Update for 2.2.3

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

RELEASE

@@ -1 +1 @@
-2.2.2
+2.2.3

VERSION

@@ -1 +1 @@
-2.2.2
+2.2.3

compat/avisynth/avisynth_c.h

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

compat/avisynth/avxsynth_c.h

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

configure

@@ -3386,7 +3386,7 @@ case "$arch" in
     tilegx|tile-gx)
         arch="tilegx"
     ;;
-    i[3-6]86|i86pc|BePC|x86pc|x86_64|x86_32|amd64)
+    i[3-6]86*|i86pc|BePC|x86pc|x86_64|x86_32|amd64)
         arch="x86"
     ;;
 esac

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         = 2.2.2
+PROJECT_NUMBER         = 2.2.3
 
 # With the PROJECT_LOGO tag one can specify a logo or icon that is included
 # in the documentation. The maximum height of the logo should not exceed 55

doc/demuxers.texi

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

doc/encoders.texi

@@ -807,7 +807,7 @@ while producing the worst quality.
 
 @item reservoir
 Enable use of bit reservoir when set to 1. Default value is 1. LAME
-has this enabled by default, but can be overriden by use
+has this enabled by default, but can be overridden by use
 @option{--nores} option.
 
 @item joint_stereo (@emph{-m j})
@@ -1525,7 +1525,7 @@ for detail retention (adaptive quantization, psy-RD, psy-trellis).
 Many libx264 encoder options are mapped to FFmpeg global codec
 options, while unique encoder options are provided through private
 options. Additionally the @option{x264opts} and @option{x264-params}
-private options allows to pass a list of key=value tuples as accepted
+private options allows one to pass a list of key=value tuples as accepted
 by the libx264 @code{x264_param_parse} function.
 
 The x264 project website is at
@@ -1853,7 +1853,7 @@ Override the x264 configuration using a :-separated list of key=value
 parameters.
 
 This option is functionally the same as the @option{x264opts}, but is
-duplicated for compability with the Libav fork.
+duplicated for compatibility with the Libav fork.
 
 For example to specify libx264 encoding options with @command{ffmpeg}:
 @example
@@ -2047,7 +2047,7 @@ Set physical density of pixels, in dots per meter, unset by default
 Apple ProRes encoder.
 
 FFmpeg contains 2 ProRes encoders, the prores-aw and prores-ks encoder.
-The used encoder can be choosen with the @code{-vcodec} option.
+The used encoder can be chosen with the @code{-vcodec} option.
 
 @subsection Private Options for prores-ks
 

doc/ffmpeg-resampler.texi

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

doc/ffmpeg-scaler.texi

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

doc/filters.texi

@@ -172,7 +172,7 @@ terminated when the next special character (belonging to the set
 
 The name and arguments of the filter are optionally preceded and
 followed by a list of link labels.
-A link label allows to name a link and associate it to a filter output
+A link label allows one to name a link and associate it to a filter output
 or input pad. The preceding labels @var{in_link_1}
 ... @var{in_link_N}, are associated to the filter input pads,
 the following labels @var{out_link_1} ... @var{out_link_M}, are
@@ -3921,7 +3921,7 @@ The high threshold selects the "strong" edge pixels, which are then
 connected through 8-connectivity with the "weak" edge pixels selected
 by the low threshold.
 
-@var{low} and @var{high} threshold values must be choosen in the range
+@var{low} and @var{high} threshold values must be chosen in the range
 [0,1], and @var{low} should be lesser or equal to @var{high}.
 
 Default value for @var{low} is @code{20/255}, and default value for @var{high}
@@ -5238,7 +5238,7 @@ Set progressive threshold.
 
 Deinterleave or interleave fields.
 
-This filter allows to process interlaced images fields without
+This filter allows one to process interlaced images fields without
 deinterlacing them. Deinterleaving splits the input frame into 2
 fields (so called half pictures). Odd lines are moved to the top
 half of the output image, even lines to the bottom half.
@@ -6860,7 +6860,7 @@ rotate=A*sin(2*PI/T*t)
 @end example
 
 @item
-Rotate the video, output size is choosen so that the whole rotating
+Rotate the video, output size is chosen so that the whole rotating
 input video is always completely contained in the output:
 @example
 rotate='2*PI*t:ow=hypot(iw,ih):oh=ow'
@@ -6983,7 +6983,7 @@ Default value is @samp{0}.
 @item flags
 Set libswscale scaling flags. See
 @ref{sws_flags,,the ffmpeg-scaler manual,ffmpeg-scaler} for the
-complete list of values. If not explictly specified the filter applies
+complete list of values. If not explicitly specified the filter applies
 the default flags.
 
