec Options
 @c man begin CODEC OPTIONS
 
@@ -1098,6 +1099,9 @@ Set sample format audio decoders should prefer. Default value is
 @code{none}.
 
 @item pkt_timebase @var{rational number}
+
+@item sub_charenc @var{encoding} (@emph{decoding,subtitles})
+Set the input subtitles character encoding.
 @end table
 
 @c man end CODEC OPTIONS

doc/ffmpeg-formats.texi

@@ -76,8 +76,13 @@ Enable RTP MP4A-LATM payload.
 Reduce the latency introduced by optional buffering
 @end table
 
+@item seek2any @var{integer} (@emph{input})
+Forces seeking to enable seek to any mode if set to 1. Default is 0.
+
 @item analyzeduration @var{integer} (@emph{input})
-Specify how many microseconds are analyzed to estimate duration.
+Specify how many microseconds are analyzed to probe the input. A
+higher value will allow to detect more accurate information, but will
+increase latency. It defaults to 5,000,000 microseconds = 5 seconds.
 
 @item cryptokey @var{hexadecimal string} (@emph{input})
 Set decryption key.
@@ -140,6 +145,12 @@ Use wallclock as timestamps.
 @item avoid_negative_ts @var{integer} (@emph{output})
 Shift timestamps to make them positive. 1 enables, 0 disables, default
 of -1 enables when required by target format.
+
+@item skip_initial_bytes @var{integer} (@emph{input})
+Set number initial bytes to skip. Default is 0.
+
+@item correct_ts_overflow @var{integer} (@emph{input})
+Correct single timestamp overflows if set to 1. Default is 1.
 @end table
 
 @c man end FORMAT OPTIONS

doc/ffmpeg-resampler.texi

@@ -43,7 +43,7 @@ value is not mandatory if the corresponding channel layout
 @option{out_channel_layout} is set.
 
 @item uch, used_channel_count
-Set the number of used channels. Default value is 0. This option is
+Set the number of used input channels. Default value is 0. This option is
 only used for special remapping.
 
 @item isr, in_sample_rate
@@ -60,6 +60,7 @@ Specify the output sample format. It is set by default to @code{none}.
 
 @item tsf, internal_sample_fmt
 Set the internal sample format. Default value is @code{none}.
+This will automatically be chosen when it is not explicitly set.
 
 @item icl, in_channel_layout
 Set the input channel layout.
@@ -68,15 +69,17 @@ Set the input channel layout.
 Set the output channel layout.
 
 @item clev, center_mix_level
-Set center mix level. It is a value expressed in deciBel, and must be
-inclusively included between -32 and +32.
+Set the center mix level. It is a value expressed in deciBel, and must be
+in the interval [-32,32].
 
 @item slev, surround_mix_level
-Set surround mix level. It is a value expressed in deciBel, and must
-be inclusively included between -32 and +32.
+Set the surround mix level. It is a value expressed in deciBel, and must
+be in the interval [-32,32].
 
-@item lfe_mix_evel
-Set LFE mix level.
+@item lfe_mix_level
+Set LFE mix into non LFE level. It is used when there is a LFE input but no
+LFE output. It is a value expressed in deciBel, and must
+be in the interval [-32,32].
 
 @item rmvol, rematrix_volume
 Set rematrix volume. Default value is 1.0.
@@ -87,7 +90,8 @@ Set flags used by the converter. Default value is 0.
 It supports the following individual flags:
 @table @option
 @item res
-force resampling
+force resampling, this flag forces resampling to be used even when the
+input and output sample rates match.
 @end table
 
 @item dither_scale
@@ -104,6 +108,21 @@ select rectangular dither
 select triangular dither
 @item triangular_hp
 select triangular dither with high pass
+@item lipshitz
+select lipshitz noise shaping dither
+@item shibata
+select shibata noise shaping dither
+@item low_shibata
+select low shibata noise shaping dither
+@item high_shibata
+select high shibata noise shaping dither
+@item f_weighted
+select f-weighted noise shaping dither
+@item modified_e_weighted
+select modified-e-weighted noise shaping dither
+@item improved_e_weighted
+select improved-e-weighted noise shaping dither
+
 @end table
 
 @item resampler
@@ -124,8 +143,8 @@ case.
 For swr only, set resampling filter size, default value is 32.
 
 @item phase_shift
-For swr only, set resampling phase shift, default value is 10, must be included
-between 0 and 30.
+For swr only, set resampling phase shift, default value is 10, and must be in
+the interval [0,30].
 
 @item linear_interp
 Use Linear Interpolation if set to 1, default value is 0.
@@ -153,6 +172,14 @@ may be stretched or squeezed for each second.
 Default value is 0, thus no compensation is applied to make the samples match
 the audio timestamps.
 
+@item first_pts
+For swr only, assume the first pts should be this value. The time unit is 1 / sample rate.
+This allows for padding/trimming at the start of stream. By default, no
+assumption is made about the first frame's expected pts, so no padding or
+trimming is done. For example, this could be set to 0 to pad the beginning with
+silence if an audio stream starts after the video stream or to trim any samples
+with a negative pts due to encoder delay.
+
 @item min_comp
 For swr only, set the minimum difference between timestamps and audio data (in
 seconds) to trigger stretching/squeezing/filling or trimming of the
@@ -208,8 +235,8 @@ select Kaiser Windowed Sinc
 @end table
 
 @item kaiser_beta
-For swr only, set Kaiser Window Beta value. Must be an integer included between
-2 and 16, default value is 9.
+For swr only, set Kaiser Window Beta value. Must be an integer in the
+interval [2,16], default value is 9.
 
 @end table
 

doc/ffmpeg.texi

@@ -256,6 +256,14 @@ libx264, and the 138th audio, which will be encoded with libvorbis.
 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.
 
+-to and -t are mutually exclusive and -t has priority.
+
+@item -to @var{position} (@emph{output})
+Stop writing the output at @var{position}.
+@var{position} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.
+
+-to and -t are mutually exclusive and -t has priority.
+
 @item -fs @var{limit_size} (@emph{output})
 Set the file size limit, expressed in bytes.
 
@@ -334,13 +342,21 @@ Stop writing to the stream after @var{framecount} frames.
 Use fixed quality scale (VBR). The meaning of @var{q} is
 codec-dependent.
 
+@anchor{filter_option}
 @item -filter[:@var{stream_specifier}] @var{filter_graph} (@emph{output,per-stream})
-@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).
+Create the filter graph specified by @var{filter_graph} and use it to
+filter the stream.
+
+@var{filter_graph} is a description of the filter graph to apply to
+the stream, and must have a single input and a single output of the
+same type of the stream. In the filter graph, the input is associated
+to the label @code{in}, and the output to the label @code{out}. See
+the ffmpeg-filters manual for more information about the filtergraph
+syntax.
+
+See the @ref{filter_complex_option,,-filter_complex option} if you
+want to create filter graphs with multiple inputs and/or outputs.
 
-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).
 
@@ -395,11 +411,11 @@ will be used.
 
 E.g. to extract the first attachment to a file named 'out.ttf':
 @example
-ffmpeg -dump_attachment:t:0 out.ttf INPUT
+ffmpeg -dump_attachment:t:0 out.ttf -i INPUT
 @end example
 To extract all attachments to files determined by the @code{filename} tag:
 @example
-ffmpeg -dump_attachment:t "" INPUT
+ffmpeg -dump_attachment:t "" -i INPUT
 @end example
 
 Technical note -- attachments are implemented as codec extradata, so this
@@ -443,21 +459,6 @@ form @var{num}:@var{den}, where @var{num} and @var{den} are the
 numerator and denominator of the aspect ratio. For example "4:3",
 "16:9", "1.3333", and "1.7777" are valid argument values.
 
-@item -croptop @var{size}
-@item -cropbottom @var{size}
-@item -cropleft @var{size}
-@item -cropright @var{size}
-All the crop options have been removed. Use -vf
-crop=width:height:x:y instead.
-
-@item -padtop @var{size}
-@item -padbottom @var{size}
-@item -padleft @var{size}
-@item -padright @var{size}
-@item -padcolor @var{hex_color}
-All the pad options have been removed. Use -vf
-pad=width:height:x:y:color instead.
-
 @item -vn (@emph{output})
 Disable video recording.
 
@@ -487,11 +488,10 @@ stream
 Set the ISO 639 language code (3 letters) of the current video stream.
 
 @item -vf @var{filter_graph} (@emph{output})
-@var{filter_graph} is a description of the filter graph to apply to
-the input video.
-Use the option "-filters" to show all the available filters (including
-also sources and sinks).  This is an alias for @code{-filter:v}.
+Create the filter graph specified by @var{filter_graph} and use it to
+filter the stream.
 
+This is an alias for @code{-filter:v}, see the @ref{filter_option,,-filter option}.
 @end table
 
 @section Advanced Video Options
@@ -545,12 +545,58 @@ Force video tag/fourcc. This is an alias for @code{-tag:v}.
 Show QP histogram
 @item -vbsf @var{bitstream_filter}
 Deprecated see -bsf
+
 @item -force_key_frames[:@var{stream_specifier}] @var{time}[,@var{time}...] (@emph{output,per-stream})
+@item -force_key_frames[:@var{stream_specifier}] expr:@var{expr} (@emph{output,per-stream})
 Force key frames at the specified timestamps, more precisely at the first
 frames after each specified time.
+
+If the argument is prefixed with @code{expr:}, the string @var{expr}
+is interpreted like an expression and is evaluated for each frame. A
+key frame is forced in case the evaluation is non-zero.
+
+If one of the times is "@code{chapters}[@var{delta}]", it is expanded into
+the time of the beginning of all chapters in the file, shifted by
+@var{delta}, expressed as a time in seconds.
 This option can be useful to ensure that a seek point is present at a
 chapter mark or any other designated place in the output file.
-The timestamps must be specified in ascending order.
+
+For example, to insert a key frame at 5 minutes, plus key frames 0.1 second
+before the beginning of every chapter:
+@example
+-force_key_frames 0:05:00,chapters-0.1
+@end example
+
+The expression in @var{expr} can contain the following constants:
+@table @option
+@item n
+the number of current processed frame, starting from 0
+@item n_forced
+the number of forced frames
+@item prev_forced_n
+the number of the previous forced frame, it is @code{NAN} when no
+keyframe was forced yet
+@item prev_forced_t
+the time of the previous forced frame, it is @code{NAN} when no
+keyframe was forced yet
+@item t
+the time of the current processed frame
+@end table
+
+For example to force a key frame every 5 seconds, you can specify:
+@example
+-force_key_frames expr:gte(t,n_forced*5)
+@end example
+
+To force a key frame 5 seconds after the time of the last forced one,
+starting from second 13:
+@example
+-force_key_frames expr:if(isnan(prev_forced_t),gte(t,13),gte(t,prev_forced_t+5))
+@end example
+
+Note that forcing too many keyframes is very harmful for the lookahead
+algorithms of certain encoders: using fixed-GOP options or similar
+would be more efficient.
 
 @item -copyinkf[:@var{stream_specifier}] (@emph{output,per-stream})
 When doing stream copy, copy also non-key frames found at the
@@ -581,11 +627,12 @@ 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}.
+Create the filter graph specified by @var{filter_graph} and use it to
+filter the stream.
+
+This is an alias for @code{-filter:a}, see the @ref{filter_option,,-filter option}.
 @end table
 
 @section Advanced Audio options:
@@ -595,6 +642,12 @@ also sources and sinks).  This is an alias for @code{-filter:a}.
 Force audio tag/fourcc. This is an alias for @code{-tag:a}.
 @item -absf @var{bitstream_filter}
 Deprecated, see -bsf
+@item -guess_layout_max @var{channels} (@emph{input,per-stream})
+If some input channel layout is not known, try to guess only if it
+corresponds to at most the specified number of channels. For example, 2
+tells to @command{ffmpeg} to recognize 1 channel as mono and 2 channels as
+stereo but not 6 channels as 5.1. The default is to always try to guess. Use
+0 to disable all guessing.
 @end table
 
 @section Subtitle options:
@@ -928,6 +981,7 @@ Specify Timecode for writing. @var{SEP} is ':' for non drop timecode and ';'
 ffmpeg -i input.mpg -timecode 01:02:03.04 -r 30000/1001 -s ntsc output.mpg
 @end example
 
+@anchor{filter_complex_option}
 @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
@@ -972,13 +1026,13 @@ ffmpeg -i video.mkv -i image.png -filter_complex 'overlay' out.mkv
 
 To generate 5 seconds of pure red video using lavfi @code{color} source:
 @example
-ffmpeg -filter_complex 'color=red' -t 5 out.mkv
+ffmpeg -filter_complex 'color=c=red' -t 5 out.mkv
 @end example
 @end table
 
 As a special exception, you can use a bitmap subtitle stream as input: it
 will be converted into a video with the same size as the largest video in
-the file, or 720×576 if no video is present. Note that this is an
+the file, or 720x576 if no video is present. Note that this is an
 experimental and temporary solution. It will be removed once libavfilter has
 proper support for subtitles.
 
@@ -1284,7 +1338,7 @@ ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
 @url{ffmpeg-scaler.html,ffmpeg-scaler},
 @url{ffmpeg-resampler.html,ffmpeg-resampler},
 @url{ffmpeg-codecs.html,ffmpeg-codecs},
-@url{ffmpeg-bitstream-filters,ffmpeg-bitstream-filters},
+@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
 @url{ffmpeg-formats.html,ffmpeg-formats},
 @url{ffmpeg-devices.html,ffmpeg-devices},
 @url{ffmpeg-protocols.html,ffmpeg-protocols},

doc/ffplay.texi

@@ -74,10 +74,15 @@ You can interactively cycle through the available show modes by
 pressing the key @key{w}.
 
 @item -vf @var{filter_graph}
+Create the filter graph specified by @var{filter_graph} and use it to
+filter the video stream.
+
 @var{filter_graph} is a description of the filter graph to apply to
-the input video.
-Use the option "-filters" to show all the available filters (including
-also sources and sinks).
+the stream, and must have a single video input and a single video
+output. In the filter graph, the input is associated to the label
+@code{in}, and the output to the label @code{out}. See the
+ffmpeg-filters manual for more information about the filtergraph
+syntax.
 
 @item -i @var{input_file}
 Read @var{input_file}.
@@ -194,7 +199,7 @@ Seek to percentage in file corresponding to fraction of width.
 @url{ffmpeg-scaler.html,ffmpeg-scaler},
 @url{ffmpeg-resampler.html,ffmpeg-resampler},
 @url{ffmpeg-codecs.html,ffmpeg-codecs},
-@url{ffmpeg-bitstream-filters,ffmpeg-bitstream-filters},
+@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
 @url{ffmpeg-formats.html,ffmpeg-formats},
 @url{ffmpeg-devices.html,ffmpeg-devices},
 @url{ffmpeg-protocols.html,ffmpeg-protocols},

doc/ffprobe.texi

@@ -495,7 +495,7 @@ DV, GXF and AVI timecodes are available in format metadata
 @url{ffmpeg-scaler.html,ffmpeg-scaler},
 @url{ffmpeg-resampler.html,ffmpeg-resampler},
 @url{ffmpeg-codecs.html,ffmpeg-codecs},
-@url{ffmpeg-bitstream-filters,ffmpeg-bitstream-filters},
+@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
 @url{ffmpeg-formats.html,ffmpeg-formats},
 @url{ffmpeg-devices.html,ffmpeg-devices},
 @url{ffmpeg-protocols.html,ffmpeg-protocols},

doc/ffserver.texi

@@ -16,34 +16,28 @@ ffserver [@var{options}]
 @chapter Description
 @c man begin DESCRIPTION
 
-ffserver is a streaming server for both audio and video. It supports
+@command{ffserver} is a streaming server for both audio and video. It
+supports several live feeds, streaming from files and time shifting on
+live feeds (you can seek to positions in the past on each live feed,
+provided you specify a big enough feed storage in
+@file{ffserver.conf}).
 
-several live feeds, streaming from files and time shifting on live feeds
-(you can seek to positions in the past on each live feed, provided you
-specify a big enough feed storage in ffserver.conf).
+@command{ffserver} receives prerecorded files or FFM streams from some
+@command{ffmpeg} instance as input, then streams them over
+RTP/RTSP/HTTP.
 
-This documentation covers only the streaming aspects of ffserver /
-ffmpeg. All questions about parameters for ffmpeg, codec questions,
-etc. are not covered here. Read @file{ffmpeg.html} for more
-information.
+An @command{ffserver} instance will listen on some port as specified
+in the configuration file. You can launch one or more instances of
+@command{ffmpeg} and send one or more FFM streams to the port where
+ffserver is expecting to receive them. Alternately, you can make
+@command{ffserver} launch such @command{ffmpeg} instances at startup.
 
-@section How does it work?
-
-ffserver receives prerecorded files or FFM streams from some ffmpeg
-instance as input, then streams them over RTP/RTSP/HTTP.
-
-An ffserver instance will listen on some port as specified in the
-configuration file. You can launch one or more instances of ffmpeg and
-send one or more FFM streams to the port where ffserver is expecting
-to receive them. Alternately, you can make ffserver launch such ffmpeg
-instances at startup.
-
-Input streams are called feeds, and each one is specified by a <Feed>
-section in the configuration file.
+Input streams are called feeds, and each one is specified by a
+@code{<Feed>} section in the configuration file.
 
 For each feed you can have different output streams in various
-formats, each one specified by a <Stream> section in the configuration
-file.
+formats, each one specified by a @code{<Stream>} section in the
+configuration file.
 
 @section Status stream
 
@@ -79,14 +73,6 @@ web server can be used to serve up the files just as well.
 It can stream prerecorded video from .ffm files, though it is somewhat tricky
 to make it work correctly.
 
-@section What do I need?
-
-I use Linux on a 900 MHz Duron with a cheap Bt848 based TV capture card. I'm
-using stock Linux 2.4.17 with the stock drivers. [Actually that isn't true,
-I needed some special drivers for my motherboard-based sound card.]
-
-I understand that FreeBSD systems work just fine as well.
-
 @section How do I make it work?
 
 First, build the kit. It *really* helps to have installed LAME first. Then when
@@ -235,7 +221,7 @@ of an infinite movie or a whole movie.
 
 FFM is version specific, and there is limited compatibility of FFM files
 generated by one version of ffmpeg/ffserver and another version of
-ffmpeg/ffserver. It may work but its not guaranteed to work.
+ffmpeg/ffserver. It may work but it is not guaranteed to work.
 
 FFM2 is extensible while maintaining compatibility and should work between
 differing versions of tools. FFM2 is the default.
@@ -269,7 +255,7 @@ The @file{doc/ffserver.conf} example,
 @url{ffmpeg-scaler.html,ffmpeg-scaler},
 @url{ffmpeg-resampler.html,ffmpeg-resampler},
 @url{ffmpeg-codecs.html,ffmpeg-codecs},
-@url{ffmpeg-bitstream-filters,ffmpeg-bitstream-filters},
+@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
 @url{ffmpeg-formats.html,ffmpeg-formats},
 @url{ffmpeg-devices.html,ffmpeg-devices},
 @url{ffmpeg-protocols.html,ffmpeg-protocols},

doc/filter_design.txt

@@ -15,13 +15,13 @@ Format negotiation
   the list of 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).
+  channel layout, sample format (the sample packing is implied by the sample
+  format) and sample rate.
 
   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
+  supported at each end 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.
 
@@ -68,15 +68,15 @@ Buffer references ownership and permissions
 
     Here are the (fairly obvious) rules for reference ownership:
 
-    * A reference received by the start_frame or filter_frame method
-      belong to the corresponding filter.
+    * A reference received by the filter_frame method (or its start_frame
+      deprecated version) belongs to the corresponding filter.
 
       Special exception: for video references: the reference may be used
       internally for automatic copying and must not be destroyed before
       end_frame; it can be given away to ff_start_frame.
 
-    * A reference passed to ff_start_frame or ff_filter_frame is given
-      away and must no longer be used.
+    * A reference passed to ff_filter_frame (or the deprecated
+      ff_start_frame) is given away and must no longer be used.
 
     * A reference created with avfilter_ref_buffer belongs to the code that
       created it.
@@ -90,27 +90,11 @@ Buffer references ownership and permissions
   Link reference fields
   ---------------------
 
-    The AVFilterLink structure has a few AVFilterBufferRef fields. Here are
-    the rules to handle them:
-
-    * cur_buf is set before the start_frame and filter_frame methods to
-      the same reference given as argument to the methods and belongs to the
-      destination filter of the link. If it has not been cleared after
-      end_frame or filter_frame, libavfilter will automatically destroy
-      the reference; therefore, any filter that needs to keep the reference
-      for longer must set cur_buf to NULL.
-
-    * out_buf belongs to the source filter of the link and can be used to
-      store a reference to the buffer that has been sent to the destination.
-      If it is not NULL after end_frame or filter_frame, libavfilter will
-      automatically destroy the reference.
-
-      If a video input pad does not have a start_frame method, the default
-      method will request a buffer on the first output of the filter, store
-      the reference in out_buf and push a second reference to the output.
-
-    * src_buf, cur_buf_copy and partial_buf are used by libavfilter
-      internally and must not be accessed by filters.
+    The AVFilterLink structure has a few AVFilterBufferRef fields. The
+    cur_buf and out_buf were used with the deprecated
+    start_frame/draw_slice/end_frame API and should no longer be used.
+    src_buf, cur_buf_copy and partial_buf are used by libavfilter internally
+    and must not be accessed by filters.
 
   Reference permissions
   ---------------------
@@ -119,8 +103,10 @@ Buffer references ownership and permissions
     the code that owns the reference is allowed to do to the buffer data.
     Different references for the same buffer can have different permissions.
 
-    For video filters, the permissions only apply to the parts of the buffer
-    that have already been covered by the draw_slice method.
+    For video filters that implement the deprecated
+    start_frame/draw_slice/end_frame API, the permissions only apply to the
+    parts of the buffer that have already been covered by the draw_slice
+    method.
 
     The value is a binary OR of the following constants:
 
@@ -179,9 +165,9 @@ Buffer references ownership and permissions
       with the WRITE permission.
 
     * Filters that intend to keep a reference after the filtering process
-      is finished (after end_frame or filter_frame returns) must have the
-      PRESERVE permission on it and remove the WRITE permission if they
-      create a new reference to give it away.
+      is finished (after filter_frame returns) must have the PRESERVE
+      permission on it and remove the WRITE permission if they create a new
+      reference to give it away.
 
     * Filters that intend to modify a reference they have kept after the end
       of the filtering process need the REUSE2 permission and must remove
@@ -198,11 +184,11 @@ Frame scheduling
   Simple filters that output one frame for each input frame should not have
   to worry about it.
 
-  start_frame / filter_frame
-  ----------------------------
+  filter_frame
+  ------------
 
-    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.
+    This method is called when a frame is pushed to the filter's input. It
+    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.
@@ -213,7 +199,7 @@ Frame scheduling
     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
+    (Example: framerate-doubling filter: filter_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.)
 
@@ -233,8 +219,8 @@ Frame scheduling
 
     This method is called when a frame is wanted on an output.
 
-    For an input, it should directly call start_frame or filter_frame on
-    the corresponding output.
+    For an input, it should directly call filter_frame 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
@@ -255,7 +241,7 @@ Frame scheduling
         }
         while (!frame_pushed) {
             input = input_where_a_frame_is_most_needed();
-            ret = avfilter_request_frame(input);
+            ret = ff_request_frame(input);
             if (ret == AVERROR_EOF) {
                 process_eof_on_input();
             } else if (ret < 0) {
@@ -266,4 +252,14 @@ Frame scheduling
 
     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_frame method will be called and do the work.
+    the filter_frame method will be called and do the work.
+
+Legacy API
+==========
+
+  Until libavfilter 3.23, the filter_frame method was split:
+
+  - for video filters, it was made of start_frame, draw_slice (that could be
+    called several times on distinct parts of the frame) and end_frame;
+
+  - for audio filters, it was called filter_samples.

doc/general.texi

@@ -24,7 +24,7 @@ instructions.  To enable using OpenJPEG in FFmpeg, pass @code{--enable-libopenjp
 @file{./configure}.
 
 
-@section OpenCORE and VisualOn libraries
+@section OpenCORE, VisualOn, and Fraunhofer libraries
 
 Spun off Google Android sources, OpenCore, VisualOn and Fraunhofer
 libraries provide encoders for a number of audio codecs.
@@ -32,9 +32,14 @@ libraries provide encoders for a number of audio codecs.
 @float NOTE
 OpenCORE and VisualOn libraries are under the Apache License 2.0
 (see @url{http://www.apache.org/licenses/LICENSE-2.0} for details), which is
-incompatible with the LGPL version 2.1 and GPL version 2. You have to
+incompatible to the LGPL version 2.1 and GPL version 2. You have to
 upgrade FFmpeg's license to LGPL version 3 (or if you have enabled
-GPL components, GPL version 3) to use it.
+GPL components, GPL version 3) by passing @code{--enable-version3} to configure in
+order to use it.
+
+The Fraunhofer AAC library is licensed under a license incompatible to the GPL
+and is not known to be compatible to the LGPL. Therefore, you have to pass
+@code{--enable-nonfree} to configure to use it.
 @end float
 
 @subsection OpenCORE AMR
@@ -814,6 +819,7 @@ following image formats are supported:
 @item DSP Group TrueSpeech   @tab     @tab  X
 @item DV audio               @tab     @tab  X
 @item Enhanced AC-3          @tab  X  @tab  X
+@item EVRC (Enhanced Variable Rate Codec) @tab     @tab  X
 @item FLAC (Free Lossless Audio Codec)  @tab  X  @tab  IX
 @item G.723.1                @tab X @tab X
 @item G.729                  @tab     @tab  X

doc/indevs.texi

@@ -583,10 +583,16 @@ command:
 ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
 @end example
 
-@section video4linux2
+@section video4linux2, v4l2
 
 Video4Linux2 input video device.
 
+"v4l2" can be used as alias for "video4linux2".
+
+If FFmpeg is built with v4l-utils support (by using the
+@code{--enable-libv4l2} configure option), the device will always rely
+on libv4l2.
+
 The name of the device to grab is a file device node, usually Linux
 systems tend to automatically create such nodes when the device
 (e.g. an USB webcam) is plugged into the system, and has a name of the
@@ -596,8 +602,8 @@ the device.
 Video4Linux2 devices usually 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.
-
-Some usage examples of the video4linux2 devices with ffmpeg and ffplay:
+Some devices, like TV cards, support one or more standards. It is possible
+to list all the supported standards using @command{-list_standards all}.
 
 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
@@ -606,19 +612,93 @@ 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.
 
-Note that if FFmpeg is build with v4l-utils support ("--enable-libv4l2"
-option), it will always be used.
+Some usage examples of the video4linux2 device with @command{ffmpeg}
+and @command{ffplay}:
+@itemize
+@item
+Grab and show the input of a video4linux2 device:
 @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
 @end example
 
-"v4l" and "v4l2" can be used as aliases for the respective "video4linux" and
-"video4linux2".
+@item
+Grab and record the input of a video4linux2 device, leave the
+framerate and size as previously set:
+@example
+ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg
+@end example
+@end itemize
+
+For more information about Video4Linux, check @url{http://linuxtv.org/}.
+
+@subsection Options
+
+@table @option
+@item standard
+Set the standard. Must be the name of a supported standard. To get a
+list of the supported standards, use the @option{list_standards}
+option.
+
+@item channel
+Set the input channel number. Default to 0.
+
+@item video_size
+Set the video frame size. The argument must be a string in the form
+@var{WIDTH}x@var{HEIGHT} or a valid size abbreviation.
+
+@item pixel_format
+Select the pixel format (only valid for raw video input).
+
+@item input_format
+Set the preferred pixel format (for raw video) or a codec name.
+This option allows to select the input format, when several are
+available.
+
+@item framerate
+Set the preferred video framerate.
+
+@item list_formats
+List available formats (supported pixel formats, codecs, and frame
+sizes) and exit.
+
+Available values are:
+@table @samp
+@item all
+Show all available (compressed and non-compressed) formats.
+
+@item raw
+Show only raw video (non-compressed) formats.
+
+@item compressed
+Show only compressed formats.
+@end table
+
+@item list_standards
+List supported standards and exit.
+
+Available values are:
+@table @samp
+@item all
+Show all supported standards.
+@end table
+
+@item timestamps, ts
+Set type of timestamps for grabbed frames.
+
+Available values are:
+@table @samp
+@item default
+Use timestamps from the kernel.
+
+@item abs
+Use absolute timestamps (wall clock).
+
+@item mono2abs
+Force conversion from monotonic to absolute timestamps.
+@end table
+
+Default value is @code{default}.
+@end table
 
 @section vfwcap
 

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

@@ -18,6 +18,23 @@ enabled muxers.
 
 A description of some of the currently available muxers follows.
 
+@anchor{aiff}
+@section aiff
+
+Audio Interchange File Format muxer.
+
+It accepts the following options:
+
+@table @option
+@item write_id3v2
+Enable ID3v2 tags writing when set to 1. Default is 0 (disabled).
+
+@item id3v2_version
+Select ID3v2 version to write. Currently only version 3 and 4 (aka.
+ID3v2.3 and ID3v2.4) are supported. The default is version 4.
+
+@end table
+
 @anchor{crc}
 @section crc
 
@@ -241,8 +258,13 @@ ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg
 @end example
 
 @table @option
-@item -start_number @var{number}
-Start the sequence from @var{number}.
+@item start_number @var{number}
+Start the sequence from @var{number}. Default value is 1. Must be a
+positive number.
+
+@item updatefirst 1|0
+If set to 1, update the first written image file again and
+again. Default value is 0.
 @end table
 
 The image muxer supports the .Y.U.V image file format. This format is
@@ -346,6 +368,8 @@ This option is implicitly set when writing ismv (Smooth Streaming) files.
 Run a second pass moving the moov atom on top of the file. This
 operation can take a while, and will not work in various situations such
 as fragmented output, thus it is not enabled by default.
+@item -movflags rtphint
+Add RTP hinting tracks to the output file.
 @end table
 
 Smooth Streaming content can be pushed in real time to a publishing
@@ -521,9 +545,11 @@ reference stream. The default value is ``auto''.
 @item segment_format @var{format}
 Override the inner container format, by default it is guessed by the filename
 extension.
+
 @item segment_list @var{name}
 Generate also a listfile named @var{name}. If not specified no
 listfile is generated.
+
 @item segment_list_flags @var{flags}
 Set flags affecting the segment list generation.
 
@@ -534,17 +560,15 @@ Allow caching (only affects M3U8 list files).
 
 @item live
 Allow live-friendly file generation.
-
-This currently only affects M3U8 lists. In particular, write a fake
-EXT-X-TARGETDURATION duration field at the top of the file, based on
-the specified @var{segment_time}.
 @end table
 
 Default value is @code{cache}.
 
 @item segment_list_size @var{size}
-Overwrite the listfile once it reaches @var{size} entries. If 0
-the listfile is never overwritten. Default value is 0.
+Update the list file so that it contains at most the last @var{size}
+segments. If 0 the list file will contain all the segments. Default
+value is 0.
+
 @item segment_list type @var{type}
 Specify the format for the segment list file.
 
@@ -572,19 +596,34 @@ auto-select this format.
 
 @code{ext} is deprecated in favor or @code{csv}.
 
+@item ffconcat
+Generate an ffconcat file for the created segments. The resulting file
+can be read using the FFmpeg @ref{concat} demuxer.
+
+A list file with the suffix @code{".ffcat"} or @code{".ffconcat"} will
+auto-select this format.
+
 @item m3u8
-Generate an extended M3U8 file, version 4, compliant with
-@url{http://tools.ietf.org/id/draft-pantos-http-live-streaming-08.txt}.
+Generate an extended M3U8 file, version 3, compliant with
+@url{http://tools.ietf.org/id/draft-pantos-http-live-streaming}.
 
 A list file with the suffix @code{".m3u8"} will auto-select this format.
 @end table
 
 If not specified the type is guessed from the list file name suffix.
+
 @item segment_time @var{time}
-Set segment duration to @var{time}. Default value is "2".
+Set segment duration to @var{time}, the value must be a duration
+specification. Default value is "2". See also the
+@option{segment_times} option.
+
+Note that splitting may not be accurate, unless you force the
+reference stream key-frames at the given time. See the introductory
+notice and the examples below.
+
 @item segment_time_delta @var{delta}
 Specify the accuracy time when selecting the start time for a
-segment. Default value is "0".
+segment, expressed as a duration specification. Default value is "0".
 
 When delta is specified a key-frame will start a new segment if its
 PTS satisfies the relation:
@@ -606,7 +645,8 @@ the specified time and the time set by @var{force_key_frames}.
 
 @item segment_times @var{times}
 Specify a list of split points. @var{times} contains a list of comma
-separated duration specifications, in increasing order.
+separated duration specifications, in increasing order. See also
+the @option{segment_time} option.
 
 @item segment_frames @var{frames}
 Specify a list of split video frame numbers. @var{frames} contains a
@@ -629,7 +669,7 @@ of the generated segments. May not work with some combinations of
 muxers/codecs. It is set to @code{0} by default.
 @end table
 
-@section Examples
+@subsection Examples
 
 @itemize
 @item
@@ -711,10 +751,61 @@ Write an mp3 with an ID3v2.3 header and an ID3v1 footer:
 ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3
 @end example
 
-Attach a picture to an mp3:
+To attach a picture to an mp3 file select both the audio and the picture stream
+with @code{map}:
 @example
-ffmpeg -i input.mp3 -i cover.png -c copy -metadata:s:v title="Album cover"
--metadata:s:v comment="Cover (Front)" out.mp3
+ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1
+-metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3
 @end example
 
+@section ogg
+
+Ogg container muxer.
+
+@table @option
+@item -page_duration @var{duration}
+Preferred page duration, in microseconds. The muxer will attempt to create
+pages that are approximately @var{duration} microseconds long. This allows the
+user to compromise between seek granularity and container overhead. The default
+is 1 second. A value of 0 will fill all segments, making pages as large as
+possible. A value of 1 will effectively use 1 packet-per-page in most
+situations, giving a small seek granularity at the cost of additional container
+overhead.
+@end table
+
+@section tee
+
+The tee muxer can be used to write the same data to several files or any
+other kind of muxer. It can be used, for example, to both stream a video to
+the network and save it to disk at the same time.
+
+It is different from specifying several outputs to the @command{ffmpeg}
+command-line tool because the audio and video data will be encoded only once
+with the tee muxer; encoding can be a very expensive process. It is not
+useful when using the libavformat API directly because it is then possible
+to feed the same packets to several muxers directly.
+
+The slave outputs are specified in the file name given to the muxer,
+separated by '|'. If any of the slave name contains the '|' separator,
+leading or trailing spaces or any special character, it must be
+escaped (see the ``Quoting and escaping'' section in the ffmpeg-utils
+manual).
+
+Options can be specified for each slave by prepending them as a list of
+@var{key}=@var{value} pairs separated by ':', between square brackets. If
+the options values contain a special character or the ':' separator, they
+must be escaped; note that this is a second level escaping.
+
+Example: encode something and both archive it in a WebM file and stream it
+as MPEG-TS over UDP (the streams need to be explicitly mapped):
+
+@example
+ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
+  "archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"
+@end example
+
+Note: some codecs may need different options depending on the output format;
+the auto-detection of this can not work with the tee muxer. The main example
+is the @option{global_header} flag.
+
 @c man end MUXERS

doc/optimization.txt

@@ -148,7 +148,7 @@ Alignment:
 Some instructions on some architectures have strict alignment restrictions,
 for example most SSE/SSE2 instructions on x86.
 The minimum guaranteed alignment is written in the .h files, for example:
-    void (*put_pixels_clamped)(const DCTELEM *block/*align 16*/, UINT8 *pixels/*align 8*/, int line_size);
+    void (*put_pixels_clamped)(const int16_t *block/*align 16*/, UINT8 *pixels/*align 8*/, int line_size);
 
 
 General Tips:

doc/platform.texi

@@ -114,7 +114,7 @@ wrapper.
 You will need the following prerequisites:
 
 @itemize
-@item @uref{https://github.com/libav/c99-to-c89/, C99-to-C89 Converter & Wrapper}
+@item @uref{http://download.videolan.org/pub/contrib/c99-to-c89/, C99-to-C89 Converter & Wrapper}
 @item @uref{http://code.google.com/p/msinttypes/, msinttypes}
 @item @uref{http://www.mingw.org/, MSYS}
 @item @uref{http://yasm.tortall.net/, YASM}

doc/protocols.texi

@@ -128,6 +128,63 @@ m3u8 files.
 
 HTTP (Hyper Text Transfer Protocol).
 
+This protocol accepts the following options.
+
+@table @option
+@item seekable
+Control seekability of connection. If set to 1 the resource is
+supposed to be seekable, if set to 0 it is assumed not to be seekable,
+if set to -1 it will try to autodetect if it is seekable. Default
+value is -1.
+
+@item chunked_post
+If set to 1 use chunked transfer-encoding for posts, default is 1.
+
+@item headers
+Set custom HTTP headers, can override built in default headers. The
+value must be a string encoding the headers.
+
+@item content_type
+Force a content type.
+
+@item user-agent
+Override User-Agent header. If not specified the protocol will use a
+string describing the libavformat build.
+
+@item multiple_requests
+Use persistent connections if set to 1. By default it is 0.
+
+@item post_data
+Set custom HTTP post data.
+
+@item timeout
+Set timeout of socket I/O operations used by the underlying low level
+operation. By default it is set to -1, which means that the timeout is
+not specified.
+
+@item mime_type
+Set MIME type.
+
+@item cookies
+Set the cookies to be sent in future requests. The format of each cookie is the
+same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
+delimited by a newline character.
+@end table
+
+@subsection HTTP Cookies
+
+Some HTTP requests will be denied unless cookie values are passed in with the
+request. The @option{cookies} option allows these cookies to be specified. At
+the very least, each cookie must specify a value along with a path and domain.
+HTTP requests that match both the domain and path will automatically include the
+cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
+by a newline.
+
+The required syntax to play a stream specifying a cookie is:
+@example
+ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
+@end example
+
 @section mmst
 
 MMS (Microsoft Media Server) protocol over TCP.

doc/swresample.txt

@@ -32,9 +32,9 @@ Special Converter         v
                         Output
 
 Planar/Packed conversion is done when needed during sample format conversion.
-Every step can be skipped without memcpy when its not needed.
+Every step can be skipped without memcpy when it is not needed.
 Either Resampling and Rematrixing can be performed first depending on which
-way its faster.
+way it is faster.
 The Buffers are needed for resampling due to resamplng being a process that
 requires future and past data, it thus also introduces inevitably a delay when
 used.

doc/syntax.texi

@@ -98,7 +98,7 @@ year-month-day.
 
 The accepted syntax is:
 @example
-[-]HH:MM:SS[.m...]
+[-][HH:]MM:SS[.m...]
 [-]S+[.m...]
 @end example
 
@@ -112,6 +112,22 @@ Specify the size of the sourced video, it may be a string of the form
 
 The following abbreviations are recognized:
 @table @samp
+@item ntsc
+720x480
+@item pal
+720x576
+@item qntsc
+352x240
+@item qpal
+352x288
+@item sntsc
+640x480
+@item spal
+768x576
+@item film
+352x240
+@item ntsc-film
+352x240
 @item sqcif
 128x96
 @item qcif
@@ -170,6 +186,18 @@ The following abbreviations are recognized:
 1280x720
 @item hd1080
 1920x1080
+@item 2k
+2048x1080
+@item 2kflat
+1998x1080
+@item 2kscope
+2048x858
+@item 4k
+4096x2160
+@item 4kflat
+3996x2160
+@item 4kscope
+4096x1716
 @end table
 
 @anchor{video rate syntax}
@@ -187,17 +215,17 @@ The following abbreviations are recognized:
 @item pal
 25/1
 @item qntsc
-30000/1
+30000/1001
 @item qpal
 25/1
 @item sntsc
-30000/1
+30000/1001
 @item spal
 25/1
 @item film
 24/1
 @item ntsc-film
-24000/1
+24000/1001
 @end table
 
 @anchor{ratio syntax}

doc/t2h.init

@@ -68,7 +68,7 @@ $print_page_head = \&FFmpeg_print_page_head;
 sub FFmpeg_print_page_head($$)
 {
     my $fh = shift;
-    my $longtitle = "$Texi2HTML::THISDOC{'title_no_texi'}";
+    my $longtitle = "$Texi2HTML::THISDOC{'fulltitle_no_texi'}";
     $longtitle .= ": $Texi2HTML::NO_TEXI{'This'}" if exists $Texi2HTML::NO_TEXI{'This'};
     my $description = $DOCUMENT_DESCRIPTION;
     $description = $longtitle if (!defined($description));

doc/texi2pod.pl

@@ -169,7 +169,7 @@ INF: while(<$inf>) {
         } elsif ($ended =~ /^(?:example|smallexample|display)$/) {
             $shift = "";
             $_ = "";        # need a paragraph break
-        } elsif ($ended =~ /^(?:itemize|enumerate|[fv]?table)$/) {
+        } elsif ($ended =~ /^(?:itemize|enumerate|(?:multi|[fv])?table)$/) {
             $_ = "\n=back\n";
             $ic = pop @icstack;
         } else {
@@ -269,7 +269,7 @@ INF: while(<$inf>) {
         $endw = "enumerate";
     };
 
-    /^\@([fv]?table)\s+(\@[a-z]+)/ and do {
+    /^\@((?:multi|[fv])?table)\s+(\@[a-z]+)/ and do {
         push @endwstack, $endw;
         push @icstack, $ic;
         $endw = $1;
@@ -278,6 +278,7 @@ INF: while(<$inf>) {
         $ic =~ s/\@(?:code|kbd)/C/;
         $ic =~ s/\@(?:dfn|var|emph|cite|i)/I/;
         $ic =~ s/\@(?:file)/F/;
+        $ic =~ s/\@(?:columnfractions)//;
         $_ = "\n=over 4\n";
     };
 
@@ -288,6 +289,21 @@ INF: while(<$inf>) {
         $_ = "";        # need a paragraph break
     };
 
+    /^\@item\s+(.*\S)\s*$/ and $endw eq "multitable" and do {
+        my $columns = $1;
+        $columns =~ s/\@tab/ : /;
+
+        $_ = "\n=item B&LT;". $columns ."&GT;\n";
+    };
+
+    /^\@tab\s+(.*\S)\s*$/ and $endw eq "multitable" and do {
+        my $columns = $1;
+        $columns =~ s/\@tab/ : /;
+
+        $_ = " : ". $columns;
+        $chapter =~ s/\n+\s+$//;
+    };
+
     /^\@itemx?\s*(.+)?$/ and do {
         if (defined $1) {
             # Entity escapes prevent munging by the <> processing below.
@@ -361,6 +377,7 @@ sub postprocess
     s/\(?\@xref\{(?:[^\}]*)\}(?:[^.<]|(?:<[^<>]*>))*\.\)?//g;
     s/\s+\(\@pxref\{(?:[^\}]*)\}\)//g;
     s/;\s+\@pxref\{(?:[^\}]*)\}//g;
+    s/\@ref\{(?:[^,\}]*,)(?:[^,\}]*,)([^,\}]*).*\}/$1/g;
     s/\@ref\{([^\}]*)\}/$1/g;
     s/\@noindent\s*//g;
     s/\@refill//g;

ffmpeg.c

@@ -109,8 +109,18 @@ const int program_birth_year = 2000;
 
 static FILE *vstats_file;
 
+const char *const forced_keyframes_const_names[] = {
+    "n",
+    "n_forced",
+    "prev_forced_n",
+    "prev_forced_t",
+    "t",
+    NULL
+};
+
 static void do_video_stats(OutputStream *ost, int frame_size);
 static int64_t getutime(void);
+static int64_t getmaxrss(void);
 
 static int run_as_daemon  = 0;
 static int64_t video_size = 0;
@@ -152,6 +162,8 @@ static struct termios oldtty;
 static int restore_tty;
 #endif
 
+static void free_input_threads(void);
+
 
 /* sub2video hack:
    Convert subtitles to video with alpha to insert them in filter graphs.
@@ -399,6 +411,11 @@ static void exit_program(void)
 {
     int i, j;
 
+    if (do_benchmark) {
+        int maxrss = getmaxrss() / 1024;
+        printf("bench: maxrss=%ikB\n", maxrss);
+    }
+
     for (i = 0; i < nb_filtergraphs; i++) {
         avfilter_graph_free(&filtergraphs[i]->graph);
         for (j = 0; j < filtergraphs[i]->nb_inputs; j++) {
@@ -437,10 +454,14 @@ static void exit_program(void)
         avcodec_free_frame(&output_streams[i]->filtered_frame);
 
         av_freep(&output_streams[i]->forced_keyframes);
+        av_expr_free(output_streams[i]->forced_keyframes_pexpr);
         av_freep(&output_streams[i]->avfilter);
         av_freep(&output_streams[i]->logfile_prefix);
         av_freep(&output_streams[i]);
     }
+#if HAVE_PTHREADS
+    free_input_threads();
+#endif
     for (i = 0; i < nb_input_files; i++) {
         avformat_close_input(&input_files[i]->ctx);
         av_freep(&input_files[i]);
@@ -449,6 +470,7 @@ static void exit_program(void)
         avcodec_free_frame(&input_streams[i]->decoded_frame);
         av_dict_free(&input_streams[i]->opts);
         free_buffer_pool(&input_streams[i]->buffer_pool);
+        avsubtitle_free(&input_streams[i]->prev_sub.subtitle);
         avfilter_unref_bufferp(&input_streams[i]->sub2video.ref);
         av_freep(&input_streams[i]->filters);
         av_freep(&input_streams[i]);
@@ -471,7 +493,6 @@ static void exit_program(void)
     if (received_sigterm) {
         av_log(NULL, AV_LOG_INFO, "Received signal %d: terminating.\n",
                (int) received_sigterm);
-        exit (255);
     }
 }
 
@@ -665,6 +686,7 @@ static void do_audio_out(AVFormatContext *s, OutputStream *ost,
     }
 }
 
+#if FF_API_DEINTERLACE
 static void pre_process_video_frame(InputStream *ist, AVPicture *picture, void **bufp)
 {
     AVCodecContext *dec;
@@ -675,7 +697,7 @@ static void pre_process_video_frame(InputStream *ist, AVPicture *picture, void *
     dec = ist->st->codec;
 
     /* deinterlace : must be done before any resize */
-    if (do_deinterlace) {
+    if (FF_API_DEINTERLACE && do_deinterlace) {
         int size;
 
         /* create temporary picture */
@@ -705,6 +727,7 @@ static void pre_process_video_frame(InputStream *ist, AVPicture *picture, void *
         *picture = *picture2;
     *bufp = buf;
 }
+#endif
 
 static void do_subtitle_out(AVFormatContext *s,
                             OutputStream *ost,
@@ -873,8 +896,9 @@ static void do_video_out(AVFormatContext *s,
         video_size += pkt.size;
         write_frame(s, &pkt, ost);
     } else {
-        int got_packet;
+        int got_packet, forced_keyframe = 0;
         AVFrame big_picture;
+        double pts_time;
 
         big_picture = *in_picture;
         /* better than nothing: use input picture interlaced
@@ -898,11 +922,41 @@ static void do_video_out(AVFormatContext *s,
         big_picture.quality = ost->st->codec->global_quality;
         if (!enc->me_threshold)
             big_picture.pict_type = 0;
+
+        pts_time = big_picture.pts != AV_NOPTS_VALUE ?
+            big_picture.pts * av_q2d(enc->time_base) : NAN;
         if (ost->forced_kf_index < ost->forced_kf_count &&
             big_picture.pts >= ost->forced_kf_pts[ost->forced_kf_index]) {
-            big_picture.pict_type = AV_PICTURE_TYPE_I;
             ost->forced_kf_index++;
+            forced_keyframe = 1;
+        } else if (ost->forced_keyframes_pexpr) {
+            double res;
+            ost->forced_keyframes_expr_const_values[FKF_T] = pts_time;
+            res = av_expr_eval(ost->forced_keyframes_pexpr,
+                               ost->forced_keyframes_expr_const_values, NULL);
+            av_dlog(NULL, "force_key_frame: n:%f n_forced:%f prev_forced_n:%f t:%f prev_forced_t:%f -> res:%f\n",
+                    ost->forced_keyframes_expr_const_values[FKF_N],
+                    ost->forced_keyframes_expr_const_values[FKF_N_FORCED],
+                    ost->forced_keyframes_expr_const_values[FKF_PREV_FORCED_N],
+                    ost->forced_keyframes_expr_const_values[FKF_T],
+                    ost->forced_keyframes_expr_const_values[FKF_PREV_FORCED_T],
+                    res);
+            if (res) {
+                forced_keyframe = 1;
+                ost->forced_keyframes_expr_const_values[FKF_PREV_FORCED_N] =
+                    ost->forced_keyframes_expr_const_values[FKF_N];
+                ost->forced_keyframes_expr_const_values[FKF_PREV_FORCED_T] =
+                    ost->forced_keyframes_expr_const_values[FKF_T];
+                ost->forced_keyframes_expr_const_values[FKF_N_FORCED] += 1;
+            }
+
+            ost->forced_keyframes_expr_const_values[FKF_N] += 1;
         }
+        if (forced_keyframe) {
+            big_picture.pict_type = AV_PICTURE_TYPE_I;
+            av_log(NULL, AV_LOG_DEBUG, "Forced keyframe at time %f\n", pts_time);
+        }
+
         update_benchmark(NULL);
         ret = avcodec_encode_video2(enc, &pkt, &big_picture, &got_packet);
         update_benchmark("encode_video %d.%d", ost->file_index, ost->index);
@@ -1062,6 +1116,12 @@ static int reap_filters(void)
             case AVMEDIA_TYPE_AUDIO:
                 avfilter_copy_buf_props(filtered_frame, picref);
                 filtered_frame->pts = frame_pts;
+                if (!(ost->st->codec->codec->capabilities & CODEC_CAP_PARAM_CHANGE) &&
+                    ost->st->codec->channels != av_frame_get_channels(filtered_frame)) {
+                    av_log(NULL, AV_LOG_ERROR,
+                           "Audio filter graph output is not normalized and encoder does not support parameter changes\n");
+                    break;
+                }
                 do_audio_out(of->ctx, ost, filtered_frame);
                 break;
             default:
@@ -1168,7 +1228,7 @@ static void print_report(int is_last_report, int64_t timer_start, int64_t cur_ti
                     p = psnr(error / scale);
                     snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "%c:%2.2f ", type[j], p);
                     av_bprintf(&buf_script, "stream_%d_%d_psnr_%c=%2.2f\n",
-                               ost->file_index, ost->index, type[i] | 32, p);
+                               ost->file_index, ost->index, type[j] | 32, p);
                 }
                 p = psnr(error_sum / scale_sum);
                 snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "*:%2.2f ", psnr(error_sum / scale_sum));
@@ -1216,7 +1276,10 @@ static void print_report(int is_last_report, int64_t timer_start, int64_t cur_ti
     av_bprintf(&buf_script, "drop_frames=%d\n", nb_frames_drop);
 
     if (print_stats || is_last_report) {
-    av_log(NULL, AV_LOG_INFO, "%s    \r", buf);
+        if (print_stats==1 && AV_LOG_INFO > av_log_get_level()) {
+            fprintf(stderr, "%s    \r", buf);
+        } else
+            av_log(NULL, AV_LOG_INFO, "%s    \r", buf);
 
     fflush(stderr);
     }
@@ -1443,6 +1506,8 @@ int guess_input_channel_layout(InputStream *ist)
     if (!dec->channel_layout) {
         char layout_name[256];
 
+        if (dec->channels > ist->guess_layout_max)
+            return 0;
         dec->channel_layout = av_get_default_channel_layout(dec->channels);
         if (!dec->channel_layout)
             return 0;
@@ -1617,7 +1682,9 @@ static int decode_video(InputStream *ist, AVPacket *pkt, int *got_output)
     }
 
     pkt->size = 0;
+#if FF_API_DEINTERLACE
     pre_process_video_frame(ist, (AVPicture *)decoded_frame, &buffer_to_free);
+#endif
 
     rate_emu_sleep(ist);
 
@@ -1843,7 +1910,10 @@ static int output_packet(InputStream *ist, const AVPacket *pkt)
                              ist->st->codec->sample_rate;
             break;
         case AVMEDIA_TYPE_VIDEO:
-            if (pkt->duration) {
+            if (ist->framerate.num) {
+                int64_t next_dts = av_rescale_q(ist->next_dts, AV_TIME_BASE_Q, av_inv_q(ist->framerate));
+                ist->next_dts = av_rescale_q(next_dts + 1, av_inv_q(ist->framerate), AV_TIME_BASE_Q);
+            } else if (pkt->duration) {
                 ist->next_dts += av_rescale_q(pkt->duration, ist->st->time_base, AV_TIME_BASE_Q);
             } else if(ist->st->codec->time_base.num != 0) {
                 int ticks= ist->st->parser ? ist->st->parser->repeat_pict + 1 : ist->st->codec->ticks_per_frame;
@@ -1870,7 +1940,7 @@ static int output_packet(InputStream *ist, const AVPacket *pkt)
 
 static void print_sdp(void)
 {
-    char sdp[2048];
+    char sdp[16384];
     int i;
     AVFormatContext **avc = av_malloc(sizeof(*avc) * nb_output_files);
 
@@ -1898,7 +1968,7 @@ static int init_input_stream(int ist_index, char *error, int error_len)
             return AVERROR(EINVAL);
         }
 
-        ist->dr1 = (codec->capabilities & CODEC_CAP_DR1) && !do_deinterlace;
+        ist->dr1 = (codec->capabilities & CODEC_CAP_DR1) && !(FF_API_DEINTERLACE && do_deinterlace);
         if (codec->type == AVMEDIA_TYPE_VIDEO && ist->dr1) {
             ist->st->codec->get_buffer     = codec_get_buffer;
             ist->st->codec->release_buffer = codec_release_buffer;
@@ -1931,19 +2001,25 @@ static InputStream *get_input_stream(OutputStream *ost)
     return NULL;
 }
 
+static int compare_int64(const void *a, const void *b)
+{
+    int64_t va = *(int64_t *)a, vb = *(int64_t *)b;
+    return va < vb ? -1 : va > vb ? +1 : 0;
+}
+
 static void parse_forced_key_frames(char *kf, OutputStream *ost,
                                     AVCodecContext *avctx)
 {
     char *p;
-    int n = 1, i;
-    int64_t t;
+    int n = 1, i, size, index = 0;
+    int64_t t, *pts;
 
     for (p = kf; *p; p++)
         if (*p == ',')
             n++;
-    ost->forced_kf_count = n;
-    ost->forced_kf_pts   = av_malloc(sizeof(*ost->forced_kf_pts) * n);
-    if (!ost->forced_kf_pts) {
+    size = n;
+    pts = av_malloc(sizeof(*pts) * size);
+    if (!pts) {
         av_log(NULL, AV_LOG_FATAL, "Could not allocate forced key frames array.\n");
         exit(1);
     }
@@ -1955,11 +2031,43 @@ static void parse_forced_key_frames(char *kf, OutputStream *ost,
         if (next)
             *next++ = 0;
 
-        t = parse_time_or_die("force_key_frames", p, 1);
-        ost->forced_kf_pts[i] = av_rescale_q(t, AV_TIME_BASE_Q, avctx->time_base);
+        if (!memcmp(p, "chapters", 8)) {
+
+            AVFormatContext *avf = output_files[ost->file_index]->ctx;
+            int j;
+
+            if (avf->nb_chapters > INT_MAX - size ||
+                !(pts = av_realloc_f(pts, size += avf->nb_chapters - 1,
+                                     sizeof(*pts)))) {
+                av_log(NULL, AV_LOG_FATAL,
+                       "Could not allocate forced key frames array.\n");
+                exit(1);
+            }
+            t = p[8] ? parse_time_or_die("force_key_frames", p + 8, 1) : 0;
+            t = av_rescale_q(t, AV_TIME_BASE_Q, avctx->time_base);
+
+            for (j = 0; j < avf->nb_chapters; j++) {
+                AVChapter *c = avf->chapters[j];
+                av_assert1(index < size);
+                pts[index++] = av_rescale_q(c->start, c->time_base,
+                                            avctx->time_base) + t;
+            }
+
+        } else {
+
+            t = parse_time_or_die("force_key_frames", p, 1);
+            av_assert1(index < size);
+            pts[index++] = av_rescale_q(t, AV_TIME_BASE_Q, avctx->time_base);
+
+        }
 
         p = next;
     }
+
+    av_assert0(index == size);
+    qsort(pts, size, sizeof(*pts), compare_int64);
+    ost->forced_kf_count = size;
+    ost->forced_kf_pts   = pts;
 }
 
 static void report_new_stream(int input_index, AVPacket *pkt)
@@ -2046,9 +2154,10 @@ static int transcode_init(void)
             codec->codec_type = icodec->codec_type;
 
             if (!codec->codec_tag) {
+                unsigned int codec_tag;
                 if (!oc->oformat->codec_tag ||
                      av_codec_get_id (oc->oformat->codec_tag, icodec->codec_tag) == codec->codec_id ||
-                     av_codec_get_tag(oc->oformat->codec_tag, icodec->codec_id) <= 0)
+                     !av_codec_get_tag2(oc->oformat->codec_tag, icodec->codec_id, &codec_tag))
                     codec->codec_tag = icodec->codec_tag;
             }
 
@@ -2100,7 +2209,15 @@ static int transcode_init(void)
                     codec->time_base.num *= icodec->ticks_per_frame;
                 }
             }
+            if (   codec->codec_tag == AV_RL32("tmcd")
+                && icodec->time_base.num < icodec->time_base.den
+                && icodec->time_base.num > 0
+                && 121LL*icodec->time_base.num > icodec->time_base.den) {
+                codec->time_base = icodec->time_base;
+            }
 
+            if (ist && !ost->frame_rate.num)
+                ost->frame_rate = ist->framerate;
             if(ost->frame_rate.num)
                 codec->time_base = av_inv_q(ost->frame_rate);
 
@@ -2119,7 +2236,7 @@ static int transcode_init(void)
                 codec->frame_size         = icodec->frame_size;
                 codec->audio_service_type = icodec->audio_service_type;
                 codec->block_align        = icodec->block_align;
-                if((codec->block_align == 1 || codec->block_align == 1152) && codec->codec_id == AV_CODEC_ID_MP3)
+                if((codec->block_align == 1 || codec->block_align == 1152 || codec->block_align == 576) && codec->codec_id == AV_CODEC_ID_MP3)
                     codec->block_align= 0;
                 if(codec->codec_id == AV_CODEC_ID_AC3)
                     codec->block_align= 0;
@@ -2193,7 +2310,7 @@ static int transcode_init(void)
                 codec->sample_fmt     = ost->filter->filter->inputs[0]->format;
                 codec->sample_rate    = ost->filter->filter->inputs[0]->sample_rate;
                 codec->channel_layout = ost->filter->filter->inputs[0]->channel_layout;
-                codec->channels       = av_get_channel_layout_nb_channels(codec->channel_layout);
+                codec->channels       = avfilter_link_get_channels(ost->filter->filter->inputs[0]);
                 codec->time_base      = (AVRational){ 1, codec->sample_rate };
                 break;
             case AVMEDIA_TYPE_VIDEO:
@@ -2225,9 +2342,23 @@ static int transcode_init(void)
                     codec->bits_per_raw_sample = frame_bits_per_raw_sample;
                 }
 
-                if (ost->forced_keyframes)
-                    parse_forced_key_frames(ost->forced_keyframes, ost,
-                                            ost->st->codec);
+                if (ost->forced_keyframes) {
+                    if (!strncmp(ost->forced_keyframes, "expr:", 5)) {
+                        ret = av_expr_parse(&ost->forced_keyframes_pexpr, ost->forced_keyframes+5,
+                                            forced_keyframes_const_names, NULL, NULL, NULL, NULL, 0, NULL);
+                        if (ret < 0) {
+                            av_log(NULL, AV_LOG_ERROR,
+                                   "Invalid force_key_frames expression '%s'\n", ost->forced_keyframes+5);
+                            return ret;
+                        }
+                        ost->forced_keyframes_expr_const_values[FKF_N] = 0;
+                        ost->forced_keyframes_expr_const_values[FKF_N_FORCED] = 0;
+                        ost->forced_keyframes_expr_const_values[FKF_PREV_FORCED_N] = NAN;
+                        ost->forced_keyframes_expr_const_values[FKF_PREV_FORCED_T] = NAN;
+                    } else {
+                        parse_forced_key_frames(ost->forced_keyframes, ost, ost->st->codec);
+                    }
+                }
                 break;
             case AVMEDIA_TYPE_SUBTITLE:
                 codec->time_base = (AVRational){1, 1000};
@@ -2316,13 +2447,20 @@ static int transcode_init(void)
 
             if (ost->st->codec->me_threshold)
                 input_streams[ost->source_index]->st->codec->debug |= FF_DEBUG_MV;
+        } else {
+            av_opt_set_dict(ost->st->codec, &ost->opts);
         }
     }
 
     /* init input streams */
     for (i = 0; i < nb_input_streams; i++)
-        if ((ret = init_input_stream(i, error, sizeof(error))) < 0)
+        if ((ret = init_input_stream(i, error, sizeof(error))) < 0) {
+            for (i = 0; i < nb_output_streams; i++) {
+                ost = output_streams[i];
+                avcodec_close(ost->st->codec);
+            }
             goto dump_format;
+        }
 
     /* discard unused programs */
     for (i = 0; i < nb_input_files; i++) {
@@ -3105,6 +3243,8 @@ static int transcode(void)
                 av_freep(&ost->st->codec->subtitle_header);
                 av_free(ost->forced_kf_pts);
                 av_dict_free(&ost->opts);
+                av_dict_free(&ost->swr_opts);
+                av_dict_free(&ost->resample_opts);
             }
         }
     }
@@ -3210,10 +3350,9 @@ int main(int argc, char **argv)
         exit(1);
     ti = getutime() - ti;
     if (do_benchmark) {
-        int maxrss = getmaxrss() / 1024;
-        printf("bench: utime=%0.3fs maxrss=%ikB\n", ti / 1000000.0, maxrss);
+        printf("bench: utime=%0.3fs\n", ti / 1000000.0);
     }
 
-    exit(0);
+    exit(received_nb_signals ? 255 : 0);
     return 0;
 }

ffmpeg.h

@@ -41,6 +41,7 @@
 
 #include "libavutil/avutil.h"
 #include "libavutil/dict.h"
+#include "libavutil/eval.h"
 #include "libavutil/fifo.h"
 #include "libavutil/pixfmt.h"
 #include "libavutil/rational.h"
@@ -113,6 +114,7 @@ typedef struct OptionsContext {
     int chapters_input_file;
 
     int64_t recording_time;
+    int64_t stop_time;
     uint64_t limit_filesize;
     float mux_preload;
     float mux_max_delay;
@@ -171,6 +173,8 @@ typedef struct OptionsContext {
     int        nb_pass;
     SpecifierOpt *passlogfiles;
     int        nb_passlogfiles;
+    SpecifierOpt *guess_layout_max;
+    int        nb_guess_layout_max;
 } OptionsContext;
 
 typedef struct InputFilter {
@@ -195,6 +199,7 @@ typedef struct FilterGraph {
     const char    *graph_desc;
 
     AVFilterGraph *graph;
+    int reconfiguration;
 
     InputFilter   **inputs;
     int          nb_inputs;
@@ -229,6 +234,7 @@ typedef struct InputStream {
     AVDictionary *opts;
     AVRational framerate;               /* framerate forced with -r */
     int top_field_first;
+    int guess_layout_max;
 
     int resample_height;
     int resample_width;
@@ -286,6 +292,17 @@ typedef struct InputFile {
 #endif
 } InputFile;
 
+enum forced_keyframes_const {
+    FKF_N,
+    FKF_N_FORCED,
+    FKF_PREV_FORCED_N,
+    FKF_PREV_FORCED_T,
+    FKF_T,
+    FKF_NB
+};
+
+extern const char *const forced_keyframes_const_names[];
+
 typedef struct OutputStream {
     int file_index;          /* file index */
     int index;               /* stream index in the output file */
@@ -317,6 +334,8 @@ typedef struct OutputStream {
     int forced_kf_count;
     int forced_kf_index;
     char *forced_keyframes;
+    AVExpr *forced_keyframes_pexpr;
+    double forced_keyframes_expr_const_values[FKF_NB];
 
     /* audio only */
     int audio_channels_map[SWR_CH_MAX];  /* list of the channels id to pick from the source stream */
@@ -329,10 +348,9 @@ typedef struct OutputStream {
     char *avfilter;
 
     int64_t sws_flags;
-    int64_t swr_filter_type;
-    int64_t swr_dither_method;
-    double swr_dither_scale;
     AVDictionary *opts;
+    AVDictionary *swr_opts;
+    AVDictionary *resample_opts;
     int finished;        /* no more packets should be written for this stream */
     int unavailable;                     /* true if the steram is unavailable (possibly temporarily) */
     int stream_copy;

ffmpeg_filter.c

@@ -24,6 +24,8 @@
 #include "libavfilter/avfiltergraph.h"
 #include "libavfilter/buffersink.h"
 