 @item size, s
@@ -7797,7 +7797,7 @@ Produce 8x8 PNG tiles of all keyframes (@option{-skip_frame nokey}) in a movie:
 ffmpeg -skip_frame nokey -i file.avi -vf 'scale=128:72,tile=8x8' -an -vsync 0 keyframes%03d.png
 @end example
 The @option{-vsync 0} is necessary to prevent @command{ffmpeg} from
-duplicating each output frame to accomodate the originally detected frame
+duplicating each output frame to accommodate the originally detected frame
 rate.
 
 @item
@@ -8394,7 +8394,7 @@ Set dithering to reduce the circular banding effects. Default is @code{1}
 (enabled).
 
 @item aspect
-Set vignette aspect. This setting allows to adjust the shape of the vignette.
+Set vignette aspect. This setting allows one to adjust the shape of the vignette.
 Setting this value to the SAR of the input will make a rectangular vignetting
 following the dimensions of the video.
 
@@ -8903,7 +8903,7 @@ horizontally, vertically, or diagonally adjacent.
 
 At each interaction the grid evolves according to the adopted rule,
 which specifies the number of neighbor alive cells which will make a
-cell stay alive or born. The @option{rule} option allows to specify
+cell stay alive or born. The @option{rule} option allows one to specify
 the rule to adopt.
 
 This source accepts the following options:
@@ -10323,7 +10323,7 @@ Note that when the movie is looped the source timestamps are not
 changed, so it will generate non monotonically increasing timestamps.
 @end table
 
-This filter allows to overlay a second video on top of main input of
+This filter allows one to overlay a second video on top of main input of
 a filtergraph as shown in this graph:
 @example
 input -----------> deltapts0 --> overlay --> output

doc/git-howto.texi

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

doc/indevs.texi

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

doc/muxers.texi

@@ -758,7 +758,7 @@ The segment muxer supports the following options:
 @table @option
 @item reference_stream @var{specifier}
 Set the reference stream, as specified by the string @var{specifier}.
-If @var{specifier} is set to @code{auto}, the reference is choosen
+If @var{specifier} is set to @code{auto}, the reference is chosen
 automatically. Otherwise it must be a stream specifier (see the ``Stream
 specifiers'' chapter in the ffmpeg manual) which specifies the
 reference stream. The default value is @code{auto}.

doc/outdevs.texi

@@ -42,7 +42,7 @@ ffmpeg -i INPUT -f alsa hw:1,7
 
 CACA output device.
 
-This output device allows to show a video stream in CACA window.
+This output device allows one to show a video stream in CACA window.
 Only one CACA window is allowed per application, so you can
 have only one instance of this output device in an application.
 
@@ -302,7 +302,7 @@ ffmpeg  -i INPUT -f pulse "stream name"
 
 SDL (Simple DirectMedia Layer) output device.
 
-This output device allows to show a video stream in an SDL
+This output device allows one to show a video stream in an SDL
 window. Only one SDL window is allowed per application, so you can
 have only one instance of this output device in an application.
 
@@ -361,7 +361,7 @@ sndio audio output device.
 
 XV (XVideo) output device.
 
-This output device allows to show a video stream in a X Window System
+This output device allows one to show a video stream in a X Window System
 window.
 
 @subsection Options

doc/protocols.texi

@@ -960,7 +960,7 @@ used as master salt.
 
 @section tcp
 
-Trasmission Control Protocol.
+Transmission Control Protocol.
 
 The required syntax for a TCP url is:
 @example
@@ -1066,7 +1066,7 @@ udp://@var{hostname}:@var{port}[?@var{options}]
 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
 
 In case threading is enabled on the system, a circular buffer is used
-to store the incoming data, which allows to reduce loss of data due to
+to store the incoming data, which allows one to reduce loss of data due to
 UDP socket buffer overruns. The @var{fifo_size} and
 @var{overrun_nonfatal} options are related to this buffer.
 

doc/utils.texi

@@ -1056,7 +1056,7 @@ which can be obtained with @code{ffmpeg -opencl_bench} or @code{av_opencl_get_de
 @item device_idx
 Select the index of the device used to run OpenCL code.
 