+#include "libavresample/avresample.h"
+
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
 #include "libavutil/bprint.h"
@@ -41,12 +43,15 @@ enum AVPixelFormat choose_pixel_fmt(AVStream *st, AVCodec *codec, enum AVPixelFo
         const AVPixFmtDescriptor *desc = av_pix_fmt_desc_get(target);
         int has_alpha = desc ? desc->nb_components % 2 == 0 : 0;
         enum AVPixelFormat best= AV_PIX_FMT_NONE;
+        const enum AVPixelFormat mjpeg_formats[] = { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUV420P, AV_PIX_FMT_YUV422P, AV_PIX_FMT_NONE };
+        const enum AVPixelFormat ljpeg_formats[] = { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUVJ444P, AV_PIX_FMT_YUV420P,
+                                                     AV_PIX_FMT_YUV422P, AV_PIX_FMT_YUV444P, AV_PIX_FMT_BGRA, AV_PIX_FMT_NONE };
+
         if (st->codec->strict_std_compliance <= FF_COMPLIANCE_UNOFFICIAL) {
             if (st->codec->codec_id == AV_CODEC_ID_MJPEG) {
-                p = (const enum AVPixelFormat[]) { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUV420P, AV_PIX_FMT_YUV422P, AV_PIX_FMT_NONE };
+                p = mjpeg_formats;
             } else if (st->codec->codec_id == AV_CODEC_ID_LJPEG) {
-                p = (const enum AVPixelFormat[]) { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUVJ444P, AV_PIX_FMT_YUV420P,
-                                                 AV_PIX_FMT_YUV422P, AV_PIX_FMT_YUV444P, AV_PIX_FMT_BGRA, AV_PIX_FMT_NONE };
+                p =ljpeg_formats;
             }
         }
         for (; *p != AV_PIX_FMT_NONE; p++) {
@@ -139,7 +144,7 @@ static char *choose_ ## var ## s(OutputStream *ost)                            \
     if (ost->st->codec->var != none) {                                         \
         get_name(ost->st->codec->var);                                         \
         return av_strdup(name);                                                \
-    } else if (ost->enc->supported_list) {                                     \
+    } else if (ost->enc && ost->enc->supported_list) {                         \
         const type *p;                                                         \
         AVIOContext *s = NULL;                                                 \
         uint8_t *ret;                                                          \
@@ -367,12 +372,16 @@ static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter,
     char *sample_fmts, *sample_rates, *channel_layouts;
     char name[255];
     int ret;
+    AVABufferSinkParams *params = av_abuffersink_params_alloc();
 
-
+    if (!params)
+        return AVERROR(ENOMEM);
+    params->all_channel_counts = 1;
     snprintf(name, sizeof(name), "output stream %d:%d", ost->file_index, ost->index);
     ret = avfilter_graph_create_filter(&ofilter->filter,
                                        avfilter_get_by_name("ffabuffersink"),
-                                       name, NULL, NULL, fg->graph);
+                                       name, NULL, params, fg->graph);
+    av_freep(&params);
     if (ret < 0)
         return ret;
 
@@ -522,7 +531,7 @@ static int sub2video_prepare(InputStream *ist)
 
     /* rectangles are AV_PIX_FMT_PAL8, but we have no guarantee that the
        palettes for all rectangles are identical or compatible */
-    ist->st->codec->pix_fmt = AV_PIX_FMT_RGB32;
+    ist->resample_pix_fmt = ist->st->codec->pix_fmt = AV_PIX_FMT_RGB32;
 
     ret = av_image_alloc(image, linesize, w, h, AV_PIX_FMT_RGB32, 32);
     if (ret < 0)
@@ -620,20 +629,25 @@ static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
     AVFilter *filter = avfilter_get_by_name("abuffer");
     InputStream *ist = ifilter->ist;
     int pad_idx = in->pad_idx;
-    char args[255], name[255];
+    AVBPrint args;
+    char name[255];
     int ret;
 
-    snprintf(args, sizeof(args), "time_base=%d/%d:sample_rate=%d:sample_fmt=%s"
-             ":channel_layout=0x%"PRIx64,
+    av_bprint_init(&args, 0, AV_BPRINT_SIZE_AUTOMATIC);
+    av_bprintf(&args, "time_base=%d/%d:sample_rate=%d:sample_fmt=%s",
              1, ist->st->codec->sample_rate,
              ist->st->codec->sample_rate,
-             av_get_sample_fmt_name(ist->st->codec->sample_fmt),
-             ist->st->codec->channel_layout);
+             av_get_sample_fmt_name(ist->st->codec->sample_fmt));
+    if (ist->st->codec->channel_layout)
+        av_bprintf(&args, ":channel_layout=0x%"PRIx64,
+                   ist->st->codec->channel_layout);
+    else
+        av_bprintf(&args, ":channels=%d", ist->st->codec->channels);
     snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index,
              ist->file_index, ist->st->index);
 
     if ((ret = avfilter_graph_create_filter(&ifilter->filter, filter,
-                                            name, args, NULL,
+                                            name, args.str, NULL,
                                             fg->graph)) < 0)
         return ret;
 
@@ -664,6 +678,8 @@ static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
         av_strlcatf(args, sizeof(args), "async=%d", audio_sync_method);
         if (audio_drift_threshold != 0.1)
             av_strlcatf(args, sizeof(args), ":min_hard_comp=%f", audio_drift_threshold);
+        if (!fg->reconfiguration)
+            av_strlcatf(args, sizeof(args), ":first_pts=0");
         AUTO_INSERT_FILTER_INPUT("-async", "aresample", args);
     }
 
@@ -721,20 +737,29 @@ int configure_filtergraph(FilterGraph *fg)
 
     if (simple) {
         OutputStream *ost = fg->outputs[0]->ost;
-        char args[255];
+        char args[512];
+        AVDictionaryEntry *e = NULL;
+
         snprintf(args, sizeof(args), "flags=0x%X", (unsigned)ost->sws_flags);
         fg->graph->scale_sws_opts = av_strdup(args);
 
         args[0] = 0;
-        if (ost->swr_filter_type != SWR_FILTER_TYPE_KAISER)
-            av_strlcatf(args, sizeof(args), "filter_type=%d:", (int)ost->swr_filter_type);
-        if (ost->swr_dither_method)
-            av_strlcatf(args, sizeof(args), "dither_method=%d:", (int)ost->swr_dither_method);
-        if (ost->swr_dither_scale != 1.0)
-            av_strlcatf(args, sizeof(args), "dither_scale=%f:", ost->swr_dither_scale);
+        while ((e = av_dict_get(ost->swr_opts, "", e,
+                                AV_DICT_IGNORE_SUFFIX))) {
+            av_strlcatf(args, sizeof(args), "%s=%s:", e->key, e->value);
+        }
         if (strlen(args))
             args[strlen(args)-1] = 0;
         av_opt_set(fg->graph, "aresample_swr_opts", args, 0);
+
+        args[0] = '\0';
+        while ((e = av_dict_get(fg->outputs[0]->ost->resample_opts, "", e,
+                                AV_DICT_IGNORE_SUFFIX))) {
+            av_strlcatf(args, sizeof(args), "%s=%s:", e->key, e->value);
+        }
+        if (strlen(args))
+            args[strlen(args) - 1] = '\0';
+        fg->graph->resample_lavr_opts = av_strdup(args);
     }
 
     if ((ret = avfilter_graph_parse2(fg->graph, graph_desc, &inputs, &outputs)) < 0)
@@ -776,6 +801,7 @@ int configure_filtergraph(FilterGraph *fg)
         }
     }
 
+    fg->reconfiguration = 1;
     return 0;
 }
 

ffmpeg_opt.c

@@ -81,7 +81,7 @@ int copy_ts           = 0;
 int copy_tb           = -1;
 int debug_ts          = 0;
 int exit_on_error     = 0;
-int print_stats       = 1;
+int print_stats       = -1;
 int qp_hist           = 0;
 int stdin_interaction = 1;
 int frame_bits_per_raw_sample = 0;
@@ -126,6 +126,7 @@ static void uninit_options(OptionsContext *o, int is_input)
     av_freep(&o->stream_maps);
     av_freep(&o->audio_channel_maps);
     av_freep(&o->streamid_map);
+    av_freep(&o->attachments);
 
     if (is_input)
         recording_time = o->recording_time;
@@ -143,24 +144,13 @@ static void init_options(OptionsContext *o, int is_input)
                 "-t is not an input option, keeping it for the next output;"
                 " consider fixing your command line.\n");
     } else
-    o->recording_time = INT64_MAX;
+        o->recording_time = INT64_MAX;
+    o->stop_time = INT64_MAX;
     o->mux_max_delay  = 0.7;
     o->limit_filesize = UINT64_MAX;
     o->chapters_input_file = INT_MAX;
 }
 
-static int opt_frame_crop(void *optctx, const char *opt, const char *arg)
-{
-    av_log(NULL, AV_LOG_FATAL, "Option '%s' has been removed, use the crop filter instead\n", opt);
-    return AVERROR(EINVAL);
-}
-
-static int opt_pad(void *optctx, const char *opt, const char *arg)
-{
-    av_log(NULL, AV_LOG_FATAL, "Option '%s' has been removed, use the pad filter instead\n", opt);
-    return -1;
-}
-
 static int opt_sameq(void *optctx, const char *opt, const char *arg)
 {
     av_log(NULL, AV_LOG_ERROR, "Option '%s' was removed. "
@@ -616,6 +606,8 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
 
             break;
         case AVMEDIA_TYPE_AUDIO:
+            ist->guess_layout_max = INT_MAX;
+            MATCH_PER_STREAM_OPT(guess_layout_max, i, ist->guess_layout_max, ic, st);
             guess_input_channel_layout(ist);
 
             ist->resample_sample_fmt     = dec->sample_fmt;
@@ -965,6 +957,8 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e
                    preset, ost->file_index, ost->index);
             exit(1);
         }
+    } else {
+        ost->opts = filter_codec_opts(o->g->codec_opts, AV_CODEC_ID_NONE, oc, st, NULL);
     }
 
     avcodec_get_context_defaults3(st->codec, ost->enc);
@@ -1011,9 +1005,12 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e
         st->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
 
     av_opt_get_int(o->g->sws_opts, "sws_flags", 0, &ost->sws_flags);
-    av_opt_get_int   (o->g->swr_opts, "filter_type"  , 0, &ost->swr_filter_type);
-    av_opt_get_int   (o->g->swr_opts, "dither_method", 0, &ost->swr_dither_method);
-    av_opt_get_double(o->g->swr_opts, "dither_scale" , 0, &ost->swr_dither_scale);
+
+    av_dict_copy(&ost->swr_opts, o->g->swr_opts, 0);
+    if (ost->enc && av_get_exact_bits_per_sample(ost->enc->id) == 24)
+        av_dict_set(&ost->swr_opts, "output_sample_bits", "24", 0);
+
+    av_dict_copy(&ost->resample_opts, o->g->resample_opts, 0);
 
     ost->source_index = source_index;
     if (source_index >= 0) {
@@ -1143,8 +1140,6 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
             if (p) p++;
         }
         video_enc->rc_override_count = i;
-        if (!video_enc->rc_initial_buffer_occupancy)
-            video_enc->rc_initial_buffer_occupancy = video_enc->rc_buffer_size * 3 / 4;
         video_enc->intra_dc_precision = intra_dc_precision - 8;
 
         if (do_psnr)
@@ -1155,9 +1150,11 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
         if (do_pass) {
             if (do_pass & 1) {
                 video_enc->flags |= CODEC_FLAG_PASS1;
+                av_dict_set(&ost->opts, "flags", "+pass1", AV_DICT_APPEND);
             }
             if (do_pass & 2) {
                 video_enc->flags |= CODEC_FLAG_PASS2;
+                av_dict_set(&ost->opts, "flags", "+pass2", AV_DICT_APPEND);
             }
         }
 
@@ -1662,6 +1659,20 @@ loop_end:
                 exit(1);
     }
 
+    if (o->stop_time != INT64_MAX && o->recording_time != INT64_MAX) {
+        o->stop_time = INT64_MAX;
+        av_log(NULL, AV_LOG_WARNING, "-t and -to cannot be used together; using -t.\n");
+    }
+
+    if (o->stop_time != INT64_MAX && o->recording_time == INT64_MAX) {
+        if (o->stop_time <= o->start_time) {
+            av_log(NULL, AV_LOG_WARNING, "-to value smaller than -ss; ignoring -to.\n");
+            o->stop_time = INT64_MAX;
+        } else {
+            o->recording_time = o->stop_time - o->start_time;
+        }
+    }
+
     GROW_ARRAY(output_files, nb_output_files);
     if (!(output_files[nb_output_files - 1] = av_mallocz(sizeof(*output_files[0]))))
         exit(1);
@@ -1695,7 +1706,8 @@ loop_end:
             print_error(filename, err);
             exit(1);
         }
-    }
+    } else if (strcmp(oc->oformat->name, "image2")==0 && !av_filename_number_test(filename))
+        assert_file_overwrite(filename);
 
     if (o->mux_preload) {
         uint8_t buf[64];
@@ -2117,12 +2129,14 @@ static int opt_vsync(void *optctx, const char *opt, const char *arg)
     return 0;
 }
 
+#if FF_API_DEINTERLACE
 static int opt_deinterlace(void *optctx, const char *opt, const char *arg)
 {
     av_log(NULL, AV_LOG_WARNING, "-%s is deprecated, use -filter:v yadif instead\n", opt);
     do_deinterlace = 1;
     return 0;
 }
+#endif
 
 static int opt_timecode(void *optctx, const char *opt, const char *arg)
 {
@@ -2150,7 +2164,7 @@ static int opt_channel_layout(void *optctx, const char *opt, const char *arg)
         return AVERROR(EINVAL);
     }
     snprintf(layout_str, sizeof(layout_str), "%"PRIu64, layout);
-    ret = opt_default(NULL, opt, layout_str);
+    ret = opt_default_new(o, opt, layout_str);
     if (ret < 0)
         return ret;
 
@@ -2297,7 +2311,7 @@ static int open_files(OptionGroupList *l, const char *inout,
                    inout, g->arg);
             return ret;
         }
-        av_log(NULL, AV_LOG_DEBUG, "Successfully openened the file.\n");
+        av_log(NULL, AV_LOG_DEBUG, "Successfully opened the file.\n");
     }
 
     return 0;
@@ -2395,6 +2409,8 @@ const OptionDef options[] = {
     { "t",              HAS_ARG | OPT_TIME | OPT_OFFSET,             { .off = OFFSET(recording_time) },
         "record or transcode \"duration\" seconds of audio/video",
         "duration" },
+    { "to",             HAS_ARG | OPT_TIME | OPT_OFFSET,             { .off = OFFSET(stop_time) },
+        "record or transcode stop time", "time_stop" },
     { "fs",             HAS_ARG | OPT_INT64 | OPT_OFFSET,            { .off = OFFSET(limit_filesize) },
         "set the limit file size in bytes", "limit_size" },
     { "ss",             HAS_ARG | OPT_TIME | OPT_OFFSET,             { .off = OFFSET(start_time) },
@@ -2461,7 +2477,7 @@ const OptionDef options[] = {
     { "profile",        HAS_ARG | OPT_EXPERT | OPT_PERFILE,          { .func_arg = opt_profile },
         "set profile", "profile" },
     { "filter",         HAS_ARG | OPT_STRING | OPT_SPEC,             { .off = OFFSET(filters) },
-        "set stream filterchain", "filter_list" },
+        "set stream filtergraph", "filter_graph" },
     { "reinit_filter",  HAS_ARG | OPT_INT | OPT_SPEC,                { .off = OFFSET(reinit_filters) },
         "reinit filtergraph on input parameter changes", "" },
     { "filter_complex", HAS_ARG | OPT_EXPERT,                        { .func_arg = opt_filter_complex },
@@ -2488,24 +2504,6 @@ const OptionDef options[] = {
         "set pixel format", "format" },
     { "bits_per_raw_sample", OPT_VIDEO | OPT_INT | HAS_ARG,                      { &frame_bits_per_raw_sample },
         "set the number of bits per raw sample", "number" },
-    { "croptop",      OPT_VIDEO | HAS_ARG,                                       { .func_arg = opt_frame_crop },
-        "Removed, use the crop filter instead", "size" },
-    { "cropbottom",   OPT_VIDEO | HAS_ARG,                                       { .func_arg = opt_frame_crop },
-        "Removed, use the crop filter instead", "size" },
-    { "cropleft",     OPT_VIDEO | HAS_ARG,                                       { .func_arg = opt_frame_crop },
-        "Removed, use the crop filter instead", "size" },
-    { "cropright",    OPT_VIDEO | HAS_ARG,                                       { .func_arg = opt_frame_crop },
-        "Removed, use the crop filter instead", "size" },
-    { "padtop",       OPT_VIDEO | HAS_ARG,                                       { .func_arg = opt_pad },
-        "Removed, use the pad filter instead", "size" },
-    { "padbottom",    OPT_VIDEO | HAS_ARG,                                       { .func_arg = opt_pad },
-        "Removed, use the pad filter instead", "size" },
-    { "padleft",      OPT_VIDEO | HAS_ARG,                                       { .func_arg = opt_pad },
-        "Removed, use the pad filter instead", "size" },
-    { "padright",     OPT_VIDEO | HAS_ARG,                                       { .func_arg = opt_pad },
-        "Removed, use the pad filter instead", "size" },
-    { "padcolor",     OPT_VIDEO | HAS_ARG,                                       { .func_arg = opt_pad },
-        "Removed, use the pad filter instead", "color" },
     { "intra",        OPT_VIDEO | OPT_BOOL | OPT_EXPERT,                         { &intra_only },
         "deprecated use -g 1" },
     { "vn",           OPT_VIDEO | OPT_BOOL  | OPT_OFFSET,                        { .off = OFFSET(video_disable) },
@@ -2526,8 +2524,10 @@ const OptionDef options[] = {
         "select the pass number (1 to 3)", "n" },
     { "passlogfile",  OPT_VIDEO | HAS_ARG | OPT_STRING | OPT_EXPERT | OPT_SPEC,  { .off = OFFSET(passlogfiles) },
         "select two pass log file name prefix", "prefix" },
+#if FF_API_DEINTERLACE
     { "deinterlace",  OPT_VIDEO | OPT_EXPERT ,                                   { .func_arg = opt_deinterlace },
         "this option is deprecated, use the yadif filter instead" },
+#endif
     { "psnr",         OPT_VIDEO | OPT_BOOL | OPT_EXPERT,                         { &do_psnr },
         "calculate PSNR of compressed frames" },
     { "vstats",       OPT_VIDEO | OPT_EXPERT ,                                   { &opt_vstats },
@@ -2535,7 +2535,7 @@ const OptionDef options[] = {
     { "vstats_file",  OPT_VIDEO | HAS_ARG | OPT_EXPERT ,                         { opt_vstats_file },
         "dump video coding statistics to file", "file" },
     { "vf",           OPT_VIDEO | HAS_ARG  | OPT_PERFILE,                        { .func_arg = opt_video_filters },
-        "video filters", "filter list" },
+        "set video filters", "filter_graph" },
     { "intra_matrix", OPT_VIDEO | HAS_ARG | OPT_EXPERT  | OPT_STRING | OPT_SPEC, { .off = OFFSET(intra_matrices) },
         "specify intra matrix coeffs", "matrix" },
     { "inter_matrix", OPT_VIDEO | HAS_ARG | OPT_EXPERT  | OPT_STRING | OPT_SPEC, { .off = OFFSET(inter_matrices) },
@@ -2580,7 +2580,9 @@ const OptionDef options[] = {
     { "channel_layout", OPT_AUDIO | HAS_ARG  | OPT_EXPERT | OPT_PERFILE,           { .func_arg = opt_channel_layout },
         "set channel layout", "layout" },
     { "af",             OPT_AUDIO | HAS_ARG  | OPT_PERFILE,                        { .func_arg = opt_audio_filters },
-        "audio filters", "filter list" },
+        "set audio filters", "filter_graph" },
+    { "guess_layout_max", OPT_AUDIO | HAS_ARG | OPT_INT | OPT_SPEC | OPT_EXPERT,   { .off = OFFSET(guess_layout_max) },
+      "set the maximum number of channels to try to guess the channel layout" },
 
     /* subtitle options */
     { "sn",     OPT_SUBTITLE | OPT_BOOL | OPT_OFFSET, { .off = OFFSET(subtitle_disable) },

ffplay.c

@@ -86,6 +86,9 @@ const int program_birth_year = 2003;
 /* we use about AUDIO_DIFF_AVG_NB A-V differences to make the average */
 #define AUDIO_DIFF_AVG_NB   20
 
+/* polls for possible required screen refresh at least this often, should be less than 1/fps */
+#define REFRESH_RATE 0.01
+
 /* NOTE: the size must be big enough to compensate the hardware audio buffersize size */
 /* TODO: We assume that a decoded and resampled frame fits into this buffer */
 #define SAMPLE_ARRAY_SIZE (8 * 65536)
@@ -116,7 +119,6 @@ typedef struct PacketQueue {
 typedef struct VideoPicture {
     double pts;             // presentation timestamp for this picture
     int64_t pos;            // byte position in file
-    int skip;
     SDL_Overlay *bmp;
     int width, height; /* source height & width */
     AVRational sample_aspect_ratio;
@@ -137,7 +139,7 @@ typedef struct SubPicture {
 typedef struct AudioParams {
     int freq;
     int channels;
-    int channel_layout;
+    int64_t channel_layout;
     enum AVSampleFormat fmt;
 } AudioParams;
 
@@ -150,7 +152,6 @@ enum {
 typedef struct VideoState {
     SDL_Thread *read_tid;
     SDL_Thread *video_tid;
-    SDL_Thread *refresh_tid;
     AVInputFormat *iformat;
     int no_background;
     int abort_request;
@@ -175,6 +176,7 @@ typedef struct VideoState {
     double external_clock_speed;             ///< speed of the external clock
 
     double audio_clock;
+    int audio_clock_serial;
     double audio_diff_cum; /* used for AV difference average computation */
     double audio_diff_avg_coef;
     double audio_diff_threshold;
@@ -182,11 +184,11 @@ typedef struct VideoState {
     AVStream *audio_st;
     PacketQueue audioq;
     int audio_hw_buf_size;
-    DECLARE_ALIGNED(16,uint8_t,audio_buf2)[AVCODEC_MAX_AUDIO_FRAME_SIZE * 4];
     uint8_t silence_buf[SDL_AUDIO_BUFFER_SIZE];
     uint8_t *audio_buf;
     uint8_t *audio_buf1;
     unsigned int audio_buf_size; /* in bytes */
+    unsigned int audio_buf1_size;
     int audio_buf_index; /* in bytes */
     int audio_write_buf_size;
     AVPacket audio_pkt_temp;
@@ -211,6 +213,7 @@ typedef struct VideoState {
     int rdft_bits;
     FFTSample *rdft_data;
     int xpos;
+    double last_vis_time;
 
     SDL_Thread *subtitle_tid;
     int subtitle_stream;
@@ -229,14 +232,14 @@ typedef struct VideoState {
     double frame_last_returned_time;
     double frame_last_filter_delay;
     int64_t frame_last_dropped_pos;
-    double video_clock;             // pts of last decoded frame / predicted pts of next decoded frame
     int video_stream;
     AVStream *video_st;
     PacketQueue videoq;
-    double video_current_pts;       // current displayed pts (different from video_clock if frame fifos are used)
+    double video_current_pts;       // current displayed pts
     double video_current_pts_drift; // video_current_pts - time (av_gettime) at which we updated video_current_pts - used to have running video pts
     int64_t video_current_pos;      // current displayed file pos
     double max_frame_duration;      // maximum duration of a frame - above this, we consider the jump a timestamp discontinuity
+    int video_clock_serial;
     VideoPicture pictq[VIDEO_PICTURE_QUEUE_SIZE];
     int pictq_size, pictq_rindex, pictq_windex;
     SDL_mutex *pictq_mutex;
@@ -257,7 +260,6 @@ typedef struct VideoState {
     FrameBuffer *buffer_pool;
 #endif
 
-    int refresh;
     int last_video_stream, last_audio_stream, last_subtitle_stream;
 
     SDL_cond *continue_read_thread;
@@ -269,10 +271,13 @@ static const char *input_filename;
 static const char *window_title;
 static int fs_screen_width;
 static int fs_screen_height;
+static int default_width  = 640;
+static int default_height = 480;
 static int screen_width  = 0;
 static int screen_height = 0;
 static int audio_disable;
 static int video_disable;
+static int subtitle_disable;
 static int wanted_stream[AVMEDIA_TYPE_NB] = {
     [AVMEDIA_TYPE_AUDIO]    = -1,
     [AVMEDIA_TYPE_VIDEO]    = -1,
@@ -304,7 +309,7 @@ static enum ShowMode show_mode = SHOW_MODE_NONE;
 static const char *audio_codec_name;
 static const char *subtitle_codec_name;
 static const char *video_codec_name;
-static int rdftspeed = 20;
+double rdftspeed = 0.02;
 static int64_t cursor_last_shown;
 static int cursor_hidden = 0;
 #if CONFIG_AVFILTER
@@ -318,7 +323,6 @@ static int64_t audio_callback_time;
 static AVPacket flush_pkt;
 
 #define FF_ALLOC_EVENT   (SDL_USEREVENT)
-#define FF_REFRESH_EVENT (SDL_USEREVENT + 1)
 #define FF_QUIT_EVENT    (SDL_USEREVENT + 2)
 
 static SDL_Surface *screen;
@@ -971,7 +975,6 @@ static void stream_close(VideoState *is)
     /* XXX: use a special url_shutdown call to abort parse cleanly */
     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);
@@ -993,8 +996,7 @@ static void stream_close(VideoState *is)
     SDL_DestroyCond(is->subpq_cond);
     SDL_DestroyCond(is->continue_read_thread);
 #if !CONFIG_AVFILTER
-    if (is->img_convert_ctx)
-        sws_freeContext(is->img_convert_ctx);
+    sws_freeContext(is->img_convert_ctx);
 #endif
     av_free(is);
 }
@@ -1023,29 +1025,30 @@ static void sigterm_handler(int sig)
     exit(123);
 }
 
-static int video_open(VideoState *is, int force_set_video_mode)
+static int video_open(VideoState *is, int force_set_video_mode, VideoPicture *vp)
 {
     int flags = SDL_HWSURFACE | SDL_ASYNCBLIT | SDL_HWACCEL;
     int w,h;
-    VideoPicture *vp = &is->pictq[is->pictq_rindex];
     SDL_Rect rect;
 
     if (is_full_screen) flags |= SDL_FULLSCREEN;
     else                flags |= SDL_RESIZABLE;
 
+    if (vp && vp->width) {
+        calculate_display_rect(&rect, 0, 0, INT_MAX, vp->height, vp);
+        default_width  = rect.w;
+        default_height = rect.h;
+    }
+
     if (is_full_screen && fs_screen_width) {
         w = fs_screen_width;
         h = fs_screen_height;
     } else if (!is_full_screen && screen_width) {
         w = screen_width;
         h = screen_height;
-    } else if (vp->width) {
-        calculate_display_rect(&rect, 0, 0, INT_MAX, vp->height, vp);
-        w = rect.w;
-        h = rect.h;
     } else {
-        w = 640;
-        h = 480;
+        w = default_width;
+        h = default_height;
     }
     if (screen && is->width == screen->w && screen->w == w
        && is->height== screen->h && screen->h == h && !force_set_video_mode)
@@ -1069,33 +1072,18 @@ static int video_open(VideoState *is, int force_set_video_mode)
 static void video_display(VideoState *is)
 {
     if (!screen)
-        video_open(is, 0);
+        video_open(is, 0, NULL);
     if (is->audio_st && is->show_mode != SHOW_MODE_VIDEO)
         video_audio_display(is);
     else if (is->video_st)
         video_image_display(is);
 }
 
-static int refresh_thread(void *opaque)
-{
-    VideoState *is= opaque;
-    while (!is->abort_request) {
-        SDL_Event event;
-        event.type = FF_REFRESH_EVENT;
-        event.user.data1 = opaque;
-        if (!is->refresh && (!is->paused || is->force_refresh)) {
-            is->refresh = 1;
-            SDL_PushEvent(&event);
-        }
-        //FIXME ideally we should wait the correct time but SDLs event passing is so slow it would be silly
-        av_usleep(is->audio_st && is->show_mode != SHOW_MODE_VIDEO ? rdftspeed*1000 : 5000);
-    }
-    return 0;
-}
-
 /* get the current audio clock value */
 static double get_audio_clock(VideoState *is)
 {
+    if (is->audio_clock_serial != is->audioq.serial)
+        return NAN;
     if (is->paused) {
         return is->audio_current_pts;
     } else {
@@ -1106,6 +1094,8 @@ static double get_audio_clock(VideoState *is)
 /* get the current video clock value */
 static double get_video_clock(VideoState *is)
 {
+    if (is->video_clock_serial != is->videoq.serial)
+        return NAN;
     if (is->paused) {
         return is->video_current_pts;
     } else {
@@ -1167,7 +1157,8 @@ static void update_external_clock_pts(VideoState *is, double pts)
 }
 
 static void check_external_clock_sync(VideoState *is, double pts) {
-    if (fabs(get_external_clock(is) - pts) > AV_NOSYNC_THRESHOLD) {
+    double ext_clock = get_external_clock(is);
+    if (isnan(ext_clock) || fabs(ext_clock - pts) > AV_NOSYNC_THRESHOLD) {
         update_external_clock_pts(is, pts);
     }
 }
@@ -1201,6 +1192,7 @@ static void stream_seek(VideoState *is, int64_t pos, int64_t rel, int seek_by_by
         if (seek_by_bytes)
             is->seek_flags |= AVSEEK_FLAG_BYTE;
         is->seek_req = 1;
+        SDL_CondSignal(is->continue_read_thread);
     }
 }
 
@@ -1218,6 +1210,20 @@ static void stream_toggle_pause(VideoState *is)
     is->paused = !is->paused;
 }
 
+static void toggle_pause(VideoState *is)
+{
+    stream_toggle_pause(is);
+    is->step = 0;
+}
+
+static void step_to_next_frame(VideoState *is)
+{
+    /* if the stream is paused unpause it, then step */
+    if (is->paused)
+        stream_toggle_pause(is);
+    is->step = 1;
+}
+
 static double compute_target_delay(double delay, VideoState *is)
 {
     double sync_threshold, diff;
@@ -1232,7 +1238,7 @@ static double compute_target_delay(double delay, VideoState *is)
            delay to compute the threshold. I still don't know
            if it is the best guess */
         sync_threshold = FFMAX(AV_SYNC_THRESHOLD, delay);
-        if (fabs(diff) < AV_NOSYNC_THRESHOLD) {
+        if (!isnan(diff) && fabs(diff) < AV_NOSYNC_THRESHOLD) {
             if (diff <= -sync_threshold)
                 delay = 0;
             else if (diff >= sync_threshold)
@@ -1257,20 +1263,23 @@ static void pictq_next_picture(VideoState *is) {
     SDL_UnlockMutex(is->pictq_mutex);
 }
 
-static void pictq_prev_picture(VideoState *is) {
+static int pictq_prev_picture(VideoState *is) {
     VideoPicture *prevvp;
+    int ret = 0;
     /* update queue size and signal for the previous picture */
     prevvp = &is->pictq[(is->pictq_rindex + VIDEO_PICTURE_QUEUE_SIZE - 1) % VIDEO_PICTURE_QUEUE_SIZE];
-    if (prevvp->allocated && !prevvp->skip) {
+    if (prevvp->allocated && prevvp->serial == is->videoq.serial) {
         SDL_LockMutex(is->pictq_mutex);
         if (is->pictq_size < VIDEO_PICTURE_QUEUE_SIZE - 1) {
             if (--is->pictq_rindex == -1)
                 is->pictq_rindex = VIDEO_PICTURE_QUEUE_SIZE - 1;
             is->pictq_size++;
+            ret = 1;
         }
         SDL_CondSignal(is->pictq_cond);
         SDL_UnlockMutex(is->pictq_mutex);
     }
+    return ret;
 }
 
 static void update_video_pts(VideoState *is, double pts, int64_t pos, int serial) {
@@ -1280,12 +1289,13 @@ static void update_video_pts(VideoState *is, double pts, int64_t pos, int serial
     is->video_current_pts_drift = is->video_current_pts - time;
     is->video_current_pos = pos;
     is->frame_last_pts = pts;
+    is->video_clock_serial = serial;
     if (is->videoq.serial == serial)
         check_external_clock_sync(is, is->video_current_pts);
 }
 
 /* called to display each frame */
-static void video_refresh(void *opaque)
+static void video_refresh(void *opaque, double *remaining_time)
 {
     VideoState *is = opaque;
     VideoPicture *vp;
@@ -1296,12 +1306,19 @@ static void video_refresh(void *opaque)
     if (!is->paused && get_master_sync_type(is) == AV_SYNC_EXTERNAL_CLOCK && is->realtime)
         check_external_clock_speed(is);
 
-    if (!display_disable && is->show_mode != SHOW_MODE_VIDEO && is->audio_st)
-        video_display(is);
+    if (!display_disable && is->show_mode != SHOW_MODE_VIDEO && is->audio_st) {
+        time = av_gettime() / 1000000.0;
+        if (is->force_refresh || is->last_vis_time + rdftspeed < time) {
+            video_display(is);
+            is->last_vis_time = time;
+        }
+        *remaining_time = FFMIN(*remaining_time, is->last_vis_time + rdftspeed - time);
+    }
 
     if (is->video_st) {
+        int redisplay = 0;
         if (is->force_refresh)
-            pictq_prev_picture(is);
+            redisplay = pictq_prev_picture(is);
 retry:
         if (is->pictq_size == 0) {
             SDL_LockMutex(is->pictq_mutex);
@@ -1316,8 +1333,9 @@ retry:
             /* dequeue the picture */
             vp = &is->pictq[is->pictq_rindex];
 
-            if (vp->skip) {
+            if (vp->serial != is->videoq.serial) {
                 pictq_next_picture(is);
+                redisplay = 0;
                 goto retry;
             }
 
@@ -1333,8 +1351,10 @@ retry:
             delay = compute_target_delay(is->frame_last_duration, is);
 
             time= av_gettime()/1000000.0;
-            if (time < is->frame_timer + delay)
+            if (time < is->frame_timer + delay) {
+                *remaining_time = FFMIN(is->frame_timer + delay - time, *remaining_time);
                 return;
+            }
 
             if (delay > 0)
                 is->frame_timer += delay * FFMAX(1, floor((time-is->frame_timer) / delay));
@@ -1346,9 +1366,11 @@ retry:
             if (is->pictq_size > 1) {
                 VideoPicture *nextvp = &is->pictq[(is->pictq_rindex + 1) % VIDEO_PICTURE_QUEUE_SIZE];
                 duration = nextvp->pts - vp->pts;
-                if(!is->step && (framedrop>0 || (framedrop && get_master_sync_type(is) != AV_SYNC_VIDEO_MASTER)) && time > is->frame_timer + duration){
-                    is->frame_drops_late++;
+                if(!is->step && (redisplay || framedrop>0 || (framedrop && get_master_sync_type(is) != AV_SYNC_VIDEO_MASTER)) && time > is->frame_timer + duration){
+                    if (!redisplay)
+                        is->frame_drops_late++;
                     pictq_next_picture(is);
+                    redisplay = 0;
                     goto retry;
                 }
             }
@@ -1459,7 +1481,7 @@ static void alloc_picture(VideoState *is)
     avfilter_unref_bufferp(&vp->picref);
 #endif
 
-    video_open(is, 0);
+    video_open(is, 0, vp);
 
     vp->bmp = SDL_CreateYUVOverlay(vp->width, vp->height,
                                    SDL_YV12_OVERLAY,
@@ -1497,29 +1519,13 @@ static void duplicate_right_border_pixels(SDL_Overlay *bmp) {
     }
 }
 
-static int queue_picture(VideoState *is, AVFrame *src_frame, double pts1, int64_t pos, int serial)
+static int queue_picture(VideoState *is, AVFrame *src_frame, double pts, int64_t pos, int serial)
 {
     VideoPicture *vp;
-    double frame_delay, pts = pts1;
-
-    /* compute the exact PTS for the picture if it is omitted in the stream
-     * pts1 is the dts of the pkt / pts of the frame */
-    if (pts != 0) {
-        /* update video clock with pts, if present */
-        is->video_clock = pts;
-    } else {
-        pts = is->video_clock;
-    }
-    /* update video clock for next frame */
-    frame_delay = av_q2d(is->video_st->codec->time_base);
-    /* for MPEG2, the frame can be repeated, so we update the
-       clock accordingly */
-    frame_delay += src_frame->repeat_pict * (frame_delay * 0.5);
-    is->video_clock += frame_delay;
 
 #if defined(DEBUG_SYNC) && 0
-    printf("frame_type=%c clock=%0.3f pts=%0.3f\n",
-           av_get_picture_type_char(src_frame->pict_type), pts, pts1);
+    printf("frame_type=%c pts=%0.3f\n",
+           av_get_picture_type_char(src_frame->pict_type), pts);
 #endif
 
     /* wait until we have space to put a new picture */
@@ -1619,7 +1625,6 @@ static int queue_picture(VideoState *is, AVFrame *src_frame, double pts1, int64_
 
         vp->pts = pts;
         vp->pos = pos;
-        vp->skip = 0;
         vp->serial = serial;
 
         /* now we can update the picture count */
@@ -1634,7 +1639,7 @@ static int queue_picture(VideoState *is, AVFrame *src_frame, double pts1, int64_
 
 static int get_video_frame(VideoState *is, AVFrame *frame, int64_t *pts, AVPacket *pkt, int *serial)
 {
-    int got_picture, i;
+    int got_picture;
 
     if (packet_queue_get(&is->videoq, pkt, 1, serial) < 0)
         return -1;
@@ -1644,9 +1649,6 @@ static int get_video_frame(VideoState *is, AVFrame *frame, int64_t *pts, AVPacke
 
         SDL_LockMutex(is->pictq_mutex);
         // Make sure there are no long delay timers (ideally we should just flush the queue but that's harder)
-        for (i = 0; i < VIDEO_PICTURE_QUEUE_SIZE; i++) {
-            is->pictq[i].skip = 1;
-        }
         while (is->pictq_size && !is->videoq.abort_request) {
             SDL_CondWait(is->pictq_cond, is->pictq_mutex);
         }
@@ -1684,7 +1686,7 @@ static int get_video_frame(VideoState *is, AVFrame *frame, int64_t *pts, AVPacke
                 double clockdiff = get_video_clock(is) - get_master_clock(is);
                 double dpts = av_q2d(is->video_st->time_base) * *pts;
                 double ptsdiff = dpts - is->frame_last_pts;
-                if (fabs(clockdiff) < AV_NOSYNC_THRESHOLD &&
+                if (!isnan(clockdiff) && fabs(clockdiff) < AV_NOSYNC_THRESHOLD &&
                      ptsdiff > 0 && ptsdiff < AV_NOSYNC_THRESHOLD &&
                      clockdiff + ptsdiff - is->frame_last_filter_delay < 0) {
                     is->frame_last_dropped_pos = pkt->pos;
@@ -1753,6 +1755,7 @@ static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const c
     if (!buffersink_params)
         return AVERROR(ENOMEM);
 
+    av_opt_get_int(sws_opts, "sws_flags", 0, &sws_flags);
     snprintf(sws_flags_str, sizeof(sws_flags_str), "flags=%"PRId64, sws_flags);
     graph->scale_sws_opts = av_strdup(sws_flags_str);
 
@@ -1814,6 +1817,7 @@ static int video_thread(void *arg)
     int last_w = 0;
     int last_h = 0;
     enum AVPixelFormat last_format = -2;
+    int last_serial = -1;
 
     if (codec->codec->capabilities & CODEC_CAP_DR1) {
         is->use_dr1 = 1;
@@ -1844,9 +1848,14 @@ static int video_thread(void *arg)
 #if CONFIG_AVFILTER
         if (   last_w != frame->width
             || last_h != frame->height
-            || last_format != frame->format) {
-            av_log(NULL, AV_LOG_INFO, "Frame changed from size:%dx%d to size:%dx%d\n",
-                   last_w, last_h, frame->width, frame->height);
+            || last_format != frame->format
+            || last_serial != serial) {
+            av_log(NULL, AV_LOG_DEBUG,
+                   "Video frame changed from size:%dx%d format:%s serial:%d to size:%dx%d format:%s serial:%d\n",
+                   last_w, last_h,
+                   (const char *)av_x_if_null(av_get_pix_fmt_name(last_format), "none"), last_serial,
+                   frame->width, frame->height,
+                   (const char *)av_x_if_null(av_get_pix_fmt_name(frame->format), "none"), serial);
             avfilter_graph_free(&graph);
             graph = avfilter_graph_alloc();
             if ((ret = configure_video_filters(graph, is, vfilters, frame)) < 0) {
@@ -1862,6 +1871,7 @@ static int video_thread(void *arg)
             last_w = frame->width;
             last_h = frame->height;
             last_format = frame->format;
+            last_serial = serial;
         }
 
         frame->pts = pts_int;
@@ -2037,7 +2047,7 @@ static int synchronize_audio(VideoState *is, int nb_samples)
 
         diff = get_audio_clock(is) - get_master_clock(is);
 
-        if (fabs(diff) < AV_NOSYNC_THRESHOLD) {
+        if (!isnan(diff) && fabs(diff) < AV_NOSYNC_THRESHOLD) {
             is->audio_diff_cum = diff + is->audio_diff_avg_coef * is->audio_diff_cum;
             if (is->audio_diff_avg_count < AUDIO_DIFF_AVG_NB) {
                 /* not enough measures to have a correct estimate */
@@ -2052,9 +2062,9 @@ static int synchronize_audio(VideoState *is, int nb_samples)
                     max_nb_samples = ((nb_samples * (100 + SAMPLE_CORRECTION_PERCENT_MAX) / 100));
                     wanted_nb_samples = FFMIN(FFMAX(wanted_nb_samples, min_nb_samples), max_nb_samples);
                 }
-                av_dlog(NULL, "diff=%f adiff=%f sample_diff=%d apts=%0.3f vpts=%0.3f %f\n",
+                av_dlog(NULL, "diff=%f adiff=%f sample_diff=%d apts=%0.3f %f\n",
                         diff, avg_diff, wanted_nb_samples - nb_samples,
-                        is->audio_clock, is->video_clock, is->audio_diff_threshold);
+                        is->audio_clock, is->audio_diff_threshold);
             }
         } else {
             /* too big difference : may be initial PTS errors, so
@@ -2067,8 +2077,14 @@ static int synchronize_audio(VideoState *is, int nb_samples)
     return wanted_nb_samples;
 }
 
-/* decode one audio frame and returns its uncompressed size */
-static int audio_decode_frame(VideoState *is, double *pts_ptr)
+/**
+ * Decode one audio frame and return its uncompressed size.
+ *
+ * The processed audio frame is decoded, converted if required, and
+ * stored in is->audio_buf, with size in bytes given by the return
+ * value.
+ */
+static int audio_decode_frame(VideoState *is)
 {
     AVPacket *pkt_temp = &is->audio_pkt_temp;
     AVPacket *pkt = &is->audio_pkt;
@@ -2076,7 +2092,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
     int len1, len2, data_size, resampled_data_size;
     int64_t dec_channel_layout;
     int got_frame;
-    double pts;
+    av_unused double audio_clock0;
     int new_packet = 0;
     int flush_complete = 0;
     int wanted_nb_samples;
@@ -2090,6 +2106,9 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
             } else
                 avcodec_get_frame_defaults(is->frame);
 
+            if (is->audioq.serial != is->audio_pkt_temp_serial)
+                break;
+
             if (is->paused)
                 return -1;
 
@@ -2112,13 +2131,13 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
                     flush_complete = 1;
                 continue;
             }
-            data_size = av_samples_get_buffer_size(NULL, is->frame->channels,
+            data_size = av_samples_get_buffer_size(NULL, av_frame_get_channels(is->frame),
                                                    is->frame->nb_samples,
                                                    is->frame->format, 1);
 
             dec_channel_layout =
-                (is->frame->channel_layout && is->frame->channels == av_get_channel_layout_nb_channels(is->frame->channel_layout)) ?
-                is->frame->channel_layout : av_get_default_channel_layout(is->frame->channels);
+                (is->frame->channel_layout && av_frame_get_channels(is->frame) == av_get_channel_layout_nb_channels(is->frame->channel_layout)) ?
+                is->frame->channel_layout : av_get_default_channel_layout(av_frame_get_channels(is->frame));
             wanted_nb_samples = synchronize_audio(is, is->frame->nb_samples);
 
             if (is->frame->format        != is->audio_src.fmt            ||
@@ -2132,20 +2151,21 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
                                                  0, NULL);
                 if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) {
                     fprintf(stderr, "Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n",
-                        is->frame->sample_rate,   av_get_sample_fmt_name(is->frame->format), (int)is->frame->channels,
-                        is->audio_tgt.freq, av_get_sample_fmt_name(is->audio_tgt.fmt), is->audio_tgt.channels);
+                            is->frame->sample_rate, av_get_sample_fmt_name(is->frame->format), av_frame_get_channels(is->frame),
+                            is->audio_tgt.freq, av_get_sample_fmt_name(is->audio_tgt.fmt), is->audio_tgt.channels);
                     break;
                 }
                 is->audio_src.channel_layout = dec_channel_layout;
-                is->audio_src.channels = is->frame->channels;
+                is->audio_src.channels       = av_frame_get_channels(is->frame);
                 is->audio_src.freq = is->frame->sample_rate;
                 is->audio_src.fmt = is->frame->format;
             }
 
             if (is->swr_ctx) {
                 const uint8_t **in = (const uint8_t **)is->frame->extended_data;
-                uint8_t *out[] = {is->audio_buf2};
-                int out_count = sizeof(is->audio_buf2) / is->audio_tgt.channels / av_get_bytes_per_sample(is->audio_tgt.fmt);
+                uint8_t **out = &is->audio_buf1;
+                int out_count = (int64_t)wanted_nb_samples * is->audio_tgt.freq / is->frame->sample_rate + 256;
+                int out_size  = av_samples_get_buffer_size(NULL, is->audio_tgt.channels, out_count, is->audio_tgt.fmt, 0);
                 if (wanted_nb_samples != is->frame->nb_samples) {
                     if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - is->frame->nb_samples) * is->audio_tgt.freq / is->frame->sample_rate,
                                                 wanted_nb_samples * is->audio_tgt.freq / is->frame->sample_rate) < 0) {
@@ -2153,6 +2173,9 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
                         break;
                     }
                 }
+                av_fast_malloc(&is->audio_buf1, &is->audio_buf1_size, out_size);
+                if (!is->audio_buf1)
+                    return AVERROR(ENOMEM);
                 len2 = swr_convert(is->swr_ctx, out, out_count, in, is->frame->nb_samples);
                 if (len2 < 0) {
                     fprintf(stderr, "swr_convert() failed\n");
@@ -2162,24 +2185,22 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
                     fprintf(stderr, "warning: audio buffer is probably too small\n");
                     swr_init(is->swr_ctx);
                 }
-                is->audio_buf = is->audio_buf2;
+                is->audio_buf = is->audio_buf1;
                 resampled_data_size = len2 * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt);
             } else {
                 is->audio_buf = is->frame->data[0];
                 resampled_data_size = data_size;
             }
 
-            /* if no pts, then compute it */
-            pts = is->audio_clock;
-            *pts_ptr = pts;
+            audio_clock0 = is->audio_clock;
             is->audio_clock += (double)data_size /
-                (is->frame->channels * is->frame->sample_rate * av_get_bytes_per_sample(is->frame->format));
+                (av_frame_get_channels(is->frame) * is->frame->sample_rate * av_get_bytes_per_sample(is->frame->format));
 #ifdef DEBUG
             {
                 static double last_clock;
-                printf("audio: delay=%0.3f clock=%0.3f pts=%0.3f\n",
+                printf("audio: delay=%0.3f clock=%0.3f clock0=%0.3f\n",
                        is->audio_clock - last_clock,
-                       is->audio_clock, pts);
+                       is->audio_clock, audio_clock0);
                 last_clock = is->audio_clock;
             }
 #endif
@@ -2191,7 +2212,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
             av_free_packet(pkt);
         memset(pkt_temp, 0, sizeof(*pkt_temp));
 
-        if (is->paused || is->audioq.abort_request) {
+        if (is->audioq.abort_request) {
             return -1;
         }
 
@@ -2212,6 +2233,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
         /* if update the audio clock with the pts */
         if (pkt->pts != AV_NOPTS_VALUE) {
             is->audio_clock = av_q2d(is->audio_st->time_base)*pkt->pts;
+            is->audio_clock_serial = is->audio_pkt_temp_serial;
         }
     }
 }
@@ -2223,13 +2245,12 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
     int audio_size, len1;
     int bytes_per_sec;
     int frame_size = av_samples_get_buffer_size(NULL, is->audio_tgt.channels, 1, is->audio_tgt.fmt, 1);
-    double pts;
 
     audio_callback_time = av_gettime();
 
     while (len > 0) {
         if (is->audio_buf_index >= is->audio_buf_size) {
-           audio_size = audio_decode_frame(is, &pts);
+           audio_size = audio_decode_frame(is);
            if (audio_size < 0) {
                 /* if error, just output silence */
                is->audio_buf      = is->silence_buf;
@@ -2254,7 +2275,7 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
     /* Let's assume the audio driver that is used by SDL has two periods. */
     is->audio_current_pts = is->audio_clock - (double)(2 * is->audio_hw_buf_size + is->audio_write_buf_size) / bytes_per_sec;
     is->audio_current_pts_drift = is->audio_current_pts - audio_callback_time / 1000000.0;
-    if (is->audioq.serial == is->audio_pkt_temp_serial)
+    if (is->audioq.serial == is->audio_clock_serial)
         check_external_clock_sync(is, is->audio_current_pts);
 }
 
@@ -2438,6 +2459,7 @@ static void stream_component_close(VideoState *is, int stream_index)
         av_free_packet(&is->audio_pkt);
         swr_free(&is->swr_ctx);
         av_freep(&is->audio_buf1);
+        is->audio_buf1_size = 0;
         is->audio_buf = NULL;
         avcodec_free_frame(&is->frame);
 
@@ -2614,7 +2636,7 @@ static int read_thread(void *arg)
                                 wanted_stream[AVMEDIA_TYPE_AUDIO],
                                 st_index[AVMEDIA_TYPE_VIDEO],
                                 NULL, 0);
-    if (!video_disable)
+    if (!video_disable && !subtitle_disable)
         st_index[AVMEDIA_TYPE_SUBTITLE] =
             av_find_best_stream(ic, AVMEDIA_TYPE_SUBTITLE,
                                 wanted_stream[AVMEDIA_TYPE_SUBTITLE],
@@ -2637,7 +2659,6 @@ static int read_thread(void *arg)
     if (st_index[AVMEDIA_TYPE_VIDEO] >= 0) {
         ret = stream_component_open(is, st_index[AVMEDIA_TYPE_VIDEO]);
     }
-    is->refresh_tid = SDL_CreateThread(refresh_thread, is);
     if (is->show_mode == SHOW_MODE_NONE)
         is->show_mode = ret >= 0 ? SHOW_MODE_VIDEO : SHOW_MODE_RDFT;
 
@@ -2698,14 +2719,15 @@ static int read_thread(void *arg)
                     packet_queue_put(&is->videoq, &flush_pkt);
                 }
                 if (is->seek_flags & AVSEEK_FLAG_BYTE) {
-                   //FIXME: use a cleaner way to signal obsolete external clock...
-                   update_external_clock_pts(is, (double)AV_NOPTS_VALUE);
+                   update_external_clock_pts(is, NAN);
                 } else {
                    update_external_clock_pts(is, seek_target / (double)AV_TIME_BASE);
                 }
             }
             is->seek_req = 0;
             eof = 0;
+            if (is->paused)
+                step_to_next_frame(is);
         }
         if (is->queue_attachments_req) {
             avformat_queue_attached_pictures(ic);
@@ -2833,11 +2855,12 @@ static VideoState *stream_open(const char *filename, AVInputFormat *iformat)
 
     is->continue_read_thread = SDL_CreateCond();
 
-    //FIXME: use a cleaner way to signal obsolete external clock...
-    update_external_clock_pts(is, (double)AV_NOPTS_VALUE);
+    update_external_clock_pts(is, NAN);
     update_external_clock_speed(is, 1.0);
     is->audio_current_pts_drift = -av_gettime() / 1000000.0;
     is->video_current_pts_drift = is->audio_current_pts_drift;
+    is->audio_clock_serial = -1;
+    is->video_clock_serial = -1;
     is->av_sync_type = av_sync_type;
     is->read_tid     = SDL_CreateThread(read_thread, is);
     if (!is->read_tid) {
@@ -2914,30 +2937,40 @@ static void toggle_full_screen(VideoState *is)
         is->pictq[i].reallocate = 1;
 #endif
     is_full_screen = !is_full_screen;
-    video_open(is, 1);
-}
-
-static void toggle_pause(VideoState *is)
-{
-    stream_toggle_pause(is);
-    is->step = 0;
-}
-
-static void step_to_next_frame(VideoState *is)
-{
-    /* if the stream is paused unpause it, then step */
-    if (is->paused)
-        stream_toggle_pause(is);
-    is->step = 1;
+    video_open(is, 1, NULL);
 }
 
 static void toggle_audio_display(VideoState *is)
 {
     int bgcolor = SDL_MapRGB(screen->format, 0x00, 0x00, 0x00);
-    is->show_mode = (is->show_mode + 1) % SHOW_MODE_NB;
-    fill_rectangle(screen,
-                is->xleft, is->ytop, is->width, is->height,
-                bgcolor, 1);
+    int next = is->show_mode;
+    do {
+        next = (next + 1) % SHOW_MODE_NB;
+    } while (next != is->show_mode && (next == SHOW_MODE_VIDEO && !is->video_st || next != SHOW_MODE_VIDEO && !is->audio_st));
+    if (is->show_mode != next) {
+        fill_rectangle(screen,
+                    is->xleft, is->ytop, is->width, is->height,
+                    bgcolor, 1);
+        is->force_refresh = 1;
+        is->show_mode = next;
+    }
+}
+
+static void refresh_loop_wait_event(VideoState *is, SDL_Event *event) {
+    double remaining_time = 0.0;
+    SDL_PumpEvents();
+    while (!SDL_PeepEvents(event, 1, SDL_GETEVENT, SDL_ALLEVENTS)) {
+        if (!cursor_hidden && av_gettime() - cursor_last_shown > CURSOR_HIDE_DELAY) {
+            SDL_ShowCursor(0);
+            cursor_hidden = 1;
+        }
+        if (remaining_time > 0.0)
+            av_usleep((int64_t)(remaining_time * 1000000.0));
+        remaining_time = REFRESH_RATE;
+        if (is->show_mode != SHOW_MODE_NONE && (!is->paused || is->force_refresh))
+            video_refresh(is, &remaining_time);
+        SDL_PumpEvents();
+    }
 }
 
 /* handle an event sent by the GUI */
@@ -2948,7 +2981,7 @@ static void event_loop(VideoState *cur_stream)
 
     for (;;) {
         double x;
-        SDL_WaitEvent(&event);
+        refresh_loop_wait_event(cur_stream, &event);
         switch (event.type) {
         case SDL_KEYDOWN:
             if (exit_on_keydown) {
@@ -2982,7 +3015,6 @@ 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;
@@ -3017,6 +3049,8 @@ static void event_loop(VideoState *cur_stream)
                         stream_seek(cur_stream, pos, incr, 1);
                     } else {
                         pos = get_master_clock(cur_stream);
+                        if (isnan(pos))
+                            pos = (double)cur_stream->seek_pos / AV_TIME_BASE;
                         pos += incr;
                         if (cur_stream->ic->start_time != AV_NOPTS_VALUE && pos < cur_stream->ic->start_time / (double)AV_TIME_BASE)
                             pos = cur_stream->ic->start_time / (double)AV_TIME_BASE;
@@ -3086,14 +3120,6 @@ static void event_loop(VideoState *cur_stream)
         case FF_ALLOC_EVENT:
             alloc_picture(event.user.data1);
             break;
-        case FF_REFRESH_EVENT:
-            if (!cursor_hidden && av_gettime() - cursor_last_shown > CURSOR_HIDE_DELAY) {
-                SDL_ShowCursor(0);
-                cursor_hidden = 1;
-            }
-            video_refresh(event.user.data1);
-            cur_stream->refresh = 0;
-            break;
         default:
             break;
         }
@@ -3212,6 +3238,7 @@ static const OptionDef options[] = {
     { "fs", OPT_BOOL, { &is_full_screen }, "force full screen" },
     { "an", OPT_BOOL, { &audio_disable }, "disable audio" },
     { "vn", OPT_BOOL, { &video_disable }, "disable video" },
+    { "sn", OPT_BOOL, { &subtitle_disable }, "disable subtitling" },
     { "ast", OPT_INT | HAS_ARG | OPT_EXPERT, { &wanted_stream[AVMEDIA_TYPE_AUDIO] }, "select desired audio stream", "stream_number" },
     { "vst", OPT_INT | HAS_ARG | OPT_EXPERT, { &wanted_stream[AVMEDIA_TYPE_VIDEO] }, "select desired video stream", "stream_number" },
     { "sst", OPT_INT | HAS_ARG | OPT_EXPERT, { &wanted_stream[AVMEDIA_TYPE_SUBTITLE] }, "select desired subtitle stream", "stream_number" },
@@ -3241,7 +3268,7 @@ static const OptionDef options[] = {
     { "infbuf", OPT_BOOL | OPT_EXPERT, { &infinite_buffer }, "don't limit the input buffer size (useful with realtime streams)", "" },
     { "window_title", OPT_STRING | HAS_ARG, { &window_title }, "set window title", "window title" },
 #if CONFIG_AVFILTER
-    { "vf", OPT_STRING | HAS_ARG, { &vfilters }, "video filters", "filter list" },
+    { "vf", OPT_STRING | HAS_ARG, { &vfilters }, "set video filters", "filter_graph" },
 #endif
     { "rdftspeed", OPT_INT | HAS_ARG| OPT_AUDIO | OPT_EXPERT, { &rdftspeed }, "rdft speed", "msecs" },
     { "showmode", HAS_ARG, { .func_arg = opt_show_mode}, "select show mode (0 = video, 1 = waves, 2 = RDFT)", "mode" },

ffprobe.c

@@ -203,6 +203,7 @@ static char *value_string(char *buf, int buf_size, struct unit_value uv)
                 vald /= pow(10, index * 3);
                 prefix_string = decimal_unit_prefixes[index];
             }
+            vali = vald;
         }
 
         if (show_float || (use_value_prefix && vald != (long long int)vald))
@@ -1472,11 +1473,11 @@ static void show_frame(WriterContext *w, AVFrame *frame, AVStream *stream,
     print_time("pkt_pts_time",          frame->pkt_pts, &stream->time_base);
     print_ts  ("pkt_dts",               frame->pkt_dts);
     print_time("pkt_dts_time",          frame->pkt_dts, &stream->time_base);
-    print_duration_ts  ("pkt_duration",      frame->pkt_duration);
-    print_duration_time("pkt_duration_time", frame->pkt_duration, &stream->time_base);
-    if (frame->pkt_pos != -1) print_fmt    ("pkt_pos", "%"PRId64, frame->pkt_pos);
+    print_duration_ts  ("pkt_duration",      av_frame_get_pkt_duration(frame));
+    print_duration_time("pkt_duration_time", av_frame_get_pkt_duration(frame), &stream->time_base);
+    if (av_frame_get_pkt_pos (frame) != -1) print_fmt    ("pkt_pos", "%"PRId64, av_frame_get_pkt_pos(frame));
     else                      print_str_opt("pkt_pos", "N/A");
-    if (frame->pkt_size != -1) print_fmt    ("pkt_size", "%d", av_frame_get_pkt_size(frame));
+    if (av_frame_get_pkt_size(frame) != -1) print_fmt    ("pkt_size", "%d", av_frame_get_pkt_size(frame));
     else                       print_str_opt("pkt_size", "N/A");
 
     switch (stream->codec->codec_type) {

ffserver.c

@@ -328,6 +328,14 @@ static AVLFG random_state;
 
 static FILE *logfile = NULL;
 
+static void htmlstrip(char *s) {
+    while (s && *s) {
+        s += strspn(s, "0123456789abcdefghijklmnopqrstuvwxyzABCDEFGHIJKLMNOPQRSTUVWXYZ,. ");
+        if (*s)
+            *s++ = '?';
+    }
+}
+
 static int64_t ffm_read_write_index(int fd)
 {
     uint8_t buf[8];
@@ -1125,7 +1133,7 @@ static int extract_rates(char *rates, int ratelen, const char *request)
         if (av_strncasecmp(p, "Pragma:", 7) == 0) {
             const char *q = p + 7;
 
-            while (*q && *q != '\n' && isspace(*q))
+            while (*q && *q != '\n' && av_isspace(*q))
                 q++;
 
             if (av_strncasecmp(q, "stream-switch-entry=", 20) == 0) {
@@ -1147,7 +1155,7 @@ static int extract_rates(char *rates, int ratelen, const char *request)
                     if (stream_no < ratelen && stream_no >= 0)
                         rates[stream_no] = rate_no;
 
-                    while (*q && *q != '\n' && !isspace(*q))
+                    while (*q && *q != '\n' && !av_isspace(*q))
                         q++;
                 }
 
@@ -1258,7 +1266,7 @@ static void get_word(char *buf, int buf_size, const char **pp)
     p = *pp;
     skip_spaces(&p);
     q = buf;
-    while (!isspace(*p) && *p != '\0') {
+    while (!av_isspace(*p) && *p != '\0') {
         if ((q - buf) < buf_size - 1)
             *q++ = *p;
         p++;
@@ -1275,7 +1283,7 @@ static void get_arg(char *buf, int buf_size, const char **pp)
     int quote;
 
     p = *pp;
-    while (isspace(*p)) p++;
+    while (av_isspace(*p)) p++;
     q = buf;
     quote = 0;
     if (*p == '\"' || *p == '\'')
@@ -1285,7 +1293,7 @@ static void get_arg(char *buf, int buf_size, const char **pp)
             if (*p == quote)
                 break;
         } else {
-            if (isspace(*p))
+            if (av_isspace(*p))
                 break;
         }
         if (*p == '\0')
@@ -1389,7 +1397,7 @@ static IPAddressACL* parse_dynamic_acl(FFStream *stream, HTTPContext *c)
             break;
         line_num++;
         p = line;
-        while (isspace(*p))
+        while (av_isspace(*p))
             p++;
         if (*p == '\0' || *p == '#')
             continue;
@@ -1540,7 +1548,7 @@ static int http_parse_request(HTTPContext *c)
     for (p = c->buffer; *p && *p != '\r' && *p != '\n'; ) {
         if (av_strncasecmp(p, "User-Agent:", 11) == 0) {
             useragent = p + 11;
-            if (*useragent && *useragent != '\n' && isspace(*useragent))
+            if (*useragent && *useragent != '\n' && av_isspace(*useragent))
                 useragent++;
             break;
         }
@@ -1668,7 +1676,7 @@ static int http_parse_request(HTTPContext *c)
             char *eoh;
             char hostbuf[260];
 
-            while (isspace(*hostinfo))
+            while (av_isspace(*hostinfo))
                 hostinfo++;
 
             eoh = strchr(hostinfo, '\n');
@@ -1887,6 +1895,7 @@ static int http_parse_request(HTTPContext *c)
  send_error:
     c->http_error = 404;
     q = c->buffer;
+    htmlstrip(msg);
     snprintf(q, c->buffer_size,
                   "HTTP/1.0 404 Not Found\r\n"
                   "Content-type: text/html\r\n"
@@ -4100,7 +4109,7 @@ static int parse_ffconfig(const char *filename)
             break;
         line_num++;
         p = line;
-        while (isspace(*p))
+        while (av_isspace(*p))
             p++;
         if (*p == '\0' || *p == '#')
             continue;
@@ -4237,7 +4246,7 @@ static int parse_ffconfig(const char *filename)
                 get_arg(arg, sizeof(arg), &p);
                 p1 = arg;
                 fsize = strtod(p1, &p1);
-                switch(toupper(*p1)) {
+                switch(av_toupper(*p1)) {
                 case 'K':
                     fsize *= 1024;
                     break;

libavcodec/012v.c

@@ -44,7 +44,7 @@ static av_cold int zero12v_decode_init(AVCodecContext *avctx)
 static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
                                 int *got_frame, AVPacket *avpkt)
 {
-    int line = 0;
+    int line = 0, ret;
     const int width = avctx->width;
     AVFrame *pic = avctx->coded_frame;
     uint16_t *y, *u, *v;
@@ -65,8 +65,8 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
     }
 
     pic->reference = 0;
-    if (ff_get_buffer(avctx, pic) < 0)
-        return AVERROR_INVALIDDATA;;
+    if ((ret = ff_get_buffer(avctx, pic)) < 0)
+        return ret;
 
     y = (uint16_t *)pic->data[0];
     u = (uint16_t *)pic->data[1];

libavcodec/4xm.c

@@ -130,7 +130,7 @@ typedef struct CFrameBuffer {
 typedef struct FourXContext {
     AVCodecContext *avctx;
     DSPContext dsp;
-    AVFrame current_picture, last_picture;
+    AVFrame *current_picture, *last_picture;
     GetBitContext pre_gb;          ///< ac/dc prefix
     GetBitContext gb;
     GetByteContext g;
@@ -138,7 +138,7 @@ typedef struct FourXContext {
     int mv[256];
     VLC pre_vlc;
     int last_dc;
-    DECLARE_ALIGNED(16, DCTELEM, block)[6][64];
+    DECLARE_ALIGNED(16, int16_t, block)[6][64];
     void *bitstream_buffer;
     unsigned int bitstream_buffer_size;
     int version;
@@ -153,7 +153,7 @@ typedef struct FourXContext {
 
 #define MULTIPLY(var, const) (((var) * (const)) >> 16)
 
-static void idct(DCTELEM block[64])
+static void idct(int16_t block[64])
 {
     int tmp0, tmp1, tmp2, tmp3, tmp4, tmp5, tmp6, tmp7;
     int tmp10, tmp11, tmp12, tmp13;
@@ -261,9 +261,9 @@ static void init_mv(FourXContext *f)
 
     for (i = 0; i < 256; i++) {
         if (f->version > 1)
-            f->mv[i] = mv[i][0] + mv[i][1] * f->current_picture.linesize[0] / 2;
+            f->mv[i] = mv[i][0] + mv[i][1] * f->current_picture->linesize[0] / 2;
         else
-            f->mv[i] = (i & 15) - 8 + ((i >> 4) - 8) * f->current_picture.linesize[0] / 2;
+            f->mv[i] = (i & 15) - 8 + ((i >> 4) - 8) * f->current_picture->linesize[0] / 2;
     }
 }
 
@@ -340,7 +340,7 @@ static void decode_p_block(FourXContext *f, uint16_t *dst, uint16_t *src,
     int code        = get_vlc2(&f->gb,
                                block_type_vlc[1 - (f->version > 1)][index].table,
                                BLOCK_TYPE_VLC_BITS, 1);
-    uint16_t *start = (uint16_t *)f->last_picture.data[0];
+    uint16_t *start = (uint16_t *)f->last_picture->data[0];
     uint16_t *end   = start + stride * (f->avctx->height - h + 1) - (1 << log2w);
 
     av_assert2(code >= 0 && code <= 6);
@@ -409,9 +409,9 @@ static int decode_p_frame(FourXContext *f, const uint8_t *buf, int length)
     int x, y;
     const int width  = f->avctx->width;
     const int height = f->avctx->height;
-    uint16_t *src    = (uint16_t *)f->last_picture.data[0];
-    uint16_t *dst    = (uint16_t *)f->current_picture.data[0];
-    const int stride =             f->current_picture.linesize[0] >> 1;
+    uint16_t *src    = (uint16_t *)f->last_picture->data[0];
+    uint16_t *dst    = (uint16_t *)f->current_picture->data[0];
+    const int stride =             f->current_picture->linesize[0] >> 1;
     unsigned int bitstream_size, bytestream_size, wordstream_size, extra,
                  bytestream_offset, wordstream_offset;
 
@@ -471,7 +471,7 @@ static int decode_p_frame(FourXContext *f, const uint8_t *buf, int length)
  * decode block and dequantize.
  * Note this is almost identical to MJPEG.
  */
-static int decode_i_block(FourXContext *f, DCTELEM *block)
+static int decode_i_block(FourXContext *f, int16_t *block)
 {
     int code, i, j, level, val;
 
@@ -521,10 +521,10 @@ static int decode_i_block(FourXContext *f, DCTELEM *block)
 
 static inline void idct_put(FourXContext *f, int x, int y)
 {
-    DCTELEM (*block)[64] = f->block;
-    int stride           = f->current_picture.linesize[0] >> 1;
+    int16_t (*block)[64] = f->block;
+    int stride           = f->current_picture->linesize[0] >> 1;
     int i;
-    uint16_t *dst = ((uint16_t*)f->current_picture.data[0]) + y * stride + x;
+    uint16_t *dst = ((uint16_t*)f->current_picture->data[0]) + y * stride + x;
 
     for (i = 0; i < 4; i++) {
         block[i][0] += 0x80 * 8 * 8;
@@ -542,7 +542,7 @@ static inline void idct_put(FourXContext *f, int x, int y)
      * cr = (-1b - 4g + 5r) / 14 */
     for (y = 0; y < 8; y++) {
         for (x = 0; x < 8; x++) {
-            DCTELEM *temp = block[(x >> 2) + 2 * (y >> 2)] +
+            int16_t *temp = block[(x >> 2) + 2 * (y >> 2)] +
                             2 * (x & 3) + 2 * 8 * (y & 3); // FIXME optimize
             int cb = block[4][x + 8 * y];
             int cr = block[5][x + 8 * y];
@@ -599,8 +599,10 @@ static const uint8_t *read_huffman_tables(FourXContext *f,
     for (;;) {
         int i;
 
-        if (start <= end && ptr_end - ptr < end - start + 1 + 1)
+        if (ptr_end - ptr < FFMAX(end - start + 1, 0) + 1) {
+            av_log(f->avctx, AV_LOG_ERROR, "invalid data in read_huffman_tables\n");
             return NULL;
+        }
         for (i = start; i <= end; i++)
             frequency[i] = *ptr++;
         start = *ptr++;
@@ -614,6 +616,11 @@ static const uint8_t *read_huffman_tables(FourXContext *f,
     while ((ptr - buf) & 3)
         ptr++; // 4byte align
 
+    if (ptr > ptr_end) {
+        av_log(f->avctx, AV_LOG_ERROR, "ptr overflow in read_huffman_tables\n");
+        return NULL;
+    }
+
     for (j = 257; j < 512; j++) {
         int min_freq[2] = { 256 * 256, 256 * 256 };
         int smallest[2] = { 0, 0 };
@@ -681,8 +688,8 @@ static int decode_i2_frame(FourXContext *f, const uint8_t *buf, int length)
     const int width  = f->avctx->width;
     const int height = f->avctx->height;
     const int mbs    = (FFALIGN(width, 16) >> 4) * (FFALIGN(height, 16) >> 4);
-    uint16_t *dst    = (uint16_t*)f->current_picture.data[0];
-    const int stride =            f->current_picture.linesize[0]>>1;
+    uint16_t *dst    = (uint16_t*)f->current_picture->data[0];
+    const int stride =            f->current_picture->linesize[0]>>1;
     const uint8_t *buf_end = buf + length;
     GetByteContext g3;
 
@@ -750,8 +757,12 @@ static int decode_i_frame(FourXContext *f, const uint8_t *buf, int length)
     }
 
     prestream = read_huffman_tables(f, prestream, buf + length - prestream);
-    if (!prestream)
-        return -1;
+    if (!prestream) {
+        av_log(f->avctx, AV_LOG_ERROR, "Error reading Huffman tables.\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    av_assert0(prestream <= buf + length);
 
     init_get_bits(&f->gb, buf + 4, 8 * bitstream_size);
 
@@ -867,9 +878,9 @@ static int decode_frame(AVCodecContext *avctx, void *data,
         frame_size = buf_size - 12;
     }
 
-    FFSWAP(AVFrame, f->current_picture, f->last_picture);
+    FFSWAP(AVFrame*, f->current_picture, f->last_picture);
 
-    p                  = &f->current_picture;
+    p                  = f->current_picture;
     avctx->coded_frame = p;
 
     // alternatively we would have to use our own buffer management
@@ -894,14 +905,14 @@ static int decode_frame(AVCodecContext *avctx, void *data,
             return ret;
         }
     } else if (frame_4cc == AV_RL32("pfrm") || frame_4cc == AV_RL32("pfr2")) {
-        if (!f->last_picture.data[0]) {
-            f->last_picture.reference = 3;
-            if ((ret = ff_get_buffer(avctx, &f->last_picture)) < 0) {
+        if (!f->last_picture->data[0]) {
+            f->last_picture->reference = 3;
+            if ((ret = ff_get_buffer(avctx, f->last_picture)) < 0) {
                 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
                 return ret;
             }
             for (i=0; i<avctx->height; i++)
-                memset(f->last_picture.data[0] + i*f->last_picture.linesize[0], 0, 2*avctx->width);
+                memset(f->last_picture->data[0] + i*f->last_picture->linesize[0], 0, 2*avctx->width);
         }
 
         p->pict_type = AV_PICTURE_TYPE_P;
@@ -927,16 +938,6 @@ static int decode_frame(AVCodecContext *avctx, void *data,
     return buf_size;
 }
 
-
-static av_cold void common_init(AVCodecContext *avctx)
-{
-    FourXContext * const f = avctx->priv_data;
-
-    ff_dsputil_init(&f->dsp, avctx);
-
-    f->avctx = avctx;
-}
-
 static av_cold int decode_init(AVCodecContext *avctx)
 {
     FourXContext * const f = avctx->priv_data;
@@ -950,10 +951,9 @@ static av_cold int decode_init(AVCodecContext *avctx)
         return AVERROR_INVALIDDATA;
     }
 
-    avcodec_get_frame_defaults(&f->current_picture);
-    avcodec_get_frame_defaults(&f->last_picture);
     f->version = AV_RL32(avctx->extradata) >> 16;
-    common_init(avctx);
+    ff_dsputil_init(&f->dsp, avctx);
+    f->avctx = avctx;
     init_vlcs(f);
 
     if (f->version > 2)
@@ -961,6 +961,14 @@ static av_cold int decode_init(AVCodecContext *avctx)
     else
         avctx->pix_fmt = AV_PIX_FMT_BGR555;
 
+    f->current_picture = avcodec_alloc_frame();
+    f->last_picture    = avcodec_alloc_frame();
+    if (!f->current_picture || !f->last_picture) {
+        avcodec_free_frame(&f->current_picture);
+        avcodec_free_frame(&f->last_picture);
+        return AVERROR(ENOMEM);
+    }
+
     return 0;
 }
 
@@ -977,10 +985,12 @@ static av_cold int decode_end(AVCodecContext *avctx)
         f->cfrm[i].allocated_size = 0;
     }
     ff_free_vlc(&f->pre_vlc);
-    if (f->current_picture.data[0])
-        avctx->release_buffer(avctx, &f->current_picture);
-    if (f->last_picture.data[0])
-        avctx->release_buffer(avctx, &f->last_picture);
+    if (f->current_picture->data[0])
+        avctx->release_buffer(avctx, f->current_picture);
+    if (f->last_picture->data[0])
+        avctx->release_buffer(avctx, f->last_picture);
+    avcodec_free_frame(&f->current_picture);
+    avcodec_free_frame(&f->last_picture);
 
     return 0;
 }

libavcodec/8bps.c

@@ -68,15 +68,16 @@ static int decode_frame(AVCodecContext *avctx, void *data,
     unsigned char count;
     unsigned int planes     = c->planes;
     unsigned char *planemap = c->planemap;
+    int ret;
 
     if (c->pic.data[0])
         avctx->release_buffer(avctx, &c->pic);
 
     c->pic.reference    = 0;
     c->pic.buffer_hints = FF_BUFFER_HINTS_VALID;
-    if (ff_get_buffer(avctx, &c->pic) < 0){
+    if ((ret = ff_get_buffer(avctx, &c->pic)) < 0) {
         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
-        return -1;
+        return ret;
     }
 
     /* Set data pointer after line lengths */
@@ -96,14 +97,14 @@ static int decode_frame(AVCodecContext *avctx, void *data,
             /* Decode a row of this plane */
             while (dlen > 0) {
                 if (dp + 1 >= buf + buf_size)
-                    return -1;
+                    return AVERROR_INVALIDDATA;
                 if ((count = *dp++) <= 127) {
                     count++;
                     dlen -= count + 1;
                     if (pixptr + count * planes > pixptr_end)
                         break;
                     if (dp + count > buf + buf_size)
-                        return -1;
+                        return AVERROR_INVALIDDATA;
                     while (count--) {
                         *pixptr = *dp++;
                         pixptr += planes;
@@ -181,7 +182,7 @@ static av_cold int decode_init(AVCodecContext *avctx)
     default:
         av_log(avctx, AV_LOG_ERROR, "Error: Unsupported color depth: %u.\n",
                avctx->bits_per_coded_sample);
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
 
     return 0;

libavcodec/8svx.c

@@ -44,7 +44,6 @@
 
 /** decoder context */
 typedef struct EightSvxContext {
-    AVFrame frame;
     uint8_t fib_acc[2];
     const int8_t *table;
 
@@ -88,6 +87,7 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
                                  int *got_frame_ptr, AVPacket *avpkt)
 {
     EightSvxContext *esc = avctx->priv_data;
+    AVFrame *frame       = data;
     int buf_size;
     int ch, ret;
     int hdr_size = 2;
@@ -135,21 +135,20 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
     }
 
     /* get output buffer */
-    esc->frame.nb_samples = buf_size * 2;
-    if ((ret = ff_get_buffer(avctx, &esc->frame)) < 0) {
+    frame->nb_samples = buf_size * 2;
+    if ((ret = ff_get_buffer(avctx, frame)) < 0) {
         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
         return ret;
     }
 
     for (ch = 0; ch < avctx->channels; ch++) {
-        delta_decode(esc->frame.data[ch], &esc->data[ch][esc->data_idx],
+        delta_decode(frame->data[ch], &esc->data[ch][esc->data_idx],
                      buf_size, &esc->fib_acc[ch], esc->table);
     }
 
     esc->data_idx += buf_size;
 
-    *got_frame_ptr   = 1;
-    *(AVFrame *)data = esc->frame;
+    *got_frame_ptr = 1;
 
     return ((avctx->frame_number == 0)*hdr_size + buf_size)*avctx->channels;
 }
@@ -172,9 +171,6 @@ static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
     }
     avctx->sample_fmt = AV_SAMPLE_FMT_U8P;
 
-    avcodec_get_frame_defaults(&esc->frame);
-    avctx->coded_frame = &esc->frame;
-
     return 0;
 }
 

libavcodec/Makefile

@@ -25,7 +25,6 @@ OBJS = allcodecs.o                                                      \
        fmtconvert.o                                                     \
        imgconvert.o                                                     \
        jrevdct.o                                                        \
-       log2_tab.o                                                       \
        mathtables.o                                                     \
        options.o                                                        \
        parser.o                                                         \
@@ -39,9 +38,9 @@ OBJS = allcodecs.o                                                      \
 # parts needed for many different codecs
 OBJS-$(CONFIG_AANDCTTABLES)            += aandcttab.o
 OBJS-$(CONFIG_AC3DSP)                  += ac3dsp.o
+OBJS-$(CONFIG_AUDIO_FRAME_QUEUE)       += audio_frame_queue.o
 OBJS-$(CONFIG_CRYSTALHD)               += crystalhd.o
 OBJS-$(CONFIG_DCT)                     += dct.o dct32_fixed.o dct32_float.o
-OBJS-$(CONFIG_DWT)                     += dwt.o snow.o
 OBJS-$(CONFIG_DXVA2)                   += dxva2.o
 OBJS-$(CONFIG_ENCODERS)                += faandct.o jfdctfst.o jfdctint.o
 OBJS-$(CONFIG_ERROR_RESILIENCE)        += error_resilience.o
@@ -49,8 +48,10 @@ FFT-OBJS-$(CONFIG_HARDCODED_TABLES)    += cos_tables.o cos_fixed_tables.o
 OBJS-$(CONFIG_FFT)                     += avfft.o fft_fixed.o fft_float.o \
                                           $(FFT-OBJS-yes)
 OBJS-$(CONFIG_GOLOMB)                  += golomb.o
+OBJS-$(CONFIG_H264CHROMA)              += h264chroma.o
 OBJS-$(CONFIG_H264DSP)                 += h264dsp.o h264idct.o
 OBJS-$(CONFIG_H264PRED)                += h264pred.o
+OBJS-$(CONFIG_H264QPEL)                += h264qpel.o
 OBJS-$(CONFIG_HUFFMAN)                 += huffman.o
 OBJS-$(CONFIG_LIBXVID)                 += libxvid_rc.o
 OBJS-$(CONFIG_LPC)                     += lpc.o
@@ -68,6 +69,7 @@ OBJS-$(CONFIG_MPEGVIDEOENC)            += mpegvideo_enc.o mpeg12data.o  \
 OBJS-$(CONFIG_RANGECODER)              += rangecoder.o
 RDFT-OBJS-$(CONFIG_HARDCODED_TABLES)   += sin_tables.o
 OBJS-$(CONFIG_RDFT)                    += rdft.o $(RDFT-OBJS-yes)
+OBJS-$(CONFIG_SHARED)                  += log2_tab.o
 OBJS-$(CONFIG_SINEWIN)                 += sinewin.o
 OBJS-$(CONFIG_VAAPI)                   += vaapi.o
 OBJS-$(CONFIG_VDPAU)                   += vdpau.o
@@ -84,8 +86,7 @@ OBJS-$(CONFIG_AAC_DECODER)             += aacdec.o aactab.o aacsbr.o aacps.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 \
@@ -158,7 +159,7 @@ OBJS-$(CONFIG_DCA_DECODER)             += dcadec.o dca.o dcadsp.o      \
                                           dca_parser.o synth_filter.o
 OBJS-$(CONFIG_DCA_ENCODER)             += dcaenc.o dca.o
 OBJS-$(CONFIG_DIRAC_DECODER)           += diracdec.o dirac.o diracdsp.o \
-                                          dirac_arith.o mpeg12data.o dwt.o
+                                          dirac_arith.o mpeg12data.o dirac_dwt.o
 OBJS-$(CONFIG_DFA_DECODER)             += dfa.o
 OBJS-$(CONFIG_DNXHD_DECODER)           += dnxhddec.o dnxhddata.o
 OBJS-$(CONFIG_DNXHD_ENCODER)           += dnxhdenc.o dnxhddata.o
@@ -188,6 +189,7 @@ OBJS-$(CONFIG_EIGHTSVX_EXP_DECODER)    += 8svx.o
 OBJS-$(CONFIG_EIGHTSVX_FIB_DECODER)    += 8svx.o
 OBJS-$(CONFIG_ESCAPE124_DECODER)       += escape124.o
 OBJS-$(CONFIG_ESCAPE130_DECODER)       += escape130.o
+OBJS-$(CONFIG_EVRC_DECODER)            += evrcdec.o acelp_vectors.o lsp.o
 OBJS-$(CONFIG_EXR_DECODER)             += exr.o
 OBJS-$(CONFIG_FFV1_DECODER)            += ffv1dec.o ffv1.o
 OBJS-$(CONFIG_FFV1_ENCODER)            += ffv1enc.o ffv1.o
@@ -218,6 +220,7 @@ OBJS-$(CONFIG_H263_DECODER)            += h263dec.o h263.o ituh263dec.o        \
                                           mpeg4video.o mpeg4videodec.o flvdec.o\
                                           intelh263dec.o
 OBJS-$(CONFIG_H263_VAAPI_HWACCEL)      += vaapi_mpeg4.o
+OBJS-$(CONFIG_H263_VDPAU_HWACCEL)      += vdpau_mpeg4.o
 OBJS-$(CONFIG_H263_ENCODER)            += mpeg4videoenc.o mpeg4video.o  \
                                           h263.o ituh263enc.o flvenc.o
 OBJS-$(CONFIG_H264_DECODER)            += h264.o                               \
@@ -228,6 +231,7 @@ OBJS-$(CONFIG_H264_DXVA2_HWACCEL)      += dxva2_h264.o
 OBJS-$(CONFIG_H264_VAAPI_HWACCEL)      += vaapi_h264.o
 OBJS-$(CONFIG_H264_VDA_HWACCEL)        += vda_h264.o
 OBJS-$(CONFIG_H264_VDA_DECODER)        += vda_h264_dec.o
+OBJS-$(CONFIG_H264_VDPAU_HWACCEL)      += vdpau_h264.o
 OBJS-$(CONFIG_HUFFYUV_DECODER)         += huffyuv.o huffyuvdec.o
 OBJS-$(CONFIG_HUFFYUV_ENCODER)         += huffyuv.o huffyuvenc.o
 OBJS-$(CONFIG_IAC_DECODER)             += imc.o
@@ -284,15 +288,18 @@ OBJS-$(CONFIG_MPC8_DECODER)            += mpc8.o mpc.o
 OBJS-$(CONFIG_MPEGVIDEO_DECODER)       += mpeg12.o mpeg12data.o \
                                           mpegvideo.o error_resilience.o
 OBJS-$(CONFIG_MPEG_XVMC_DECODER)       += mpegvideo_xvmc.o
+OBJS-$(CONFIG_MPEG1_VDPAU_HWACCEL)     += vdpau_mpeg12.o
 OBJS-$(CONFIG_MPEG1VIDEO_DECODER)      += mpeg12.o mpeg12data.o
 OBJS-$(CONFIG_MPEG1VIDEO_ENCODER)      += mpeg12enc.o mpeg12.o          \
                                           timecode.o
 OBJS-$(CONFIG_MPEG2_DXVA2_HWACCEL)     += dxva2_mpeg2.o
 OBJS-$(CONFIG_MPEG2_VAAPI_HWACCEL)     += vaapi_mpeg2.o
+OBJS-$(CONFIG_MPEG2_VDPAU_HWACCEL)     += vdpau_mpeg12.o
 OBJS-$(CONFIG_MPEG2VIDEO_DECODER)      += mpeg12.o mpeg12data.o
 OBJS-$(CONFIG_MPEG2VIDEO_ENCODER)      += mpeg12enc.o mpeg12.o          \
                                           timecode.o
 OBJS-$(CONFIG_MPEG4_VAAPI_HWACCEL)     += vaapi_mpeg4.o
+OBJS-$(CONFIG_MPEG4_VDPAU_HWACCEL)     += vdpau_mpeg4.o
 OBJS-$(CONFIG_MPL2_DECODER)            += mpl2dec.o ass.o
 OBJS-$(CONFIG_MSMPEG4V1_DECODER)       += msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_MSMPEG4V2_DECODER)       += msmpeg4.o msmpeg4data.o h263dec.o \
@@ -317,8 +324,7 @@ OBJS-$(CONFIG_MVC1_DECODER)            += mvcdec.o
 OBJS-$(CONFIG_MVC2_DECODER)            += mvcdec.o
 OBJS-$(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-$(CONFIG_PAF_VIDEO_DECODER)       += paf.o
 OBJS-$(CONFIG_PAF_AUDIO_DECODER)       += paf.o
@@ -358,8 +364,7 @@ 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-$(CONFIG_RA_288_DECODER)          += ra288.o celp_filters.o
 OBJS-$(CONFIG_RALF_DECODER)            += ralf.o
 OBJS-$(CONFIG_RAWVIDEO_DECODER)        += rawdec.o
@@ -391,8 +396,8 @@ OBJS-$(CONFIG_SIPR_DECODER)            += sipr.o acelp_pitch_delay.o \
 OBJS-$(CONFIG_SMACKAUD_DECODER)        += smacker.o
 OBJS-$(CONFIG_SMACKER_DECODER)         += smacker.o
 OBJS-$(CONFIG_SMC_DECODER)             += smc.o
-OBJS-$(CONFIG_SNOW_DECODER)            += snowdec.o snow.o
-OBJS-$(CONFIG_SNOW_ENCODER)            += snowenc.o snow.o              \
+OBJS-$(CONFIG_SNOW_DECODER)            += snowdec.o snow.o snow_dwt.o
+OBJS-$(CONFIG_SNOW_ENCODER)            += snowenc.o snow.o snow_dwt.o             \
                                           h263.o ituh263enc.o
 OBJS-$(CONFIG_SOL_DPCM_DECODER)        += dpcm.o
 OBJS-$(CONFIG_SONIC_DECODER)           += sonic.o
@@ -453,12 +458,12 @@ OBJS-$(CONFIG_VC1_DECODER)             += vc1dec.o vc1.o vc1data.o vc1dsp.o \
                                           intrax8.o intrax8dsp.o
 OBJS-$(CONFIG_VC1_DXVA2_HWACCEL)       += dxva2_vc1.o
 OBJS-$(CONFIG_VC1_VAAPI_HWACCEL)       += vaapi_vc1.o
+OBJS-$(CONFIG_VC1_VDPAU_HWACCEL)       += vdpau_vc1.o
 OBJS-$(CONFIG_VCR1_DECODER)            += vcr1.o
-OBJS-$(CONFIG_VCR1_ENCODER)            += vcr1.o
 OBJS-$(CONFIG_VMDAUDIO_DECODER)        += vmdav.o
 OBJS-$(CONFIG_VMDVIDEO_DECODER)        += vmdav.o
 OBJS-$(CONFIG_VMNC_DECODER)            += vmnc.o
-OBJS-$(CONFIG_VORBIS_DECODER)          += vorbisdec.o vorbis.o \
+OBJS-$(CONFIG_VORBIS_DECODER)          += vorbisdec.o vorbisdsp.o vorbis.o \
                                           vorbis_data.o xiph.o
 OBJS-$(CONFIG_VORBIS_ENCODER)          += vorbisenc.o vorbis.o \
                                           vorbis_data.o
@@ -482,10 +487,10 @@ OBJS-$(CONFIG_WMAVOICE_DECODER)        += wmavoice.o \
                                           celp_filters.o \
                                           acelp_vectors.o acelp_filters.o
 OBJS-$(CONFIG_WMV1_DECODER)            += msmpeg4.o msmpeg4data.o
-OBJS-$(CONFIG_WMV2_DECODER)            += wmv2dec.o wmv2.o        \
+OBJS-$(CONFIG_WMV2_DECODER)            += wmv2dec.o wmv2.o wmv2dsp.o \
                                           msmpeg4.o msmpeg4data.o \
                                           intrax8.o intrax8dsp.o
-OBJS-$(CONFIG_WMV2_ENCODER)            += wmv2enc.o wmv2.o \
+OBJS-$(CONFIG_WMV2_ENCODER)            += wmv2enc.o wmv2.o wmv2dsp.o \
                                           msmpeg4.o msmpeg4enc.o msmpeg4data.o \
                                           mpeg4videodec.o ituh263dec.o h263dec.o
 OBJS-$(CONFIG_WNV1_DECODER)            += wnv1.o
@@ -660,46 +665,44 @@ 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-$(CONFIG_LIBFDK_AAC_ENCODER)         += libfdk-aacenc.o audio_frame_queue.o
+OBJS-$(CONFIG_LIBFAAC_ENCODER)            += libfaac.o
+OBJS-$(CONFIG_LIBFDK_AAC_ENCODER)         += libfdk-aacenc.o
 OBJS-$(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_LIBILBC_DECODER)            += libilbc.o
 OBJS-$(CONFIG_LIBILBC_ENCODER)            += libilbc.o
-OBJS-$(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-$(CONFIG_LIBMP3LAME_ENCODER)         += libmp3lame.o mpegaudiodecheader.o
+OBJS-$(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-$(CONFIG_LIBOPUS_DECODER)            += libopusdec.o libopus.o     \
                                              vorbis_data.o
 OBJS-$(CONFIG_LIBOPUS_ENCODER)            += libopusenc.o libopus.o     \
-                                             vorbis_data.o audio_frame_queue.o
+                                             vorbis_data.o
 OBJS-$(CONFIG_LIBSCHROEDINGER_DECODER)    += libschroedingerdec.o \
                                              libschroedinger.o
 OBJS-$(CONFIG_LIBSCHROEDINGER_ENCODER)    += libschroedingerenc.o \
                                              libschroedinger.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_LIBTWOLAME_ENCODER)         += libtwolame.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_LIBVO_AACENC_ENCODER)       += libvo-aacenc.o mpeg4audio.o
 OBJS-$(CONFIG_LIBVO_AMRWBENC_ENCODER)     += libvo-amrwbenc.o
 OBJS-$(CONFIG_LIBVORBIS_DECODER)          += libvorbisdec.o
-OBJS-$(CONFIG_LIBVORBIS_ENCODER)          += libvorbisenc.o audio_frame_queue.o \
+OBJS-$(CONFIG_LIBVORBIS_ENCODER)          += libvorbisenc.o \
                                              vorbis_data.o vorbis_parser.o xiph.o
-OBJS-$(CONFIG_LIBVPX_DECODER)             += libvpxdec.o
-OBJS-$(CONFIG_LIBVPX_ENCODER)             += libvpxenc.o
+OBJS-$(CONFIG_LIBVPX_VP8_DECODER)         += libvpxdec.o
+OBJS-$(CONFIG_LIBVPX_VP8_ENCODER)         += libvpxenc.o
+OBJS-$(CONFIG_LIBVPX_VP9_DECODER)         += libvpxdec.o
+OBJS-$(CONFIG_LIBVPX_VP9_ENCODER)         += libvpxenc.o
 OBJS-$(CONFIG_LIBX264_ENCODER)            += libx264.o
 OBJS-$(CONFIG_LIBXAVS_ENCODER)            += libxavs.o
 OBJS-$(CONFIG_LIBXVID_ENCODER)            += libxvid.o

libavcodec/aac.h

@@ -32,7 +32,6 @@
 
 #include "libavutil/float_dsp.h"
 #include "avcodec.h"
-#include "dsputil.h"
 #include "fft.h"
 #include "mpeg4audio.h"
 #include "sbr.h"
@@ -82,7 +81,7 @@ enum BandType {
     INTENSITY_BT   = 15,    ///< Scalefactor data are intensity stereo positions.
 };
 
-#define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10)
+#define IS_CODEBOOK_UNSIGNED(x) (((x) - 1) & 10)
 
 enum ChannelPosition {
     AAC_CHANNEL_OFF   = 0,
@@ -260,10 +259,10 @@ typedef struct ChannelElement {
 /**
  * main AAC context
  */
-typedef struct AACContext {
+struct AACContext {
     AVClass        *class;
     AVCodecContext *avctx;
-    AVFrame frame;
+    AVFrame *frame;
 
     int is_saved;                 ///< Set if elements have stored overlap from previous frame.
     DynamicRangeControl che_drc;
@@ -292,7 +291,6 @@ typedef struct AACContext {
     FFTContext mdct;
     FFTContext mdct_small;
     FFTContext mdct_ltp;
-    DSPContext dsp;
     FmtConvertContext fmt_conv;
     AVFloatDSPContext fdsp;
     int random_state;
@@ -318,6 +316,18 @@ typedef struct AACContext {
 
     OutputConfiguration oc[2];
     int warned_num_aac_frames;
-} AACContext;
+
+    /* aacdec functions pointers */
+    void (*imdct_and_windowing)(AACContext *ac, SingleChannelElement *sce);
+    void (*apply_ltp)(AACContext *ac, SingleChannelElement *sce);
+    void (*apply_tns)(float coef[1024], TemporalNoiseShaping *tns,
+                      IndividualChannelStream *ics, int decode);
+    void (*windowing_and_mdct_ltp)(AACContext *ac, float *out,
+                                   float *in, IndividualChannelStream *ics);
+    void (*update_ltp)(AACContext *ac, SingleChannelElement *sce);
+
+};
+
+void ff_aacdec_init_mips(AACContext *c);
 
 #endif /* AVCODEC_AAC_H */

libavcodec/aaccoder.c

@@ -710,7 +710,7 @@ 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;
+    int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->channels * (lambda / 120.f);
     float dists[128] = { 0 }, uplims[128];
     float maxvals[128];
     int fflag, minscaler;

libavcodec/aacdec.c

@@ -84,7 +84,6 @@
 #include "avcodec.h"
 #include "internal.h"
 #include "get_bits.h"
-#include "dsputil.h"
 #include "fft.h"
 #include "fmtconvert.h"
 #include "lpc.h"
@@ -108,11 +107,17 @@
 
 #if ARCH_ARM
 #   include "arm/aac.h"
+#elif ARCH_MIPS
+#   include "mips/aacdec_mips.h"
 #endif
 
 static VLC vlc_scalefactors;
 static VLC vlc_spectral[11];
 
+static int output_configure(AACContext *ac,
+                            uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
+                            enum OCStatus oc_type, int get_new_frame);
+
 #define overread_err "Input buffer exhausted before END element found\n"
 
 static int count_channels(uint8_t (*layout)[3], int tags)
@@ -184,9 +189,12 @@ static int frame_configure_elements(AVCodecContext *avctx)
         }
     }
 
+    if (!avctx->channels)
+        return 1;
+
     /* get output buffer */
-    ac->frame.nb_samples = 2048;
-    if ((ret = ff_get_buffer(avctx, &ac->frame)) < 0) {
+    ac->frame->nb_samples = 2048;
+    if ((ret = ff_get_buffer(avctx, ac->frame)) < 0) {
         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
         return ret;
     }
@@ -194,7 +202,7 @@ static int frame_configure_elements(AVCodecContext *avctx)
     /* map output channel pointers to AVFrame data */
     for (ch = 0; ch < avctx->channels; ch++) {
         if (ac->output_element[ch])
-            ac->output_element[ch]->ret = (float *)ac->frame.extended_data[ch];
+            ac->output_element[ch]->ret = (float *)ac->frame->extended_data[ch];
     }
 
     return 0;
@@ -406,6 +414,8 @@ static void pop_output_configuration(AACContext *ac) {
         ac->oc[1] = ac->oc[0];
         ac->avctx->channels = ac->oc[1].channels;
         ac->avctx->channel_layout = ac->oc[1].channel_layout;
+        output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
+                         ac->oc[1].status, 0);
     }
 }
 
@@ -774,13 +784,15 @@ static int decode_audio_specific_config(AACContext *ac,
 {
     GetBitContext gb;
     int i;
+    int ret;
 
     av_dlog(avctx, "audio specific config size %d\n", bit_size >> 3);
     for (i = 0; i < bit_size >> 3; i++)
          av_dlog(avctx, "%02x ", data[i]);
     av_dlog(avctx, "\n");
 
-    init_get_bits(&gb, data, bit_size);
+    if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
+        return ret;
 
     if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size, sync_extension)) < 0)
         return -1;
@@ -870,6 +882,8 @@ static void reset_predictor_group(PredictorState *ps, int group_num)
         ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
         size);
 
+static void aacdec_init(AACContext *ac);
+
 static av_cold int aac_decode_init(AVCodecContext *avctx)
 {
     AACContext *ac = avctx->priv_data;
@@ -877,6 +891,8 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
     ac->avctx = avctx;
     ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
 
+    aacdec_init(ac);
+
     avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
 
     if (avctx->extradata_size > 0) {
@@ -914,6 +930,11 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
         }
     }
 
+    if (avctx->channels > MAX_CHANNELS) {
+        av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
+        return AVERROR_INVALIDDATA;
+    }
+
     AAC_INIT_VLC_STATIC( 0, 304);
     AAC_INIT_VLC_STATIC( 1, 270);
     AAC_INIT_VLC_STATIC( 2, 550);
@@ -928,7 +949,6 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
 
     ff_aac_sbr_init();
 
-    ff_dsputil_init(&ac->dsp, avctx);
     ff_fmt_convert_init(&ac->fmt_conv, avctx);
     avpriv_float_dsp_init(&ac->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
 
@@ -952,9 +972,6 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
 
     cbrt_tableinit();
 
-    avcodec_get_frame_defaults(&ac->frame);
-    avctx->coded_frame = &ac->frame;
-
     return 0;
 }
 
@@ -1394,7 +1411,7 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
                         cfo[k] = ac->random_state;
                     }
 
-                    band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len);
+                    band_energy = ac->fdsp.scalarproduct_float(cfo, cfo, off_len);
                     scale = sf[idx] / sqrtf(band_energy);
                     ac->fdsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
                 }
@@ -1729,9 +1746,9 @@ static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
             if (cpe->ms_mask[idx] &&
                     cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
                 for (group = 0; group < ics->group_len[g]; group++) {
-                    ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
-                                              ch1 + group * 128 + offsets[i],
-                                              offsets[i+1] - offsets[i]);
+                    ac->fdsp.butterflies_float(ch0 + group * 128 + offsets[i],
+                                               ch1 + group * 128 + offsets[i],
+                                               offsets[i+1] - offsets[i]);
                 }
             }
         }
@@ -2131,9 +2148,9 @@ static void windowing_and_mdct_ltp(AACContext *ac, float *out,
         ac->fdsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
     }
     if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
-        ac->dsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
+        ac->fdsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
     } else {
-        ac->dsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
+        ac->fdsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
         memset(in + 1024 + 576, 0, 448 * sizeof(float));
     }
     ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
@@ -2159,10 +2176,10 @@ static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
             predTime[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
         memset(&predTime[i], 0, (2048 - i) * sizeof(float));
 
-        windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
+        ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
 
         if (sce->tns.present)
-            apply_tns(predFreq, &sce->tns, &sce->ics, 0);
+            ac->apply_tns(predFreq, &sce->tns, &sce->ics, 0);
 
         for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
             if (ltp->used[sfb])
@@ -2186,17 +2203,17 @@ static void update_ltp(AACContext *ac, SingleChannelElement *sce)
     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
         memcpy(saved_ltp,       saved, 512 * sizeof(float));
         memset(saved_ltp + 576, 0,     448 * sizeof(float));
-        ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
+        ac->fdsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
         for (i = 0; i < 64; i++)
             saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
     } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
         memcpy(saved_ltp,       ac->buf_mdct + 512, 448 * sizeof(float));
         memset(saved_ltp + 576, 0,                  448 * sizeof(float));
-        ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
+        ac->fdsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
         for (i = 0; i < 64; i++)
             saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
     } else { // LONG_STOP or ONLY_LONG
-        ac->dsp.vector_fmul_reverse(saved_ltp,       ac->buf_mdct + 512,     &lwindow[512],     512);
+        ac->fdsp.vector_fmul_reverse(saved_ltp,       ac->buf_mdct + 512,     &lwindow[512],     512);
         for (i = 0; i < 512; i++)
             saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];
     }
@@ -2237,35 +2254,35 @@ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
      */
     if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
             (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
-        ac->dsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, 512);
+        ac->fdsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, 512);
     } else {
-        memcpy(                        out,               saved,            448 * sizeof(float));
+        memcpy(                         out,               saved,            448 * sizeof(float));
 
         if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
-            ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, 64);
-            ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      64);
-            ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      64);
-            ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      64);
-            ac->dsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      64);
-            memcpy(                    out + 448 + 4*128, temp, 64 * sizeof(float));
+            ac->fdsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, 64);
+            ac->fdsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      64);
+            ac->fdsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      64);
+            ac->fdsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      64);
+            ac->fdsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      64);
+            memcpy(                     out + 448 + 4*128, temp, 64 * sizeof(float));
         } else {
-            ac->dsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, 64);
-            memcpy(                    out + 576,         buf + 64,         448 * sizeof(float));
+            ac->fdsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, 64);
+            memcpy(                     out + 576,         buf + 64,         448 * sizeof(float));
         }
     }
 
     // buffer update
     if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
-        memcpy(                    saved,       temp + 64,         64 * sizeof(float));
-        ac->dsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 64);
-        ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
-        ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
-        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
+        memcpy(                     saved,       temp + 64,         64 * sizeof(float));
+        ac->fdsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 64);
+        ac->fdsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
+        ac->fdsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
+        memcpy(                     saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
     } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
-        memcpy(                    saved,       buf + 512,        448 * sizeof(float));
-        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
+        memcpy(                     saved,       buf + 512,        448 * sizeof(float));
+        memcpy(                     saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
     } else { // LONG_STOP or ONLY_LONG
-        memcpy(                    saved,       buf + 512,        512 * sizeof(float));
+        memcpy(                     saved,       buf + 512,        512 * sizeof(float));
     }
 }
 
@@ -2374,25 +2391,25 @@ static void spectral_to_sample(AACContext *ac)
                 if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
                     if (che->ch[0].ics.predictor_present) {
                         if (che->ch[0].ics.ltp.present)
-                            apply_ltp(ac, &che->ch[0]);
+                            ac->apply_ltp(ac, &che->ch[0]);
                         if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
-                            apply_ltp(ac, &che->ch[1]);
+                            ac->apply_ltp(ac, &che->ch[1]);
                     }
                 }
                 if (che->ch[0].tns.present)
-                    apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
+                    ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
                 if (che->ch[1].tns.present)
-                    apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
+                    ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
                 if (type <= TYPE_CPE)
                     apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
                 if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
-                    imdct_and_windowing(ac, &che->ch[0]);
+                    ac->imdct_and_windowing(ac, &che->ch[0]);
                     if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
-                        update_ltp(ac, &che->ch[0]);
+                        ac->update_ltp(ac, &che->ch[0]);
                     if (type == TYPE_CPE) {
-                        imdct_and_windowing(ac, &che->ch[1]);
+                        ac->imdct_and_windowing(ac, &che->ch[1]);
                         if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
-                            update_ltp(ac, &che->ch[1]);
+                            ac->update_ltp(ac, &che->ch[1]);
                     }
                     if (ac->oc[1].m4ac.sbr > 0) {
                         ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
@@ -2472,6 +2489,8 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
     int samples = 0, multiplier, audio_found = 0, pce_found = 0;
     int is_dmono, sce_count = 0;
 
+    ac->frame = data;
+
     if (show_bits(gb, 12) == 0xfff) {
         if (parse_adts_frame_header(ac, gb) < 0) {
             av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
@@ -2543,7 +2562,6 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
             if (pce_found) {
                 av_log(avctx, AV_LOG_ERROR,
                        "Not evaluating a further program_config_element as this construct is dubious at best.\n");
-                pop_output_configuration(ac);
             } else {
                 err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
                 if (!err)
@@ -2592,10 +2610,8 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
     is_dmono = ac->dmono_mode && sce_count == 2 &&
                ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
 
-    if (samples) {
-        ac->frame.nb_samples = samples;
-        *(AVFrame *)data = ac->frame;
-    }
+    if (samples)
+        ac->frame->nb_samples = samples;
     *got_frame_ptr = !!samples;
 
     if (is_dmono) {
@@ -2613,7 +2629,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
 
     if (multiplier) {
         int side_size;
-        uint32_t *side = av_packet_get_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, &side_size);
+        const uint8_t *side = av_packet_get_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, &side_size);
         if (side && side_size>=4)
             AV_WL32(side, 2*AV_RL32(side));
     }
@@ -2665,6 +2681,9 @@ static int aac_decode_frame(AVCodecContext *avctx, void *data,
     if (ac->force_dmono_mode >= 0)
         ac->dmono_mode = ac->force_dmono_mode;
 
+    if (INT_MAX / 8 <= buf_size)
+        return AVERROR_INVALIDDATA;
+
     init_get_bits(&gb, buf, buf_size * 8);
 
     if ((err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt)) < 0)
@@ -2754,7 +2773,7 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
         if(latmctx->initialized) {
             av_log(avctx, AV_LOG_INFO, "audio config changed\n");
         } else {
-            av_log(avctx, AV_LOG_INFO, "initializing latmctx\n");
+            av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n");
         }
         latmctx->initialized = 0;
 
@@ -2915,7 +2934,8 @@ static int latm_decode_frame(AVCodecContext *avctx, void *out,
     int                 muxlength, err;
     GetBitContext       gb;
 
-    init_get_bits(&gb, avpkt->data, avpkt->size * 8);
+    if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0)
+        return err;
 
     // check for LOAS sync word
     if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
@@ -2969,6 +2989,17 @@ static av_cold int latm_decode_init(AVCodecContext *avctx)
     return ret;
 }
 
+static void aacdec_init(AACContext *c)
+{
+    c->imdct_and_windowing                      = imdct_and_windowing;
+    c->apply_ltp                                = apply_ltp;
+    c->apply_tns                                = apply_tns;
+    c->windowing_and_mdct_ltp                   = windowing_and_mdct_ltp;
+    c->update_ltp                               = update_ltp;
+
+    if(ARCH_MIPS)
+        ff_aacdec_init_mips(c);
+}
 /**
  * AVOptions for Japanese DTV specific extensions (ADTS only)
  */

libavcodec/aacenc.c

@@ -34,7 +34,6 @@
 #include "libavutil/opt.h"
 #include "avcodec.h"
 #include "put_bits.h"
-#include "dsputil.h"
 #include "internal.h"
 #include "mpeg4audio.h"
 #include "kbdwin.h"
@@ -183,7 +182,7 @@ static void put_audio_specific_config(AVCodecContext *avctx)
 }
 
 #define WINDOW_FUNC(type) \
-static void apply_ ##type ##_window(DSPContext *dsp, AVFloatDSPContext *fdsp, \
+static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
                                     SingleChannelElement *sce, \
                                     const float *audio)
 
@@ -193,8 +192,8 @@ WINDOW_FUNC(only_long)
     const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
     float *out = sce->ret_buf;
 
-    fdsp->vector_fmul       (out,        audio,        lwindow, 1024);
-    dsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
+    fdsp->vector_fmul        (out,        audio,        lwindow, 1024);
+    fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
 }
 
 WINDOW_FUNC(long_start)
@@ -205,7 +204,7 @@ WINDOW_FUNC(long_start)
 
     fdsp->vector_fmul(out, audio, lwindow, 1024);
     memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
-    dsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
+    fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
     memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
 }
 
@@ -218,7 +217,7 @@ WINDOW_FUNC(long_stop)
     memset(out, 0, sizeof(out[0]) * 448);
     fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
     memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
-    dsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
+    fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
 }
 
 WINDOW_FUNC(eight_short)
@@ -230,15 +229,15 @@ WINDOW_FUNC(eight_short)
     int w;
 
     for (w = 0; w < 8; w++) {
-        fdsp->vector_fmul       (out, in, w ? pwindow : swindow, 128);
+        fdsp->vector_fmul        (out, in, w ? pwindow : swindow, 128);
         out += 128;
         in  += 128;
-        dsp->vector_fmul_reverse(out, in, swindow, 128);
+        fdsp->vector_fmul_reverse(out, in, swindow, 128);
         out += 128;
     }
 }
 
-static void (*const apply_window[4])(DSPContext *dsp, AVFloatDSPContext *fdsp,
+static void (*const apply_window[4])(AVFloatDSPContext *fdsp,
                                      SingleChannelElement *sce,
                                      const float *audio) = {
     [ONLY_LONG_SEQUENCE]   = apply_only_long_window,
@@ -253,7 +252,7 @@ static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
     int i;
     float *output = sce->ret_buf;
 
-    apply_window[sce->ics.window_sequence[0]](&s->dsp, &s->fdsp, sce, audio);
+    apply_window[sce->ics.window_sequence[0]](&s->fdsp, sce, audio);
 
     if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
         s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
@@ -517,7 +516,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
 
     /* add current frame to queue */
     if (frame) {
-        if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
+        if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
             return ret;
     }
 
@@ -571,10 +570,8 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         }
         start_ch += chans;
     }
-    if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels))) {
-        av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+    if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels)) < 0)
         return ret;
-    }
     do {
         int frame_bits;
 
@@ -692,7 +689,6 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
 {
     int ret = 0;
 
-    ff_dsputil_init(&s->dsp, avctx);
     avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
 
     // window init

libavcodec/aacenc.h

@@ -25,7 +25,6 @@
 #include "libavutil/float_dsp.h"
 #include "avcodec.h"
 #include "put_bits.h"
-#include "dsputil.h"
 
 #include "aac.h"
 #include "audio_frame_queue.h"
@@ -61,7 +60,6 @@ typedef struct AACEncContext {
     PutBitContext pb;
     FFTContext mdct1024;                         ///< long (1024 samples) frame transform context
     FFTContext mdct128;                          ///< short (128 samples) frame transform context
-    DSPContext  dsp;
     AVFloatDSPContext fdsp;
     float *planar_samples[6];                    ///< saved preprocessed input
 

libavcodec/aacps.c

@@ -21,13 +21,13 @@
 
 #include <stdint.h>
 #include "libavutil/common.h"
+#include "libavutil/internal.h"
 #include "libavutil/mathematics.h"
 #include "avcodec.h"
 #include "get_bits.h"
 #include "aacps.h"
 #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,
@@ -429,6 +429,7 @@ static void hybrid_synthesis(PSDSPContext *dsp, float out[2][38][64],
 #define DECAY_SLOPE      0.05f
 /// Number of frequency bands that can be addressed by the parameter index, b(k)
 static const int   NR_PAR_BANDS[]      = { 20, 34 };
+static const int   NR_IPDOPD_BANDS[]   = { 11, 17 };
 /// Number of frequency bands that can be addressed by the sub subband index, k
 static const int   NR_BANDS[]          = { 71, 91 };
 /// Start frequency band for the all-pass filter decay slope
@@ -823,7 +824,8 @@ static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2
             h12 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][1];
             h21 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][2];
             h22 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][3];
-            if (!PS_BASELINE && ps->enable_ipdopd && b < ps->nr_ipdopd_par) {
+
+            if (!PS_BASELINE && ps->enable_ipdopd && b < NR_IPDOPD_BANDS[is34]) {
                 //The spec say says to only run this smoother when enable_ipdopd
                 //is set but the reference decoder appears to run it constantly
                 float h11i, h12i, h21i, h22i;

libavcodec/aacpsdsp.c

@@ -211,4 +211,6 @@ av_cold void ff_psdsp_init(PSDSPContext *s)
 
     if (ARCH_ARM)
         ff_psdsp_init_arm(s);
+    if (ARCH_MIPS)
+        ff_psdsp_init_mips(s);
 }

libavcodec/aacpsdsp.h

@@ -49,5 +49,6 @@ typedef struct PSDSPContext {
 
 void ff_psdsp_init(PSDSPContext *s);
 void ff_psdsp_init_arm(PSDSPContext *s);
+void ff_psdsp_init_mips(PSDSPContext *s);
 
 #endif /* LIBAVCODEC_AACPSDSP_H */

libavcodec/aacpsy.c

@@ -541,8 +541,10 @@ static float calc_reduced_thr_3gpp(AacPsyBand *band, float min_snr,
     float thr = band->thr;
 
     if (band->energy > thr) {
-        thr = powf(thr, 0.25f) + reduction;
-        thr = powf(thr, 4.0f);
+        thr = sqrtf(thr);
+        thr = sqrtf(thr) + reduction;
+        thr *= thr;
+        thr *= thr;
 
         /* This deviates from the 3GPP spec to match the reference encoder.
          * It performs min(thr_reduced, max(thr, energy/min_snr)) only for bands
@@ -582,13 +584,15 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
             AacPsyBand *band = &pch->band[w+g];
 
             float form_factor = 0.0f;
+            float Temp;
             band->energy = 0.0f;
             for (i = 0; i < band_sizes[g]; i++) {
                 band->energy += coefs[start+i] * coefs[start+i];
                 form_factor  += sqrtf(fabs(coefs[start+i]));
             }
+            Temp = band->energy > 0 ? sqrtf((float)band_sizes[g] / band->energy) : 0;
             band->thr      = band->energy * 0.001258925f;
-            band->nz_lines = band->energy>0 ? form_factor / powf(band->energy / band_sizes[g], 0.25f) : 0;
+            band->nz_lines = form_factor * sqrtf(Temp);
 
             start += band_sizes[g];
         }

libavcodec/aacsbr.c

@@ -33,6 +33,7 @@
 #include "fft.h"
 #include "aacps.h"
 #include "sbrdsp.h"
+#include "libavutil/internal.h"
 #include "libavutil/libm.h"
 #include "libavutil/avassert.h"
 
@@ -43,6 +44,10 @@
 #define ENVELOPE_ADJUSTMENT_OFFSET 2
 #define NOISE_FLOOR_OFFSET 6.0f
 
+#if ARCH_MIPS
+#include "mips/aacsbr_mips.h"
+#endif /* ARCH_MIPS */
+
 /**
  * SBR VLC tables
  */
@@ -86,6 +91,8 @@ static const int8_t vlc_sbr_lav[10] =
 #define SBR_VLC_ROW(name) \
     { name ## _codes, name ## _bits, sizeof(name ## _codes), sizeof(name ## _codes[0]) }
 
+static void aacsbr_func_ptr_init(AACSBRContext *c);
+
 av_cold void ff_aac_sbr_init(void)
 {
     int n;
@@ -154,6 +161,7 @@ av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr)
     ff_mdct_init(&sbr->mdct_ana, 7, 1, -2.0 * 32768.0);
     ff_ps_ctx_init(&sbr->ps);
     ff_sbrdsp_init(&sbr->dsp);
+    aacsbr_func_ptr_init(&sbr->c);
 }
 
 av_cold void ff_aac_sbr_ctx_close(SpectralBandReplication *sbr)
@@ -391,6 +399,8 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
         max_qmf_subbands = 35;
     } else if (sbr->sample_rate >= 48000)
         max_qmf_subbands = 32;
+    else
+        av_assert0(0);
 
     if (sbr->k[2] - sbr->k[0] > max_qmf_subbands) {
         av_log(ac->avctx, AV_LOG_ERROR,
@@ -1121,7 +1131,12 @@ static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
             for (k = 0; k < sbr->n[sbr->data[0].bs_freq_res[e]]; k++) {
                 float temp1 = exp2f(sbr->data[0].env_facs[e][k] * alpha + 7.0f);
                 float temp2 = exp2f((pan_offset - sbr->data[1].env_facs[e][k]) * alpha);
-                float fac   = temp1 / (1.0f + temp2);
+                float fac;
+                if (temp1 > 1E20) {
+                    av_log(NULL, AV_LOG_ERROR, "envelope scalefactor overflow in dequant\n");
+                    temp1 = 1;
+                }
+                fac   = temp1 / (1.0f + temp2);
                 sbr->data[0].env_facs[e][k] = fac;
                 sbr->data[1].env_facs[e][k] = fac * temp2;
             }
@@ -1130,7 +1145,12 @@ static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
             for (k = 0; k < sbr->n_q; k++) {
                 float temp1 = exp2f(NOISE_FLOOR_OFFSET - sbr->data[0].noise_facs[e][k] + 1);
                 float temp2 = exp2f(12 - sbr->data[1].noise_facs[e][k]);
-                float fac   = temp1 / (1.0f + temp2);
+                float fac;
+                if (temp1 > 1E20) {
+                    av_log(NULL, AV_LOG_ERROR, "envelope scalefactor overflow in dequant\n");
+                    temp1 = 1;
+                }
+                fac = temp1 / (1.0f + temp2);
                 sbr->data[0].noise_facs[e][k] = fac;
                 sbr->data[1].noise_facs[e][k] = fac * temp2;
             }
@@ -1139,9 +1159,15 @@ static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
         for (ch = 0; ch < (id_aac == TYPE_CPE) + 1; ch++) {
             float alpha = sbr->data[ch].bs_amp_res ? 1.0f : 0.5f;
             for (e = 1; e <= sbr->data[ch].bs_num_env; e++)
-                for (k = 0; k < sbr->n[sbr->data[ch].bs_freq_res[e]]; k++)
+                for (k = 0; k < sbr->n[sbr->data[ch].bs_freq_res[e]]; k++){
                     sbr->data[ch].env_facs[e][k] =
                         exp2f(alpha * sbr->data[ch].env_facs[e][k] + 6.0f);
+                    if (sbr->data[ch].env_facs[e][k] > 1E20) {
+                        av_log(NULL, AV_LOG_ERROR, "envelope scalefactor overflow in dequant\n");
+                        sbr->data[ch].env_facs[e][k] = 1;
+                    }
+                }
+
             for (e = 1; e <= sbr->data[ch].bs_num_noise; e++)
                 for (k = 0; k < sbr->n_q; k++)
                     sbr->data[ch].noise_facs[e][k] =
@@ -1156,7 +1182,8 @@ 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,
+#ifndef sbr_qmf_analysis
+static void sbr_qmf_analysis(AVFloatDSPContext *dsp, FFTContext *mdct,
                              SBRDSPContext *sbrdsp, const float *in, float *x,
                              float z[320], float W[2][32][32][2], int buf_idx)
 {
@@ -1173,13 +1200,15 @@ static void sbr_qmf_analysis(DSPContext *dsp, FFTContext *mdct,
         x += 32;
     }
 }
+#endif
 
 /**
  * Synthesis QMF Bank (14496-3 sp04 p206) and Downsampled Synthesis QMF Bank
  * (14496-3 sp04 p206)
  */
-static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
-                              SBRDSPContext *sbrdsp, AVFloatDSPContext *fdsp,
+#ifndef sbr_qmf_synthesis
+static void sbr_qmf_synthesis(FFTContext *mdct,
+                              SBRDSPContext *sbrdsp, AVFloatDSPContext *dsp,
                               float *out, float X[2][38][64],
                               float mdct_buf[2][64],
                               float *v0, int *v_off, const unsigned int div)
@@ -1210,7 +1239,7 @@ static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
             mdct->imdct_half(mdct, mdct_buf[1], X[1][i]);
             sbrdsp->qmf_deint_bfly(v, mdct_buf[1], mdct_buf[0]);
         }
-        fdsp->vector_fmul   (out, v                , sbr_qmf_window                       , 64 >> div);
+        dsp->vector_fmul    (out, v                , sbr_qmf_window                       , 64 >> div);
         dsp->vector_fmul_add(out, v + ( 192 >> div), sbr_qmf_window + ( 64 >> div), out   , 64 >> div);
         dsp->vector_fmul_add(out, v + ( 256 >> div), sbr_qmf_window + (128 >> div), out   , 64 >> div);
         dsp->vector_fmul_add(out, v + ( 448 >> div), sbr_qmf_window + (192 >> div), out   , 64 >> div);
@@ -1223,6 +1252,7 @@ static void sbr_qmf_synthesis(DSPContext *dsp, FFTContext *mdct,
         out += 64 >> div;
     }
 }
+#endif
 
 /** High Frequency Generation (14496-3 sp04 p214+) and Inverse Filtering
  * (14496-3 sp04 p214)
@@ -1668,13 +1698,13 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int 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->fdsp, &sbr->mdct_ana, &sbr->dsp, ch ? R : L, sbr->data[ch].analysis_filterbank_samples,
                          (float*)sbr->qmf_filter_scratch,
                          sbr->data[ch].W, sbr->data[ch].Ypos);
-        sbr_lf_gen(ac, sbr, sbr->X_low, sbr->data[ch].W, sbr->data[ch].Ypos);
+        sbr->c.sbr_lf_gen(ac, sbr, sbr->X_low, sbr->data[ch].W, sbr->data[ch].Ypos);
         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->c.sbr_hf_inverse_filter(&sbr->dsp, 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,
@@ -1685,14 +1715,14 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
             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->c.sbr_hf_assemble(sbr->data[ch].Y[sbr->data[ch].Ypos],
                                 sbr->X_high, sbr, &sbr->data[ch],
                                 sbr->data[ch].e_a);
             }
         }
 
         /* synthesis */
-        sbr_x_gen(sbr, sbr->X[ch],
+        sbr->c.sbr_x_gen(sbr, sbr->X[ch],
                   sbr->data[ch].Y[1-sbr->data[ch].Ypos],
                   sbr->data[ch].Y[  sbr->data[ch].Ypos],
                   sbr->X_low, ch);
@@ -1707,15 +1737,26 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
         nch = 2;
     }
 
-    sbr_qmf_synthesis(&ac->dsp, &sbr->mdct, &sbr->dsp, &ac->fdsp,
+    sbr_qmf_synthesis(&sbr->mdct, &sbr->dsp, &ac->fdsp,
                       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, &ac->fdsp,
+        sbr_qmf_synthesis(&sbr->mdct, &sbr->dsp, &ac->fdsp,
                           R, sbr->X[1], sbr->qmf_filter_scratch,
                           sbr->data[1].synthesis_filterbank_samples,
                           &sbr->data[1].synthesis_filterbank_samples_offset,
                           downsampled);
 }
+
+static void aacsbr_func_ptr_init(AACSBRContext *c)
+{
+    c->sbr_lf_gen            = sbr_lf_gen;
+    c->sbr_hf_assemble       = sbr_hf_assemble;
+    c->sbr_x_gen             = sbr_x_gen;
+    c->sbr_hf_inverse_filter = sbr_hf_inverse_filter;
+
+    if(ARCH_MIPS)
+        ff_aacsbr_func_ptr_init_mips(c);
+}

libavcodec/aacsbr.h

@@ -34,11 +34,11 @@
 #include "sbr.h"
 
 /** Initialize SBR. */
-av_cold void ff_aac_sbr_init(void);
+void ff_aac_sbr_init(void);
 /** Initialize one SBR context. */
-av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr);
+void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr);
 /** Close one SBR context. */
-av_cold void ff_aac_sbr_ctx_close(SpectralBandReplication *sbr);
+void ff_aac_sbr_ctx_close(SpectralBandReplication *sbr);
 /** Decode one SBR element. */
 int ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
                             GetBitContext *gb, int crc, int cnt, int id_aac);
@@ -46,4 +46,6 @@ int ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
 void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
                   float* L, float *R);
 
+void ff_aacsbr_func_ptr_init_mips(AACSBRContext *c);
+
 #endif /* AVCODEC_AACSBR_H */

libavcodec/aasc.c

@@ -29,7 +29,6 @@
 #include <string.h>
 
 #include "avcodec.h"
-#include "dsputil.h"
 #include "msrledec.h"
 
 typedef struct AascContext {
@@ -60,7 +59,7 @@ static av_cold int aasc_decode_init(AVCodecContext *avctx)
         }
         break;
     case 16:
-        avctx->pix_fmt = AV_PIX_FMT_RGB555;
+        avctx->pix_fmt = AV_PIX_FMT_RGB555LE;
         break;
     case 24:
         avctx->pix_fmt = AV_PIX_FMT_BGR24;
@@ -81,7 +80,7 @@ static int aasc_decode_frame(AVCodecContext *avctx,
     const uint8_t *buf = avpkt->data;
     int buf_size       = avpkt->size;
     AascContext *s     = avctx->priv_data;
-    int compr, i, stride, psize;
+    int compr, i, stride, psize, ret;
 
     if (buf_size < 4) {
         av_log(avctx, AV_LOG_ERROR, "frame too short\n");
@@ -90,9 +89,9 @@ static int aasc_decode_frame(AVCodecContext *avctx,
 
     s->frame.reference = 3;
     s->frame.buffer_hints = FF_BUFFER_HINTS_VALID | FF_BUFFER_HINTS_PRESERVE | FF_BUFFER_HINTS_REUSABLE;
-    if (avctx->reget_buffer(avctx, &s->frame)) {
+    if ((ret = avctx->reget_buffer(avctx, &s->frame)) < 0) {
         av_log(avctx, AV_LOG_ERROR, "reget_buffer() failed\n");
-        return -1;
+        return ret;
     }
 
     compr     = AV_RL32(buf);
@@ -124,7 +123,7 @@ static int aasc_decode_frame(AVCodecContext *avctx,
         break;
     default:
         av_log(avctx, AV_LOG_ERROR, "Unknown compression type %d\n", compr);
-        return -1;
+        return AVERROR_INVALIDDATA;
     }
         break;
     default:

libavcodec/ac3dec.c

@@ -170,6 +170,7 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
     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);
+    avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
     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);
@@ -184,9 +185,6 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
     }
     s->downmixed = 1;
 
-    avcodec_get_frame_defaults(&s->frame);
-    avctx->coded_frame = &s->frame;
-
     for (i = 0; i < AC3_MAX_CHANNELS; i++) {
         s->xcfptr[i] = s->transform_coeffs[i];
         s->dlyptr[i] = s->delay[i];
@@ -441,8 +439,9 @@ static void ac3_decode_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, ma
         int mantissa;
         switch (bap) {
         case 0:
+            /* random noise with approximate range of -0.707 to 0.707 */
             if (dither)
-                mantissa = (av_lfg_get(&s->dith_state) & 0x7FFFFF) - 0x400000;
+                mantissa = (((av_lfg_get(&s->dith_state)>>8)*181)>>8) - 5931008;
             else
                 mantissa = 0;
             break;
@@ -606,15 +605,15 @@ static inline void do_imdct(AC3DecodeContext *s, int channels)
             for (i = 0; i < 128; i++)
                 x[i] = s->transform_coeffs[ch][2 * i];
             s->imdct_256.imdct_half(&s->imdct_256, s->tmp_output, x);
-            s->dsp.vector_fmul_window(s->outptr[ch - 1], s->delay[ch - 1],
-                                      s->tmp_output, s->window, 128);
+            s->fdsp.vector_fmul_window(s->outptr[ch - 1], s->delay[ch - 1],
+                                       s->tmp_output, s->window, 128);
             for (i = 0; i < 128; i++)
                 x[i] = s->transform_coeffs[ch][2 * i + 1];
             s->imdct_256.imdct_half(&s->imdct_256, s->delay[ch - 1], x);
         } else {
             s->imdct_512.imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]);
-            s->dsp.vector_fmul_window(s->outptr[ch - 1], s->delay[ch - 1],
-                                      s->tmp_output, s->window, 128);
+            s->fdsp.vector_fmul_window(s->outptr[ch - 1], s->delay[ch - 1],
+                                       s->tmp_output, s->window, 128);
             memcpy(s->delay[ch - 1], s->tmp_output + 128, 128 * sizeof(float));
         }
     }
@@ -1265,6 +1264,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
 static int ac3_decode_frame(AVCodecContext * avctx, void *data,
                             int *got_frame_ptr, AVPacket *avpkt)
 {
+    AVFrame *frame     = data;
     const uint8_t *buf = avpkt->data;
     int buf_size = avpkt->size;
     AC3DecodeContext *s = avctx->priv_data;
@@ -1336,8 +1336,10 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
     if (!err) {
         avctx->sample_rate = s->sample_rate;
         avctx->bit_rate    = s->bit_rate;
+    }
 
-        /* channel config */
+    /* channel config */
+    if (!err || (s->channels && s->out_channels != s->channels)) {
         s->out_channels = s->channels;
         s->output_mode  = s->channel_mode;
         if (s->lfe_on)
@@ -1360,24 +1362,20 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
                 s->fbw_channels == s->out_channels)) {
             set_downmix_coeffs(s);
         }
-    } else if (!s->out_channels) {
-        s->out_channels = avctx->channels;
-        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");
+    } else if (!s->channels) {
+        av_log(avctx, AV_LOG_ERROR, "unable to determine channel mode\n");
         return AVERROR_INVALIDDATA;
     }
+    avctx->channels = s->out_channels;
+
     /* 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 = ff_get_buffer(avctx, &s->frame)) < 0) {
+    frame->nb_samples = s->num_blocks * 256;
+    if ((ret = ff_get_buffer(avctx, frame)) < 0) {
         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
         return ret;
     }
@@ -1390,7 +1388,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
     }
     for (ch = 0; ch < s->channels; ch++) {
         if (ch < s->out_channels)
-            s->outptr[channel_map[ch]] = (float *)s->frame.data[ch];
+            s->outptr[channel_map[ch]] = (float *)frame->data[ch];
     }
     for (blk = 0; blk < s->num_blocks; blk++) {
         if (!err && decode_audio_block(s, blk)) {
@@ -1399,7 +1397,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
         }
         if (err)
             for (ch = 0; ch < s->out_channels; ch++)
-                memcpy(((float*)s->frame.data[ch]) + AC3_BLOCK_SIZE*blk, output[ch], 1024);
+                memcpy(((float*)frame->data[ch]) + AC3_BLOCK_SIZE*blk, output[ch], 1024);
         for (ch = 0; ch < s->out_channels; ch++)
             output[ch] = s->outptr[channel_map[ch]];
         for (ch = 0; ch < s->out_channels; ch++) {
@@ -1408,14 +1406,13 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
         }
     }
 
-    s->frame.decode_error_flags = err ? FF_DECODE_ERROR_INVALID_BITSTREAM : 0;
+    av_frame_set_decode_error_flags(frame, err ? FF_DECODE_ERROR_INVALID_BITSTREAM : 0);
 
     /* keep last block for error concealment in next frame */
     for (ch = 0; ch < s->out_channels; ch++)
         memcpy(s->output[ch], output[ch], 1024);
 
-    *got_frame_ptr   = 1;
-    *(AVFrame *)data = s->frame;
+    *got_frame_ptr = 1;
 
     return FFMIN(buf_size, s->frame_size);
 }

libavcodec/ac3dec.h

@@ -50,6 +50,7 @@
 #ifndef AVCODEC_AC3DEC_H
 #define AVCODEC_AC3DEC_H
 
+#include "libavutil/float_dsp.h"
 #include "libavutil/lfg.h"
 #include "ac3.h"
 #include "ac3dsp.h"
@@ -68,7 +69,6 @@
 typedef struct AC3DecodeContext {
     AVClass        *class;                  ///< class for AVOptions
     AVCodecContext *avctx;                  ///< parent context
-    AVFrame frame;                          ///< AVFrame for decoded output
     GetBitContext gbc;                      ///< bitstream reader
 
 ///@name Bit stream information
@@ -199,6 +199,7 @@ typedef struct AC3DecodeContext {
 
 ///@name Optimization
     DSPContext dsp;                         ///< for optimization
+    AVFloatDSPContext fdsp;
     AC3DSPContext ac3dsp;
     FmtConvertContext fmt_conv;             ///< optimized conversion functions
 ///@}

libavcodec/ac3enc.c

@@ -34,10 +34,10 @@
 #include "libavutil/avstring.h"
 #include "libavutil/channel_layout.h"
 #include "libavutil/crc.h"
+#include "libavutil/internal.h"
 #include "libavutil/opt.h"
 #include "avcodec.h"
 #include "put_bits.h"
-#include "dsputil.h"
 #include "ac3dsp.h"
 #include "ac3.h"
 #include "fft.h"

libavcodec/ac3enc_template.c

@@ -28,6 +28,7 @@
 
 #include <stdint.h>
 
+#include "libavutil/internal.h"
 
 /* prototypes for static functions in ac3enc_fixed.c and ac3enc_float.c */
 
@@ -132,7 +133,7 @@ static void apply_channel_coupling(AC3EncodeContext *s)
 #else
     int32_t (*fixed_cpl_coords)[AC3_MAX_CHANNELS][16] = cpl_coords;
 #endif
-    int blk, ch, bnd, i, j;
+    int av_uninit(blk), ch, bnd, i, j;
     CoefSumType energy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][16] = {{{0}}};
     int cpl_start, num_cpl_coefs;
 
@@ -336,7 +337,7 @@ static void compute_rematrixing_strategy(AC3EncodeContext *s)
 {
     int nb_coefs;
     int blk, bnd;
-    AC3Block *block, *block0;
+    AC3Block *block, *block0 = NULL;
 
     if (s->channel_mode != AC3_CHMODE_STEREO)
         return;
@@ -434,7 +435,7 @@ int AC3_NAME(encode_frame)(AVCodecContext *avctx, AVPacket *avpkt,
 
     ff_ac3_quantize_mantissas(s);
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, s->frame_size)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, s->frame_size)) < 0)
         return ret;
     ff_ac3_output_frame(s, avpkt->data);
 

libavcodec/acelp_pitch_delay.c

@@ -21,10 +21,10 @@
  */
 
 #include "libavutil/common.h"
+#include "libavutil/float_dsp.h"
 #include "libavutil/libm.h"
 #include "libavutil/mathematics.h"
 #include "avcodec.h"
-#include "dsputil.h"
 #include "acelp_pitch_delay.h"
 #include "celp_math.h"
 
@@ -132,7 +132,7 @@ float ff_amr_set_fixed_gain(float fixed_gain_factor, float fixed_mean_energy,
     // Note 10^(0.05 * -10log(average x2)) = 1/sqrt((average x2)).
     float val = fixed_gain_factor *
         exp2f(M_LOG2_10 * 0.05 *
-              (ff_scalarproduct_float_c(pred_table, prediction_error, 4) +
+              (avpriv_scalarproduct_float_c(pred_table, prediction_error, 4) +
                energy_mean)) /
         sqrtf(fixed_mean_energy);
 

libavcodec/acelp_vectors.c

@@ -23,8 +23,8 @@
 #include <inttypes.h>
 
 #include "libavutil/common.h"
+#include "libavutil/float_dsp.h"
 #include "avcodec.h"
-#include "dsputil.h"
 #include "acelp_vectors.h"
 
 const uint8_t ff_fc_2pulses_9bits_track1[16] =
@@ -203,7 +203,7 @@ void ff_adaptive_gain_control(float *out, const float *in, float speech_energ,
                               int size, float alpha, float *gain_mem)
 {
     int i;
-    float postfilter_energ = ff_scalarproduct_float_c(in, in, size);
+    float postfilter_energ = avpriv_scalarproduct_float_c(in, in, size);
     float gain_scale_factor = 1.0;
     float mem = *gain_mem;
 
@@ -224,7 +224,7 @@ void ff_scale_vector_to_given_sum_of_squares(float *out, const float *in,
                                              float sum_of_squares, const int n)
 {
     int i;
-    float scalefactor = ff_scalarproduct_float_c(in, in, n);
+    float scalefactor = avpriv_scalarproduct_float_c(in, in, n);
     if (scalefactor)
         scalefactor = sqrt(sum_of_squares / scalefactor);
     for (i = 0; i < n; i++)

libavcodec/adpcm.c

@@ -85,7 +85,6 @@ static const int swf_index_tables[4][16] = {
 /* end of tables */
 
 typedef struct ADPCMDecodeContext {
-    AVFrame frame;
     ADPCMChannelStatus status[6];
     int vqa_version;                /**< VQA version. Used for ADPCM_IMA_WS */
 } ADPCMDecodeContext;
@@ -159,9 +158,6 @@ static av_cold int adpcm_decode_init(AVCodecContext * avctx)
             avctx->sample_fmt = AV_SAMPLE_FMT_S16;
     }
 
-    avcodec_get_frame_defaults(&c->frame);
-    avctx->coded_frame = &c->frame;
-
     return 0;
 }
 
@@ -453,9 +449,11 @@ static void adpcm_swf_decode(AVCodecContext *avctx, const uint8_t *buf, int buf_
  * @param[out] coded_samples set to the number of samples as coded in the
  *                           packet, or 0 if the codec does not encode the
  *                           number of samples in each frame.
+ * @param[out] approx_nb_samples set to non-zero if the number of samples
+ *                               returned is an approximation.
  */
 static int get_nb_samples(AVCodecContext *avctx, GetByteContext *gb,
-                          int buf_size, int *coded_samples)
+                          int buf_size, int *coded_samples, int *approx_nb_samples)
 {
     ADPCMDecodeContext *s = avctx->priv_data;
     int nb_samples        = 0;
@@ -464,6 +462,7 @@ static int get_nb_samples(AVCodecContext *avctx, GetByteContext *gb,
     int header_size;
 
     *coded_samples = 0;
+    *approx_nb_samples = 0;
 
     if(ch <= 0)
         return 0;
@@ -534,10 +533,12 @@ static int get_nb_samples(AVCodecContext *avctx, GetByteContext *gb,
         case AV_CODEC_ID_ADPCM_EA_R2:
             header_size    = 4 + 5 * ch;
             *coded_samples = bytestream2_get_le32(gb);
+            *approx_nb_samples = 1;
             break;
         case AV_CODEC_ID_ADPCM_EA_R3:
             header_size    = 4 + 5 * ch;
             *coded_samples = bytestream2_get_be32(gb);
+            *approx_nb_samples = 1;
             break;
         }
         *coded_samples -= *coded_samples % 28;
@@ -619,6 +620,7 @@ static int get_nb_samples(AVCodecContext *avctx, GetByteContext *gb,
 static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
                               int *got_frame_ptr, AVPacket *avpkt)
 {
+    AVFrame *frame     = data;
     const uint8_t *buf = avpkt->data;
     int buf_size = avpkt->size;
     ADPCMDecodeContext *c = avctx->priv_data;
@@ -628,31 +630,31 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
     int16_t **samples_p;
     int st; /* stereo */
     int count1, count2;
-    int nb_samples, coded_samples, ret;
+    int nb_samples, coded_samples, approx_nb_samples, ret;
     GetByteContext gb;
 
     bytestream2_init(&gb, buf, buf_size);
-    nb_samples = get_nb_samples(avctx, &gb, buf_size, &coded_samples);
+    nb_samples = get_nb_samples(avctx, &gb, buf_size, &coded_samples, &approx_nb_samples);
     if (nb_samples <= 0) {
         av_log(avctx, AV_LOG_ERROR, "invalid number of samples in packet\n");
         return AVERROR_INVALIDDATA;
     }
 
     /* get output buffer */
-    c->frame.nb_samples = nb_samples;
-    if ((ret = ff_get_buffer(avctx, &c->frame)) < 0) {
+    frame->nb_samples = nb_samples;
+    if ((ret = ff_get_buffer(avctx, frame)) < 0) {
         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
         return ret;
     }
-    samples = (short *)c->frame.data[0];
-    samples_p = (int16_t **)c->frame.extended_data;
+    samples = (short *)frame->data[0];
+    samples_p = (int16_t **)frame->extended_data;
 
     /* use coded_samples when applicable */
     /* it is always <= nb_samples, so the output buffer will be large enough */
     if (coded_samples) {
-        if (coded_samples != nb_samples)
+        if (!approx_nb_samples && coded_samples != nb_samples)
             av_log(avctx, AV_LOG_WARNING, "mismatch in coded sample count\n");
-        c->frame.nb_samples = nb_samples = coded_samples;
+        frame->nb_samples = nb_samples = coded_samples;
     }
 
     st = avctx->channels == 2 ? 1 : 0;
@@ -738,7 +740,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
         }
 
         for (i = 0; i < avctx->channels; i++) {
-            samples = (int16_t *)c->frame.data[i];
+            samples = (int16_t *)frame->data[i];
             cs = &c->status[i];
             for (n = nb_samples >> 1; n > 0; n--) {
                 int v = bytestream2_get_byteu(&gb);
@@ -866,6 +868,9 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
             *samples++ = c->status[0].predictor + c->status[1].predictor;
             *samples++ = c->status[0].predictor - c->status[1].predictor;
         }
+
+        if ((bytestream2_tell(&gb) & 1))
+            bytestream2_skip(&gb, 1);
         break;
     }
     case AV_CODEC_ID_ADPCM_IMA_ISS:
@@ -1135,7 +1140,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
             }
         }
 
-        c->frame.nb_samples = count * 28;
+        frame->nb_samples = count * 28;
         bytestream2_seek(&gb, 0, SEEK_END);
         break;
     }
@@ -1378,8 +1383,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
         return AVERROR_INVALIDDATA;
     }
 
-    *got_frame_ptr   = 1;
-    *(AVFrame *)data = c->frame;
+    *got_frame_ptr = 1;
 
     return bytestream2_tell(&gb);
 }

libavcodec/adpcmenc.c

@@ -19,7 +19,6 @@
  */
 
 #include "avcodec.h"
-#include "get_bits.h"
 #include "put_bits.h"
 #include "bytestream.h"
 #include "adpcm.h"
@@ -495,7 +494,7 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         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)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, pkt_size)) < 0)
         return ret;
     dst = avpkt->data;
 
@@ -558,10 +557,11 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
             put_bits(&pb, 7,  status->step_index);
             if (avctx->trellis > 0) {
                 uint8_t buf[64];
-                adpcm_compress_trellis(avctx, &samples_p[ch][1], buf, status,
+                adpcm_compress_trellis(avctx, &samples_p[ch][0], buf, status,
                                        64, 1);
                 for (i = 0; i < 64; i++)
                     put_bits(&pb, 4, buf[i ^ 1]);
+                status->prev_sample = status->predictor;
             } else {
                 for (i = 0; i < 64; i += 2) {
                     int t1, t2;

libavcodec/adx.h

@@ -40,7 +40,6 @@ typedef struct ADXChannelState {
 } ADXChannelState;
 
 typedef struct ADXContext {
-    AVFrame frame;
     int channels;
     ADXChannelState prev[2];
     int header_parsed;

libavcodec/adxdec.c

@@ -52,9 +52,6 @@ static av_cold int adx_decode_init(AVCodecContext *avctx)
 
     avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
 
-    avcodec_get_frame_defaults(&c->frame);
-    avctx->coded_frame = &c->frame;
-
     return 0;
 }
 
@@ -98,6 +95,7 @@ static int adx_decode(ADXContext *c, int16_t *out, int offset,
 static int adx_decode_frame(AVCodecContext *avctx, void *data,
                             int *got_frame_ptr, AVPacket *avpkt)
 {
+    AVFrame *frame      = data;
     int buf_size        = avpkt->size;
     ADXContext *c       = avctx->priv_data;
     int16_t **samples;
@@ -143,12 +141,12 @@ static int adx_decode_frame(AVCodecContext *avctx, void *data,
     }
 
     /* get output buffer */
-    c->frame.nb_samples = num_blocks * BLOCK_SAMPLES;
-    if ((ret = ff_get_buffer(avctx, &c->frame)) < 0) {
+    frame->nb_samples = num_blocks * BLOCK_SAMPLES;
+    if ((ret = ff_get_buffer(avctx, frame)) < 0) {
         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
         return ret;
     }
-    samples = (int16_t **)c->frame.extended_data;
+    samples = (int16_t **)frame->extended_data;
     samples_offset = 0;
 
     while (num_blocks--) {
@@ -164,8 +162,7 @@ static int adx_decode_frame(AVCodecContext *avctx, void *data,
         samples_offset += BLOCK_SAMPLES;
     }
 
-    *got_frame_ptr   = 1;
-    *(AVFrame *)data = c->frame;
+    *got_frame_ptr = 1;
 
     return buf - avpkt->data;
 }

libavcodec/adxenc.c

@@ -107,6 +107,14 @@ static int adx_encode_header(AVCodecContext *avctx, uint8_t *buf, int bufsize)
     return HEADER_SIZE;
 }
 
+#if FF_API_OLD_ENCODE_AUDIO
+static av_cold int adx_encode_close(AVCodecContext *avctx)
+{
+    av_freep(&avctx->coded_frame);
+    return 0;
+}
+#endif
+
 static av_cold int adx_encode_init(AVCodecContext *avctx)
 {
     ADXContext *c = avctx->priv_data;
@@ -118,8 +126,8 @@ 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;
+    if (!(avctx->coded_frame = avcodec_alloc_frame()))
+        return AVERROR(ENOMEM);
 #endif
 
     /* the cutoff can be adjusted, but this seems to work pretty well */
@@ -167,6 +175,9 @@ AVCodec ff_adpcm_adx_encoder = {
     .id             = AV_CODEC_ID_ADPCM_ADX,
     .priv_data_size = sizeof(ADXContext),
     .init           = adx_encode_init,
+#if FF_API_OLD_ENCODE_AUDIO
+    .close          = adx_encode_close,
+#endif
     .encode2        = adx_encode_frame,
     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
                                                       AV_SAMPLE_FMT_NONE },

libavcodec/alac.c

@@ -58,7 +58,6 @@
 
 typedef struct {
     AVCodecContext *avctx;
-    AVFrame frame;
     GetBitContext gb;
     int channels;
 
@@ -254,7 +253,7 @@ static void append_extra_bits(int32_t *buffer[2], int32_t *extra_bits_buffer[2],
             buffer[ch][i] = (buffer[ch][i] << extra_bits) | extra_bits_buffer[ch][i];
 }
 
-static int decode_element(AVCodecContext *avctx, void *data, int ch_index,
+static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index,
                           int channels)
 {
     ALACContext *alac = avctx->priv_data;
@@ -289,8 +288,8 @@ static int decode_element(AVCodecContext *avctx, void *data, int ch_index,
     }
     if (!alac->nb_samples) {
         /* get output buffer */
-        alac->frame.nb_samples = output_samples;
-        if ((ret = ff_get_buffer(avctx, &alac->frame)) < 0) {
+        frame->nb_samples = output_samples;
+        if ((ret = ff_get_buffer(avctx, frame)) < 0) {
             av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
             return ret;
         }
@@ -302,7 +301,7 @@ static int decode_element(AVCodecContext *avctx, void *data, int ch_index,
     alac->nb_samples = output_samples;
     if (alac->direct_output) {
         for (ch = 0; ch < channels; ch++)
-            alac->output_samples_buffer[ch] = (int32_t *)alac->frame.extended_data[ch_index + ch];
+            alac->output_samples_buffer[ch] = (int32_t *)frame->extended_data[ch_index + ch];
     }
 
     if (is_compressed) {
@@ -390,7 +389,7 @@ static int decode_element(AVCodecContext *avctx, void *data, int ch_index,
     switch(alac->sample_size) {
     case 16: {
         for (ch = 0; ch < channels; ch++) {
-            int16_t *outbuffer = (int16_t *)alac->frame.extended_data[ch_index + ch];
+            int16_t *outbuffer = (int16_t *)frame->extended_data[ch_index + ch];
             for (i = 0; i < alac->nb_samples; i++)
                 *outbuffer++ = alac->output_samples_buffer[ch][i];
         }}
@@ -405,7 +404,7 @@ static int decode_element(AVCodecContext *avctx, void *data, int ch_index,
     }else{
         switch(alac->sample_size) {
         case 16: {
-            int16_t *outbuffer = ((int16_t *)alac->frame.extended_data[0]) + ch_index;
+            int16_t *outbuffer = ((int16_t *)frame->extended_data[0]) + ch_index;
             for (i = 0; i < alac->nb_samples; i++) {
                 for (ch = 0; ch < channels; ch++)
                     *outbuffer++ = alac->output_samples_buffer[ch][i];
@@ -414,7 +413,7 @@ static int decode_element(AVCodecContext *avctx, void *data, int ch_index,
             }
             break;
         case 24: {
-            int32_t *outbuffer = ((int32_t *)alac->frame.extended_data[0]) + ch_index;
+            int32_t *outbuffer = ((int32_t *)frame->extended_data[0]) + ch_index;
             for (i = 0; i < alac->nb_samples; i++) {
                 for (ch = 0; ch < channels; ch++)
                     *outbuffer++ = alac->output_samples_buffer[ch][i] << 8;
@@ -423,7 +422,7 @@ static int decode_element(AVCodecContext *avctx, void *data, int ch_index,
             }
             break;
         case 32: {
-            int32_t *outbuffer = ((int32_t *)alac->frame.extended_data[0]) + ch_index;
+            int32_t *outbuffer = ((int32_t *)frame->extended_data[0]) + ch_index;
             for (i = 0; i < alac->nb_samples; i++) {
                 for (ch = 0; ch < channels; ch++)
                     *outbuffer++ = alac->output_samples_buffer[ch][i];
@@ -441,6 +440,7 @@ static int alac_decode_frame(AVCodecContext *avctx, void *data,
                              int *got_frame_ptr, AVPacket *avpkt)
 {
     ALACContext *alac = avctx->priv_data;
+    AVFrame *frame    = data;
     enum AlacRawDataBlockType element;
     int channels;
     int ch, ret, got_end;
@@ -469,7 +469,7 @@ static int alac_decode_frame(AVCodecContext *avctx, void *data,
             return AVERROR_INVALIDDATA;
         }
 
-        ret = decode_element(avctx, data,
+        ret = decode_element(avctx, frame,
                              ff_alac_channel_layout_offsets[alac->channels - 1][ch],
                              channels);
         if (ret < 0 && get_bits_left(&alac->gb))
@@ -487,8 +487,7 @@ static int alac_decode_frame(AVCodecContext *avctx, void *data,
                avpkt->size * 8 - get_bits_count(&alac->gb));
     }
 
-    *got_frame_ptr   = 1;
-    *(AVFrame *)data = alac->frame;
+    *got_frame_ptr = 1;
 
     return avpkt->size;
 }
@@ -616,9 +615,6 @@ static av_cold int alac_decode_init(AVCodecContext * avctx)
         return ret;
     }
 
-    avcodec_get_frame_defaults(&alac->frame);
-    avctx->coded_frame = &alac->frame;
-
     return 0;
 }
 

libavcodec/alacenc.c

@@ -21,7 +21,6 @@
 
 #include "avcodec.h"
 #include "put_bits.h"
-#include "dsputil.h"
 #include "internal.h"
 #include "lpc.h"
 #include "mathops.h"
@@ -275,7 +274,7 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch)
         // generate warm-up samples
         residual[0] = samples[0];
         for (i = 1; i <= lpc.lpc_order; i++)
-            residual[i] = samples[i] - samples[i-1];
+            residual[i] = sign_extend(samples[i] - samples[i-1], s->write_sample_size);
 
         // perform lpc on remaining samples
         for (i = lpc.lpc_order + 1; i < s->frame_size; i++) {
@@ -614,7 +613,7 @@ static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     else
         max_frame_size = s->max_coded_frame_size;
 
-    if ((ret = ff_alloc_packet2(avctx, avpkt, 2 * max_frame_size)))
+    if ((ret = ff_alloc_packet2(avctx, avpkt, 2 * max_frame_size)) < 0)
         return ret;
 
     /* use verbatim mode for compression_level 0 */

libavcodec/allcodecs.c

@@ -74,18 +74,23 @@ void avcodec_register_all(void)
 
     /* hardware accelerators */
     REGISTER_HWACCEL(H263_VAAPI,        h263_vaapi);
+    REGISTER_HWACCEL(H263_VDPAU,        h263_vdpau);
     REGISTER_HWACCEL(H264_DXVA2,        h264_dxva2);
     REGISTER_HWACCEL(H264_VAAPI,        h264_vaapi);
     REGISTER_HWACCEL(H264_VDA,          h264_vda);
+    REGISTER_HWACCEL(H264_VDPAU,        h264_vdpau);
     REGISTER_HWACCEL(MPEG1_VDPAU,       mpeg1_vdpau);
     REGISTER_HWACCEL(MPEG2_DXVA2,       mpeg2_dxva2);
     REGISTER_HWACCEL(MPEG2_VAAPI,       mpeg2_vaapi);
     REGISTER_HWACCEL(MPEG2_VDPAU,       mpeg2_vdpau);
     REGISTER_HWACCEL(MPEG4_VAAPI,       mpeg4_vaapi);
+    REGISTER_HWACCEL(MPEG4_VDPAU,       mpeg4_vdpau);
     REGISTER_HWACCEL(VC1_DXVA2,         vc1_dxva2);
     REGISTER_HWACCEL(VC1_VAAPI,         vc1_vaapi);
+    REGISTER_HWACCEL(VC1_VDPAU,         vc1_vdpau);
     REGISTER_HWACCEL(WMV3_DXVA2,        wmv3_dxva2);
     REGISTER_HWACCEL(WMV3_VAAPI,        wmv3_vaapi);
+    REGISTER_HWACCEL(WMV3_VDPAU,        wmv3_vdpau);
 
     /* video codecs */
     REGISTER_ENCODER(A64MULTI,          a64multi);
@@ -318,6 +323,7 @@ void avcodec_register_all(void)
     REGISTER_ENCDEC (DCA,               dca);
     REGISTER_DECODER(DSICINAUDIO,       dsicinaudio);
     REGISTER_ENCDEC (EAC3,              eac3);
+    REGISTER_DECODER(EVRC,              evrc);
     REGISTER_DECODER(FFWAVESYNTH,       ffwavesynth);
     REGISTER_ENCDEC (FLAC,              flac);
     REGISTER_ENCDEC (G723_1,            g723_1);
@@ -483,7 +489,8 @@ void avcodec_register_all(void)
     REGISTER_ENCODER(LIBVO_AACENC,      libvo_aacenc);
     REGISTER_ENCODER(LIBVO_AMRWBENC,    libvo_amrwbenc);
     REGISTER_ENCDEC (LIBVORBIS,         libvorbis);
-    REGISTER_ENCDEC (LIBVPX,            libvpx);
+    REGISTER_ENCDEC (LIBVPX_VP8,        libvpx_vp8);
+    REGISTER_ENCDEC (LIBVPX_VP9,        libvpx_vp9);
     REGISTER_ENCODER(LIBX264,           libx264);
     REGISTER_ENCODER(LIBX264RGB,        libx264rgb);
     REGISTER_ENCODER(LIBXAVS,           libxavs);

libavcodec/alpha/dsputil_alpha.c

@@ -19,20 +19,21 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
+#include "libavutil/attributes.h"
 #include "libavcodec/dsputil.h"
 #include "dsputil_alpha.h"
 #include "asm.h"
 
-void (*put_pixels_clamped_axp_p)(const DCTELEM *block, uint8_t *pixels,
+void (*put_pixels_clamped_axp_p)(const int16_t *block, uint8_t *pixels,
                                  int line_size);
-void (*add_pixels_clamped_axp_p)(const DCTELEM *block, uint8_t *pixels,
+void (*add_pixels_clamped_axp_p)(const int16_t *block, uint8_t *pixels,
                                  int line_size);
 
 #if 0
 /* These functions were the base for the optimized assembler routines,
    and remain here for documentation purposes.  */
-static void put_pixels_clamped_mvi(const DCTELEM *block, uint8_t *pixels,
-                                   int line_size)
+static void put_pixels_clamped_mvi(const int16_t *block, uint8_t *pixels,
+                                   ptrdiff_t line_size)
 {
     int i = 8;
     uint64_t clampmask = zap(-1, 0xaa); /* 0x00ff00ff00ff00ff */
@@ -55,8 +56,8 @@ static void put_pixels_clamped_mvi(const DCTELEM *block, uint8_t *pixels,
     } while (--i);
 }
 
-void add_pixels_clamped_mvi(const DCTELEM *block, uint8_t *pixels,
-                            int line_size)
+void add_pixels_clamped_mvi(const int16_t *block, uint8_t *pixels,
+                            ptrdiff_t line_size)
 {
     int h = 8;
     /* Keep this function a leaf function by generating the constants
@@ -100,9 +101,9 @@ void add_pixels_clamped_mvi(const DCTELEM *block, uint8_t *pixels,
 }
 #endif
 
-static void clear_blocks_axp(DCTELEM *blocks) {
+static void clear_blocks_axp(int16_t *blocks) {
     uint64_t *p = (uint64_t *) blocks;
-    int n = sizeof(DCTELEM) * 6 * 64;
+    int n = sizeof(int16_t) * 6 * 64;
 
     do {
         p[0] = 0;
@@ -211,8 +212,8 @@ static inline uint64_t avg4(uint64_t l1, uint64_t l2, uint64_t l3, uint64_t l4)
 
 #define MAKE_OP(OPNAME, SUFF, OPKIND, STORE)                                \
 static void OPNAME ## _pixels ## SUFF ## _axp                               \
-        (uint8_t *av_restrict block, const uint8_t *av_restrict pixels,           \
-         int line_size, int h)                                              \
+        (uint8_t *restrict block, const uint8_t *restrict pixels,           \
+         ptrdiff_t line_size, int h)                                        \
 {                                                                           \
     if ((size_t) pixels & 0x7) {                                            \
         OPKIND(uldq, STORE);                                                \
@@ -222,8 +223,8 @@ static void OPNAME ## _pixels ## SUFF ## _axp                               \
 }                                                                           \
                                                                             \
 static void OPNAME ## _pixels16 ## SUFF ## _axp                             \
-        (uint8_t *av_restrict block, const uint8_t *av_restrict pixels,           \
-         int line_size, int h)                                              \
+        (uint8_t *restrict block, const uint8_t *restrict pixels,           \
+         ptrdiff_t line_size, int h)                                        \
 {                                                                           \
     OPNAME ## _pixels ## SUFF ## _axp(block,     pixels,     line_size, h); \
     OPNAME ## _pixels ## SUFF ## _axp(block + 8, pixels + 8, line_size, h); \
@@ -262,13 +263,13 @@ PIXOP(put_no_rnd, STORE);
 PIXOP(avg_no_rnd, STORE);
 
 static void put_pixels16_axp_asm(uint8_t *block, const uint8_t *pixels,
-                                 int line_size, int h)
+                                 ptrdiff_t line_size, int h)
 {
     put_pixels_axp_asm(block,     pixels,     line_size, h);
     put_pixels_axp_asm(block + 8, pixels + 8, line_size, h);
 }
 
-void ff_dsputil_init_alpha(DSPContext* c, AVCodecContext *avctx)
+av_cold void ff_dsputil_init_alpha(DSPContext *c, AVCodecContext *avctx)
 {
     const int high_bit_depth = avctx->bits_per_raw_sample > 8;
 
@@ -288,10 +289,10 @@ void ff_dsputil_init_alpha(DSPContext* c, AVCodecContext *avctx)
     c->avg_pixels_tab[0][2] = avg_pixels16_y2_axp;
     c->avg_pixels_tab[0][3] = avg_pixels16_xy2_axp;
 
-    c->avg_no_rnd_pixels_tab[0][0] = avg_no_rnd_pixels16_axp;
-    c->avg_no_rnd_pixels_tab[0][1] = avg_no_rnd_pixels16_x2_axp;
-    c->avg_no_rnd_pixels_tab[0][2] = avg_no_rnd_pixels16_y2_axp;
-    c->avg_no_rnd_pixels_tab[0][3] = avg_no_rnd_pixels16_xy2_axp;
+    c->avg_no_rnd_pixels_tab[0] = avg_no_rnd_pixels16_axp;
+    c->avg_no_rnd_pixels_tab[1] = avg_no_rnd_pixels16_x2_axp;
+    c->avg_no_rnd_pixels_tab[2] = avg_no_rnd_pixels16_y2_axp;
+    c->avg_no_rnd_pixels_tab[3] = avg_no_rnd_pixels16_xy2_axp;
 
     c->put_pixels_tab[1][0] = put_pixels_axp_asm;
     c->put_pixels_tab[1][1] = put_pixels_x2_axp;
@@ -308,11 +309,6 @@ void ff_dsputil_init_alpha(DSPContext* c, AVCodecContext *avctx)
     c->avg_pixels_tab[1][2] = avg_pixels_y2_axp;
     c->avg_pixels_tab[1][3] = avg_pixels_xy2_axp;
 
-    c->avg_no_rnd_pixels_tab[1][0] = avg_no_rnd_pixels_axp;
-    c->avg_no_rnd_pixels_tab[1][1] = avg_no_rnd_pixels_x2_axp;
-    c->avg_no_rnd_pixels_tab[1][2] = avg_no_rnd_pixels_y2_axp;
-    c->avg_no_rnd_pixels_tab[1][3] = avg_no_rnd_pixels_xy2_axp;
-
     c->clear_blocks = clear_blocks_axp;
     }
 

libavcodec/alpha/dsputil_alpha.h

@@ -19,26 +19,27 @@
 #ifndef AVCODEC_ALPHA_DSPUTIL_ALPHA_H
 #define AVCODEC_ALPHA_DSPUTIL_ALPHA_H
 
-#include "libavcodec/dsputil.h"
+#include <stddef.h>
+#include <stdint.h>
 
-void ff_simple_idct_axp(DCTELEM *block);
-void ff_simple_idct_put_axp(uint8_t *dest, int line_size, DCTELEM *block);
-void ff_simple_idct_add_axp(uint8_t *dest, int line_size, DCTELEM *block);
+void ff_simple_idct_axp(int16_t *block);
+void ff_simple_idct_put_axp(uint8_t *dest, int line_size, int16_t *block);
+void ff_simple_idct_add_axp(uint8_t *dest, int line_size, int16_t *block);
 
 void put_pixels_axp_asm(uint8_t *block, const uint8_t *pixels,
-                        int line_size, int h);
-void put_pixels_clamped_mvi_asm(const DCTELEM *block, uint8_t *pixels,
+                        ptrdiff_t line_size, int h);
+void put_pixels_clamped_mvi_asm(const int16_t *block, uint8_t *pixels,
                                 int line_size);
-void add_pixels_clamped_mvi_asm(const DCTELEM *block, uint8_t *pixels,
+void add_pixels_clamped_mvi_asm(const int16_t *block, uint8_t *pixels,
                                 int line_size);
-extern void (*put_pixels_clamped_axp_p)(const DCTELEM *block, uint8_t *pixels,
+extern void (*put_pixels_clamped_axp_p)(const int16_t *block, uint8_t *pixels,
                                         int line_size);
-extern void (*add_pixels_clamped_axp_p)(const DCTELEM *block, uint8_t *pixels,
+extern void (*add_pixels_clamped_axp_p)(const int16_t *block, uint8_t *pixels,
                                         int line_size);
 
-void get_pixels_mvi(DCTELEM *av_restrict block,
-                    const uint8_t *av_restrict pixels, int line_size);
-void diff_pixels_mvi(DCTELEM *block, const uint8_t *s1, const uint8_t *s2,
+void get_pixels_mvi(int16_t *restrict block,
+                    const uint8_t *restrict pixels, int line_size);
+void diff_pixels_mvi(int16_t *block, const uint8_t *s1, const uint8_t *s2,
                      int stride);
 int pix_abs8x8_mvi(void *v, uint8_t *pix1, uint8_t *pix2, int line_size, int h);
 int pix_abs16x16_mvi_asm(void *v, uint8_t *pix1, uint8_t *pix2, int line_size, int h);

libavcodec/alpha/dsputil_alpha_asm.S

@@ -135,7 +135,7 @@ $aligned:
         .end put_pixels_axp_asm
 
 /************************************************************************
- * void put_pixels_clamped_mvi_asm(const DCTELEM *block, uint8_t *pixels,
+ * void put_pixels_clamped_mvi_asm(const int16_t *block, uint8_t *pixels,
  *                                 int line_size)
  */
         .align 6
@@ -185,7 +185,7 @@ put_pixels_clamped_mvi_asm:
         .end put_pixels_clamped_mvi_asm
 
 /************************************************************************
- * void add_pixels_clamped_mvi_asm(const DCTELEM *block, uint8_t *pixels,
+ * void add_pixels_clamped_mvi_asm(const int16_t *block, uint8_t *pixels,
  *                                 int line_size)
  */
         .align 6

libavcodec/alpha/motion_est_alpha.c

@@ -19,12 +19,11 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include "libavcodec/dsputil.h"
 #include "dsputil_alpha.h"
 #include "asm.h"
 
-void get_pixels_mvi(DCTELEM *av_restrict block,
-                    const uint8_t *av_restrict pixels, int line_size)
+void get_pixels_mvi(int16_t *restrict block,
+                    const uint8_t *restrict pixels, int line_size)
 {
     int h = 8;
 
@@ -40,7 +39,7 @@ void get_pixels_mvi(DCTELEM *av_restrict block,
     } while (--h);
 }
 
-void diff_pixels_mvi(DCTELEM *block, const uint8_t *s1, const uint8_t *s2,
+void diff_pixels_mvi(int16_t *block, const uint8_t *s1, const uint8_t *s2,
                      int stride) {
     int h = 8;
     uint64_t mask = 0x4040;

libavcodec/alpha/mpegvideo_alpha.c

@@ -19,11 +19,11 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include "libavcodec/dsputil.h"
+#include "libavutil/attributes.h"
 #include "libavcodec/mpegvideo.h"
 #include "asm.h"
 
-static void dct_unquantize_h263_axp(DCTELEM *block, int n_coeffs,
+static void dct_unquantize_h263_axp(int16_t *block, int n_coeffs,
                                     uint64_t qscale, uint64_t qadd)
 {
     uint64_t qmul = qscale << 1;
@@ -69,12 +69,12 @@ static void dct_unquantize_h263_axp(DCTELEM *block, int n_coeffs,
     }
 }
 
-static void dct_unquantize_h263_intra_axp(MpegEncContext *s, DCTELEM *block,
+static void dct_unquantize_h263_intra_axp(MpegEncContext *s, int16_t *block,
                                     int n, int qscale)
 {
     int n_coeffs;
     uint64_t qadd;
-    DCTELEM block0 = block[0];
+    int16_t block0 = block[0];
 
     if (!s->h263_aic) {
         if (n < 4)
@@ -96,14 +96,14 @@ static void dct_unquantize_h263_intra_axp(MpegEncContext *s, DCTELEM *block,
     block[0] = block0;
 }
 
-static void dct_unquantize_h263_inter_axp(MpegEncContext *s, DCTELEM *block,
+static void dct_unquantize_h263_inter_axp(MpegEncContext *s, int16_t *block,
                                     int n, int qscale)
 {
     int n_coeffs = s->inter_scantable.raster_end[s->block_last_index[n]];
     dct_unquantize_h263_axp(block, n_coeffs, qscale, (qscale - 1) | 1);
 }
 
-void ff_MPV_common_init_axp(MpegEncContext *s)
+av_cold void ff_MPV_common_init_axp(MpegEncContext *s)
 {
     s->dct_unquantize_h263_intra = dct_unquantize_h263_intra_axp;
     s->dct_unquantize_h263_inter = dct_unquantize_h263_inter_axp;

libavcodec/alpha/simple_idct_alpha.c

@@ -26,7 +26,6 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include "libavcodec/dsputil.h"
 #include "dsputil_alpha.h"
 #include "asm.h"
 
@@ -44,7 +43,7 @@
 #define COL_SHIFT 20
 
 /* 0: all entries 0, 1: only first entry nonzero, 2: otherwise  */
-static inline int idct_row(DCTELEM *row)
+static inline int idct_row(int16_t *row)
 {
     int a0, a1, a2, a3, b0, b1, b2, b3, t;
     uint64_t l, r, t2;
@@ -152,7 +151,7 @@ static inline int idct_row(DCTELEM *row)
     return 2;
 }
 
-static inline void idct_col(DCTELEM *col)
+static inline void idct_col(int16_t *col)
 {
     int a0, a1, a2, a3, b0, b1, b2, b3;
 
@@ -229,7 +228,7 @@ static inline void idct_col(DCTELEM *col)
 
 /* If all rows but the first one are zero after row transformation,
    all rows will be identical after column transformation.  */
-static inline void idct_col2(DCTELEM *col)
+static inline void idct_col2(int16_t *col)
 {
     int i;
     uint64_t l, r;
@@ -251,7 +250,7 @@ static inline void idct_col2(DCTELEM *col)
     stq(l, col + 14 * 4); stq(r, col + 15 * 4);
 }
 
-void ff_simple_idct_axp(DCTELEM *block)
+void ff_simple_idct_axp(int16_t *block)
 {
 
     int i;
@@ -291,13 +290,13 @@ void ff_simple_idct_axp(DCTELEM *block)
     }
 }
 
-void ff_simple_idct_put_axp(uint8_t *dest, int line_size, DCTELEM *block)
+void ff_simple_idct_put_axp(uint8_t *dest, int line_size, int16_t *block)
 {
     ff_simple_idct_axp(block);
     put_pixels_clamped_axp_p(block, dest, line_size);
 }
 
-void ff_simple_idct_add_axp(uint8_t *dest, int line_size, DCTELEM *block)
+void ff_simple_idct_add_axp(uint8_t *dest, int line_size, int16_t *block)
 {
     ff_simple_idct_axp(block);
     add_pixels_clamped_axp_p(block, dest, line_size);

libavcodec/alsdec.c

@@ -192,7 +192,6 @@ typedef struct {
 
 typedef struct {
     AVCodecContext *avctx;
-    AVFrame frame;
     ALSSpecificConfig sconf;
     GetBitContext gb;
     DSPContext dsp;
@@ -285,12 +284,14 @@ static av_cold int read_specific_config(ALSDecContext *ctx)
     GetBitContext gb;
     uint64_t ht_size;
     int i, config_offset;
-    MPEG4AudioConfig m4ac;
+    MPEG4AudioConfig m4ac = {0};
     ALSSpecificConfig *sconf = &ctx->sconf;
     AVCodecContext *avctx    = ctx->avctx;
     uint32_t als_id, header_size, trailer_size;
+    int ret;
 
-    init_get_bits(&gb, avctx->extradata, avctx->extradata_size * 8);
+    if ((ret = init_get_bits8(&gb, avctx->extradata, avctx->extradata_size)) < 0)
+        return ret;
 
     config_offset = avpriv_mpeg4audio_get_config(&m4ac, avctx->extradata,
                                                  avctx->extradata_size * 8, 1);
@@ -1448,6 +1449,7 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
                         AVPacket *avpkt)
 {
     ALSDecContext *ctx       = avctx->priv_data;
+    AVFrame *frame           = data;
     ALSSpecificConfig *sconf = &ctx->sconf;
     const uint8_t *buffer    = avpkt->data;
     int buffer_size          = avpkt->size;
@@ -1477,8 +1479,8 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
     ctx->frame_id++;
 
     /* get output buffer */
-    ctx->frame.nb_samples = ctx->cur_frame_length;
-    if ((ret = ff_get_buffer(avctx, &ctx->frame)) < 0) {
+    frame->nb_samples = ctx->cur_frame_length;
+    if ((ret = ff_get_buffer(avctx, frame)) < 0) {
         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed.\n");
         return ret;
     }
@@ -1486,7 +1488,7 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
     // transform decoded frame into output format
     #define INTERLEAVE_OUTPUT(bps)                                                   \
     {                                                                                \
-        int##bps##_t *dest = (int##bps##_t*)ctx->frame.data[0];                      \
+        int##bps##_t *dest = (int##bps##_t*)frame->data[0];                          \
         shift = bps - ctx->avctx->bits_per_raw_sample;                               \
         if (!ctx->cs_switch) {                                                       \
             for (sample = 0; sample < ctx->cur_frame_length; sample++)               \
@@ -1510,7 +1512,7 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
         int swap = HAVE_BIGENDIAN != sconf->msb_first;
 
         if (ctx->avctx->bits_per_raw_sample == 24) {
-            int32_t *src = (int32_t *)ctx->frame.data[0];
+            int32_t *src = (int32_t *)frame->data[0];
 
             for (sample = 0;
                  sample < ctx->cur_frame_length * avctx->channels;
@@ -1531,7 +1533,7 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
 
             if (swap) {
                 if (ctx->avctx->bits_per_raw_sample <= 16) {
-                    int16_t *src  = (int16_t*) ctx->frame.data[0];
+                    int16_t *src  = (int16_t*) frame->data[0];
                     int16_t *dest = (int16_t*) ctx->crc_buffer;
                     for (sample = 0;
                          sample < ctx->cur_frame_length * avctx->channels;
@@ -1539,12 +1541,12 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
                         *dest++ = av_bswap16(src[sample]);
                 } else {
                     ctx->dsp.bswap_buf((uint32_t*)ctx->crc_buffer,
-                                       (uint32_t *)ctx->frame.data[0],
+                                       (uint32_t *)frame->data[0],
                                        ctx->cur_frame_length * avctx->channels);
                 }
                 crc_source = ctx->crc_buffer;
             } else {
-                crc_source = ctx->frame.data[0];
+                crc_source = frame->data[0];
             }
 
             ctx->crc = av_crc(ctx->crc_table, ctx->crc, crc_source,
@@ -1560,9 +1562,7 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
         }
     }
 
-    *got_frame_ptr   = 1;
-    *(AVFrame *)data = ctx->frame;
-
+    *got_frame_ptr = 1;
 
     bytes_read = invalid_frame ? buffer_size :
                                  (get_bits_count(&ctx->gb) + 7) >> 3;
@@ -1759,9 +1759,6 @@ static av_cold int decode_init(AVCodecContext *avctx)
 
     ff_dsputil_init(&ctx->dsp, avctx);
 
-    avcodec_get_frame_defaults(&ctx->frame);
-    avctx->coded_frame = &ctx->frame;
-
     return 0;
 }
 

libavcodec/amrnbdec.c

@@ -44,8 +44,8 @@
 #include <math.h>
 
 #include "libavutil/channel_layout.h"
+#include "libavutil/float_dsp.h"
 #include "avcodec.h"
-#include "dsputil.h"
 #include "libavutil/common.h"
 #include "libavutil/avassert.h"
 #include "celp_math.h"
@@ -98,7 +98,6 @@
 #define AMR_AGC_ALPHA      0.9
 
 typedef struct AMRContext {
-    AVFrame                         avframe; ///< AVFrame for decoded samples
     AMRNBFrame                        frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc)
     uint8_t             bad_frame_indicator; ///< bad frame ? 1 : 0
     enum Mode                cur_frame_mode;
@@ -185,9 +184,6 @@ static av_cold int amrnb_decode_init(AVCodecContext *avctx)
     for (i = 0; i < 4; i++)
         p->prediction_error[i] = MIN_ENERGY;
 
-    avcodec_get_frame_defaults(&p->avframe);
-    avctx->coded_frame = &p->avframe;
-
     ff_acelp_filter_init(&p->acelpf_ctx);
     ff_acelp_vectors_init(&p->acelpv_ctx);
     ff_celp_filter_init(&p->celpf_ctx);
@@ -810,7 +806,7 @@ static int synthesis(AMRContext *p, float *lpc,
     // emphasize pitch vector contribution
     if (p->pitch_gain[4] > 0.5 && !overflow) {
         float energy = p->celpm_ctx.dot_productf(excitation, excitation,
-                                                AMR_SUBFRAME_SIZE);
+                                                    AMR_SUBFRAME_SIZE);
         float pitch_factor =
             p->pitch_gain[4] *
             (p->cur_frame_mode == MODE_12k2 ?
@@ -911,7 +907,7 @@ static void postfilter(AMRContext *p, float *lpc, float *buf_out)
     float *samples          = p->samples_in + LP_FILTER_ORDER; // Start of input
 
     float speech_gain       = p->celpm_ctx.dot_productf(samples, samples,
-                                                       AMR_SUBFRAME_SIZE);
+                                                           AMR_SUBFRAME_SIZE);
 
     float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER];  // Output of pole filter
     const float *gamma_n, *gamma_d;                       // Formant filter factor table
@@ -954,6 +950,7 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
 {
 
     AMRContext *p = avctx->priv_data;        // pointer to private data
+    AVFrame *frame     = data;
     const uint8_t *buf = avpkt->data;
     int buf_size       = avpkt->size;
     float *buf_out;                          // pointer to the output data buffer
@@ -965,12 +962,12 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
     const float *synth_fixed_vector;         // pointer to the fixed vector that synthesis should use
 
     /* get output buffer */
-    p->avframe.nb_samples = AMR_BLOCK_SIZE;
-    if ((ret = ff_get_buffer(avctx, &p->avframe)) < 0) {
+    frame->nb_samples = AMR_BLOCK_SIZE;
+    if ((ret = ff_get_buffer(avctx, frame)) < 0) {
         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
         return ret;
     }
-    buf_out = (float *)p->avframe.data[0];
+    buf_out = (float *)frame->data[0];
 
     p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);
     if (p->cur_frame_mode == NO_DATA) {
@@ -1018,8 +1015,8 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
         p->fixed_gain[4] =
             ff_amr_set_fixed_gain(fixed_gain_factor,
                        p->celpm_ctx.dot_productf(p->fixed_vector,
-                                                           p->fixed_vector,
-                                                           AMR_SUBFRAME_SIZE) /
+                                                               p->fixed_vector,
+                                                               AMR_SUBFRAME_SIZE) /
                                   AMR_SUBFRAME_SIZE,
                        p->prediction_error,
                        energy_mean[p->cur_frame_mode], energy_pred_fac);
@@ -1078,8 +1075,7 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
     p->acelpv_ctx.weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
                             0.84, 0.16, LP_FILTER_ORDER);
 
-    *got_frame_ptr   = 1;
-    *(AVFrame *)data = p->avframe;
+    *got_frame_ptr = 1;
 
     /* return the amount of bytes consumed if everything was OK */
     return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC

libavcodec/amrwbdec.c

@@ -26,10 +26,10 @@
 
 #include "libavutil/channel_layout.h"
 #include "libavutil/common.h"
+#include "libavutil/float_dsp.h"
 #include "libavutil/lfg.h"
 
 #include "avcodec.h"
-#include "dsputil.h"
 #include "lsp.h"
 #include "celp_filters.h"
 #include "celp_math.h"
@@ -45,7 +45,6 @@
 #include "mips/amrwbdec_mips.h"
 
 typedef struct {
-    AVFrame                              avframe; ///< AVFrame for decoded samples
     AMRWBFrame                             frame; ///< AMRWB parameters decoded from bitstream
     enum Mode                        fr_cur_mode; ///< mode index of current frame
     uint8_t                           fr_quality; ///< frame quality index (FQI)
@@ -121,9 +120,6 @@ static av_cold int amrwb_decode_init(AVCodecContext *avctx)
     for (i = 0; i < 4; i++)
         ctx->prediction_error[i] = MIN_ENERGY;
 
-    avcodec_get_frame_defaults(&ctx->avframe);
-    avctx->coded_frame = &ctx->avframe;
-
     ff_acelp_filter_init(&ctx->acelpf_ctx);
     ff_acelp_vectors_init(&ctx->acelpv_ctx);
     ff_celp_filter_init(&ctx->celpf_ctx);
@@ -612,11 +608,11 @@ static float voice_factor(float *p_vector, float p_gain,
                           CELPMContext *ctx)
 {
     double p_ener = (double) ctx->dot_productf(p_vector, p_vector,
-                                             AMRWB_SFR_SIZE) *
-                                             p_gain * p_gain;
+                                                          AMRWB_SFR_SIZE) *
+                    p_gain * p_gain;
     double f_ener = (double) ctx->dot_productf(f_vector, f_vector,
-                                             AMRWB_SFR_SIZE) *
-                                             f_gain * f_gain;
+                                                          AMRWB_SFR_SIZE) *
+                    f_gain * f_gain;
 
     return (p_ener - f_ener) / (p_ener + f_ener);
 }
@@ -785,7 +781,7 @@ static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
     if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) {
         int i;
         float energy = ctx->celpm_ctx.dot_productf(excitation, excitation,
-                                                AMRWB_SFR_SIZE);
+                                                    AMRWB_SFR_SIZE);
 
         // XXX: Weird part in both ref code and spec. A unknown parameter
         // {beta} seems to be identical to the current pitch gain
@@ -846,8 +842,8 @@ static void upsample_5_4(float *out, const float *in, int o_size, CELPMContext *
 
         for (k = 1; k < 5; k++) {
             out[i] = ctx->dot_productf(in0 + int_part,
-                                              upsample_fir[4 - frac_part],
-                                              UPS_MEM_SIZE);
+                                                  upsample_fir[4 - frac_part],
+                                                  UPS_MEM_SIZE);
             int_part++;
             frac_part--;
             i++;
@@ -893,7 +889,8 @@ static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
                                  const float *synth_exc, float hb_gain)
 {
     int i;
-    float energy = ctx->celpm_ctx.dot_productf(synth_exc, synth_exc, AMRWB_SFR_SIZE);
+    float energy = ctx->celpm_ctx.dot_productf(synth_exc, synth_exc,
+                                                AMRWB_SFR_SIZE);
 
     /* Generate a white-noise excitation */
     for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
@@ -1096,6 +1093,7 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
                               int *got_frame_ptr, AVPacket *avpkt)
 {
     AMRWBContext *ctx  = avctx->priv_data;
+    AVFrame *frame     = data;
     AMRWBFrame   *cf   = &ctx->frame;
     const uint8_t *buf = avpkt->data;
     int buf_size       = avpkt->size;
@@ -1113,12 +1111,12 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
     int sub, i, ret;
 
     /* get output buffer */
-    ctx->avframe.nb_samples = 4 * AMRWB_SFR_SIZE_16k;
-    if ((ret = ff_get_buffer(avctx, &ctx->avframe)) < 0) {
+    frame->nb_samples = 4 * AMRWB_SFR_SIZE_16k;
+    if ((ret = ff_get_buffer(avctx, frame)) < 0) {
         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
         return ret;
     }
-    buf_out = (float *)ctx->avframe.data[0];
+    buf_out = (float *)frame->data[0];
 
     header_size      = decode_mime_header(ctx, buf);
     if (ctx->fr_cur_mode > MODE_SID) {
@@ -1189,8 +1187,8 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
         ctx->fixed_gain[0] =
             ff_amr_set_fixed_gain(fixed_gain_factor,
                                   ctx->celpm_ctx.dot_productf(ctx->fixed_vector,
-                                                           ctx->fixed_vector,
-                                                           AMRWB_SFR_SIZE) /
+                                                               ctx->fixed_vector,
+                                                               AMRWB_SFR_SIZE) /
                                   AMRWB_SFR_SIZE,
                        ctx->prediction_error,
                        ENERGY_MEAN, energy_pred_fac);
@@ -1264,8 +1262,7 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
     memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0]));
     memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float));
 
-    *got_frame_ptr   = 1;
-    *(AVFrame *)data = ctx->avframe;
+    *got_frame_ptr = 1;
 
     return expected_fr_size;
 }

libavcodec/anm.c

@@ -114,7 +114,7 @@ static int decode_frame(AVCodecContext *avctx,
     uint8_t *dst, *dst_end;
     int count, ret;
 
-    if ((ret = avctx->reget_buffer(avctx, &s->frame)) < 0) {
+    if ((ret = avctx->reget_buffer(avctx, &s->frame)) < 0){
         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
         return ret;
     }

libavcodec/ansi.c

@@ -112,7 +112,7 @@ static void hscroll(AVCodecContext *avctx)
     AnsiContext *s = avctx->priv_data;
     int i;
 
-    if (s->y < avctx->height - s->font_height) {
+    if (s->y <= avctx->height - 2*s->font_height) {
         s->y += s->font_height;
         return;
     }
@@ -165,7 +165,7 @@ static void draw_char(AVCodecContext *avctx, int c)
     ff_draw_pc_font(s->frame.data[0] + s->y * s->frame.linesize[0] + s->x,
                     s->frame.linesize[0], s->font, s->font_height, c, fg, bg);
     s->x += FONT_WIDTH;
-    if (s->x >= avctx->width) {
+    if (s->x > avctx->width - FONT_WIDTH) {
         s->x = 0;
         hscroll(avctx);
     }
@@ -239,6 +239,8 @@ static int execute_code(AVCodecContext * avctx, int c)
         default:
             av_log_ask_for_sample(avctx, "unsupported screen mode\n");
         }
+        s->x = av_clip(s->x, 0, width  - FONT_WIDTH);
+        s->y = av_clip(s->y, 0, height - s->font_height);
         if (width != avctx->width || height != avctx->height) {
             if (s->frame.data[0])
                 avctx->release_buffer(avctx, &s->frame);
@@ -335,6 +337,8 @@ static int execute_code(AVCodecContext * avctx, int c)
         av_log_ask_for_sample(avctx, "unsupported escape code\n");
         break;
     }
+    s->x = av_clip(s->x, 0, avctx->width  - FONT_WIDTH);
+    s->y = av_clip(s->y, 0, avctx->height - s->font_height);
     return 0;
 }
 
@@ -415,7 +419,7 @@ static int decode_frame(AVCodecContext *avctx,
             switch(buf[0]) {
             case '0': case '1': case '2': case '3': case '4':
             case '5': case '6': case '7': case '8': case '9':
-                if (s->nb_args < MAX_NB_ARGS)
+                if (s->nb_args < MAX_NB_ARGS && s->args[s->nb_args] < 6553)
                     s->args[s->nb_args] = FFMAX(s->args[s->nb_args], 0) * 10 + buf[0] - '0';
                 break;
             case ';':
@@ -434,8 +438,8 @@ static int decode_frame(AVCodecContext *avctx,
                     av_log(avctx, AV_LOG_WARNING, "args overflow (%i)\n", s->nb_args);
                 if (s->nb_args < MAX_NB_ARGS && s->args[s->nb_args] >= 0)
                     s->nb_args++;
-                if (execute_code(avctx, buf[0]) < 0)
-                    return -1;
+                if ((ret = execute_code(avctx, buf[0])) < 0)
+                    return ret;
                 s->state = STATE_NORMAL;
             }
             break;

libavcodec/apedec.c

@@ -129,7 +129,6 @@ typedef struct APEPredictor {
 typedef struct APEContext {
     AVClass *class;                          ///< class for AVOptions
     AVCodecContext *avctx;
-    AVFrame frame;
     DSPContext dsp;
     int channels;
     int samples;                             ///< samples left to decode in current frame
@@ -235,9 +234,6 @@ static av_cold int ape_decode_init(AVCodecContext *avctx)
     ff_dsputil_init(&s->dsp, avctx);
     avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
 
-    avcodec_get_frame_defaults(&s->frame);
-    avctx->coded_frame = &s->frame;
-
     return 0;
 filter_alloc_fail:
     ape_decode_close(avctx);
@@ -826,6 +822,7 @@ static void ape_unpack_stereo(APEContext *ctx, int count)
 static int ape_decode_frame(AVCodecContext *avctx, void *data,
                             int *got_frame_ptr, AVPacket *avpkt)
 {
+    AVFrame *frame     = data;
     const uint8_t *buf = avpkt->data;
     APEContext *s = avctx->priv_data;
     uint8_t *sample8;
@@ -906,8 +903,8 @@ static int ape_decode_frame(AVCodecContext *avctx, void *data,
     s->decoded[1] = s->decoded_buffer + FFALIGN(blockstodecode, 8);
 
     /* get output buffer */
-    s->frame.nb_samples = blockstodecode;
-    if ((ret = ff_get_buffer(avctx, &s->frame)) < 0) {
+    frame->nb_samples = blockstodecode;
+    if ((ret = ff_get_buffer(avctx, frame)) < 0) {
         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
         return ret;
     }
@@ -929,21 +926,21 @@ static int ape_decode_frame(AVCodecContext *avctx, void *data,
     switch (s->bps) {
     case 8:
         for (ch = 0; ch < s->channels; ch++) {
-            sample8 = (uint8_t *)s->frame.data[ch];
+            sample8 = (uint8_t *)frame->data[ch];
             for (i = 0; i < blockstodecode; i++)
                 *sample8++ = (s->decoded[ch][i] + 0x80) & 0xff;
         }
         break;
     case 16:
         for (ch = 0; ch < s->channels; ch++) {
-            sample16 = (int16_t *)s->frame.data[ch];
+            sample16 = (int16_t *)frame->data[ch];
             for (i = 0; i < blockstodecode; i++)
                 *sample16++ = s->decoded[ch][i];
         }
         break;
     case 24:
         for (ch = 0; ch < s->channels; ch++) {
-            sample24 = (int32_t *)s->frame.data[ch];
+            sample24 = (int32_t *)frame->data[ch];
             for (i = 0; i < blockstodecode; i++)
                 *sample24++ = s->decoded[ch][i] << 8;
         }
@@ -952,8 +949,7 @@ static int ape_decode_frame(AVCodecContext *avctx, void *data,
 
     s->samples -= blockstodecode;
 
-    *got_frame_ptr   = 1;
-    *(AVFrame *)data = s->frame;
+    *got_frame_ptr = 1;
 
     return !s->samples ? avpkt->size : 0;
 }

libavcodec/arm/Makefile

@@ -1,3 +1,5 @@
+ARCH_HEADERS = mathops.h
+
 OBJS-$(CONFIG_AC3DSP)                  += arm/ac3dsp_init_arm.o         \
                                           arm/ac3dsp_arm.o
 
@@ -15,6 +17,7 @@ OBJS-$(CONFIG_MPEGAUDIODSP)            += arm/mpegaudiodsp_init_arm.o
 ARMV6-OBJS-$(CONFIG_MPEGAUDIODSP)      += arm/mpegaudiodsp_fixed_armv6.o
 
 OBJS-$(CONFIG_MPEGVIDEO)               += arm/mpegvideo_arm.o
+OBJS-$(CONFIG_VORBIS_DECODER)          += arm/vorbisdsp_init_arm.o
 OBJS-$(CONFIG_VP3DSP)                  += arm/vp3dsp_init_arm.o
 OBJS-$(CONFIG_VP5_DECODER)             += arm/vp56dsp_init_arm.o
 OBJS-$(CONFIG_VP6_DECODER)             += arm/vp56dsp_init_arm.o
@@ -23,8 +26,10 @@ ARMV6-OBJS-$(CONFIG_VP8_DECODER)       += arm/vp8_armv6.o               \
                                           arm/vp8dsp_init_armv6.o       \
                                           arm/vp8dsp_armv6.o
 
+OBJS-$(CONFIG_H264CHROMA)              += arm/h264chroma_init_arm.o
 OBJS-$(CONFIG_H264DSP)                 += arm/h264dsp_init_arm.o
 OBJS-$(CONFIG_H264PRED)                += arm/h264pred_init_arm.o
+OBJS-$(CONFIG_H264QPEL)                += arm/h264qpel_init_arm.o
 
 OBJS-$(CONFIG_RV30_DECODER)            += arm/rv34dsp_init_arm.o
 OBJS-$(CONFIG_RV40_DECODER)            += arm/rv34dsp_init_arm.o        \
@@ -55,9 +60,6 @@ ARMV6-OBJS                             += arm/dsputil_init_armv6.o      \
 
 VFP-OBJS-$(HAVE_ARMV6)                 += arm/fmtconvert_vfp.o
 
-VFP-OBJS                               += arm/dsputil_vfp.o             \
-                                          arm/dsputil_init_vfp.o        \
-
 NEON-OBJS-$(CONFIG_FFT)                += arm/fft_neon.o                \
                                           arm/fft_fixed_neon.o          \
 
@@ -66,12 +68,14 @@ NEON-OBJS-$(CONFIG_MDCT)               += arm/mdct_neon.o               \
 
 NEON-OBJS-$(CONFIG_RDFT)               += arm/rdft_neon.o               \
 
+NEON-OBJS-$(CONFIG_H264CHROMA)         += arm/h264cmc_neon.o
 NEON-OBJS-$(CONFIG_H264DSP)            += arm/h264dsp_neon.o            \
                                           arm/h264idct_neon.o           \
-                                          arm/h264cmc_neon.o            \
 
 NEON-OBJS-$(CONFIG_H264PRED)           += arm/h264pred_neon.o           \
 
+NEON-OBJS-$(CONFIG_H264QPEL)           += arm/h264qpel_neon.o           \
+
 NEON-OBJS-$(CONFIG_AC3DSP)             += arm/ac3dsp_neon.o
 
 NEON-OBJS-$(CONFIG_AAC_DECODER)        += arm/sbrdsp_neon.o             \
@@ -84,7 +88,8 @@ NEON-OBJS-$(CONFIG_MPEGVIDEO)          += arm/mpegvideo_neon.o
 NEON-OBJS-$(CONFIG_RV30_DECODER)       += arm/rv34dsp_neon.o
 NEON-OBJS-$(CONFIG_RV40_DECODER)       += arm/rv34dsp_neon.o            \
                                           arm/rv40dsp_neon.o            \
-                                          arm/h264cmc_neon.o            \
+
+NEON-OBJS-$(CONFIG_VORBIS_DECODER)     += arm/vorbisdsp_neon.o
 
 NEON-OBJS-$(CONFIG_VP3DSP)             += arm/vp3dsp_neon.o
 

libavcodec/arm/dca.h

@@ -22,8 +22,9 @@
 #define AVCODEC_ARM_DCA_H
 
 #include <stdint.h>
+
 #include "config.h"
-#include "libavutil/intmath.h"
+#include "libavcodec/mathops.h"
 
 #if HAVE_ARMV6_INLINE && AV_GCC_VERSION_AT_LEAST(4,4) && !CONFIG_THUMB
 
@@ -33,46 +34,44 @@ static inline int decode_blockcodes(int code1, int code2, int levels,
 {
     int v0, v1, v2, v3, v4, v5;
 
-    __asm__ ("smmul   %8,  %14, %18           \n"
-             "smmul   %11, %15, %18           \n"
-             "smlabb  %14, %8,  %17, %14      \n"
-             "smlabb  %15, %11, %17, %15      \n"
-             "smmul   %9,  %8,  %18           \n"
-             "smmul   %12, %11, %18           \n"
-             "sub     %14, %14, %16, lsr #1   \n"
-             "sub     %15, %15, %16, lsr #1   \n"
-             "smlabb  %8,  %9,  %17, %8       \n"
-             "smlabb  %11, %12, %17, %11      \n"
-             "smmul   %10, %9,  %18           \n"
-             "smmul   %13, %12, %18           \n"
-             "str     %14, %0                 \n"
-             "str     %15, %4                 \n"
-             "sub     %8,  %8,  %16, lsr #1   \n"
-             "sub     %11, %11, %16, lsr #1   \n"
-             "smlabb  %9,  %10, %17, %9       \n"
-             "smlabb  %12, %13, %17, %12      \n"
-             "smmul   %14, %10, %18           \n"
-             "smmul   %15, %13, %18           \n"
-             "str     %8,  %1                 \n"
-             "str     %11, %5                 \n"
-             "sub     %9,  %9,  %16, lsr #1   \n"
-             "sub     %12, %12, %16, lsr #1   \n"
-             "smlabb  %10, %14, %17, %10      \n"
-             "smlabb  %13, %15, %17, %13      \n"
-             "str     %9,  %2                 \n"
-             "str     %12, %6                 \n"
-             "sub     %10, %10, %16, lsr #1   \n"
-             "sub     %13, %13, %16, lsr #1   \n"
-             "str     %10, %3                 \n"
-             "str     %13, %7                 \n"
-             : "=m"(values[0]), "=m"(values[1]),
-               "=m"(values[2]), "=m"(values[3]),
-               "=m"(values[4]), "=m"(values[5]),
-               "=m"(values[6]), "=m"(values[7]),
-               "=&r"(v0), "=&r"(v1), "=&r"(v2),
+    __asm__ ("smmul   %0,  %6,  %10           \n"
+             "smmul   %3,  %7,  %10           \n"
+             "smlabb  %6,  %0,  %9,  %6       \n"
+             "smlabb  %7,  %3,  %9,  %7       \n"
+             "smmul   %1,  %0,  %10           \n"
+             "smmul   %4,  %3,  %10           \n"
+             "sub     %6,  %6,  %8,  lsr #1   \n"
+             "sub     %7,  %7,  %8,  lsr #1   \n"
+             "smlabb  %0,  %1,  %9,  %0       \n"
+             "smlabb  %3,  %4,  %9,  %3       \n"
+             "smmul   %2,  %1,  %10           \n"
+             "smmul   %5,  %4,  %10           \n"
+             "str     %6,  [%11, #0]          \n"
+             "str     %7,  [%11, #16]         \n"
+             "sub     %0,  %0,  %8,  lsr #1   \n"
+             "sub     %3,  %3,  %8,  lsr #1   \n"
+             "smlabb  %1,  %2,  %9,  %1       \n"
+             "smlabb  %4,  %5,  %9,  %4       \n"
+             "smmul   %6,  %2,  %10           \n"
+             "smmul   %7,  %5,  %10           \n"
+             "str     %0,  [%11, #4]          \n"
+             "str     %3,  [%11, #20]         \n"
+             "sub     %1,  %1,  %8,  lsr #1   \n"
+             "sub     %4,  %4,  %8,  lsr #1   \n"
+             "smlabb  %2,  %6,  %9,  %2       \n"
+             "smlabb  %5,  %7,  %9,  %5       \n"
+             "str     %1,  [%11, #8]          \n"
+             "str     %4,  [%11, #24]         \n"
+             "sub     %2,  %2,  %8,  lsr #1   \n"
+             "sub     %5,  %5,  %8,  lsr #1   \n"
+             "str     %2,  [%11, #12]         \n"
+             "str     %5,  [%11, #28]         \n"
+             : "=&r"(v0), "=&r"(v1), "=&r"(v2),
                "=&r"(v3), "=&r"(v4), "=&r"(v5),
                "+&r"(code1), "+&r"(code2)
-             : "r"(levels - 1), "r"(-levels), "r"(ff_inverse[levels]));
+             : "r"(levels - 1), "r"(-levels),
+               "r"(ff_inverse[levels]), "r"(values)
+             : "memory");
 
     return code1 | code2;
 }