-The specifed index must be one of the indexes in the device list which
+The specified index must be one of the indexes in the device list which
 can be obtained with @code{ffmpeg -opencl_bench} or @code{av_opencl_get_device_list()}.
 
 @end table

ffprobe.c

@@ -191,6 +191,7 @@ static const char unit_hertz_str[]          = "Hz"   ;
 static const char unit_byte_str[]           = "byte" ;
 static const char unit_bit_per_second_str[] = "bit/s";
 
+static int nb_streams;
 static uint64_t *nb_streams_packets;
 static uint64_t *nb_streams_frames;
 static int *selected_streams;
@@ -1632,6 +1633,14 @@ static void writer_register_all(void)
 #define print_section_header(s) writer_print_section_header(w, s)
 #define print_section_footer(s) writer_print_section_footer(w, s)
 
+#define REALLOCZ_ARRAY_STREAM(ptr, cur_n, new_n)                        \
+{                                                                       \
+    ret = av_reallocp_array(&(ptr), (new_n), sizeof(*(ptr)));           \
+    if (ret < 0)                                                        \
+        goto end;                                                       \
+    memset( (ptr) + (cur_n), 0, ((new_n) - (cur_n)) * sizeof(*(ptr)) ); \
+}
+
 static inline int show_tags(WriterContext *w, AVDictionary *tags, int section_id)
 {
     AVDictionaryEntry *tag = NULL;
@@ -1893,6 +1902,12 @@ static int read_interval_packets(WriterContext *w, AVFormatContext *fmt_ctx,
         goto end;
     }
     while (!av_read_frame(fmt_ctx, &pkt)) {
+        if (fmt_ctx->nb_streams > nb_streams) {
+            REALLOCZ_ARRAY_STREAM(nb_streams_frames,  nb_streams, fmt_ctx->nb_streams);
+            REALLOCZ_ARRAY_STREAM(nb_streams_packets, nb_streams, fmt_ctx->nb_streams);
+            REALLOCZ_ARRAY_STREAM(selected_streams,   nb_streams, fmt_ctx->nb_streams);
+            nb_streams = fmt_ctx->nb_streams;
+        }
         if (selected_streams[pkt.stream_index]) {
             AVRational tb = fmt_ctx->streams[pkt.stream_index]->time_base;
 
@@ -2367,9 +2382,10 @@ static int probe_file(WriterContext *wctx, const char *filename)
 
 #define CHECK_END if (ret < 0) goto end
 
-    nb_streams_frames  = av_calloc(fmt_ctx->nb_streams, sizeof(*nb_streams_frames));
-    nb_streams_packets = av_calloc(fmt_ctx->nb_streams, sizeof(*nb_streams_packets));
-    selected_streams   = av_calloc(fmt_ctx->nb_streams, sizeof(*selected_streams));
+    nb_streams = fmt_ctx->nb_streams;
+    REALLOCZ_ARRAY_STREAM(nb_streams_frames,0,fmt_ctx->nb_streams);
+    REALLOCZ_ARRAY_STREAM(nb_streams_packets,0,fmt_ctx->nb_streams);
+    REALLOCZ_ARRAY_STREAM(selected_streams,0,fmt_ctx->nb_streams);
 
     for (i = 0; i < fmt_ctx->nb_streams; i++) {
         if (stream_specifier) {

libavcodec/aac.h

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

libavcodec/dcadec.c

@@ -2173,7 +2173,7 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
             if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO ||
                                     s->core_downmix_amode == DCA_STEREO_TOTAL)) {
                 int sign, code;
-                for (i = 0; i < s->prim_channels + !!s->lfe; i++) {
+                for (i = 0; i < num_core_channels + !!s->lfe; i++) {
                     sign = s->core_downmix_codes[i][0] & 0x100 ? 1 : -1;
                     code = s->core_downmix_codes[i][0] & 0x0FF;
                     s->downmix_coef[i][0] = (!code ? 0.0f :
@@ -2191,19 +2191,19 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
                            "Invalid channel mode %d\n", am);
                     return AVERROR_INVALIDDATA;
                 }
-                if (s->prim_channels + !!s->lfe >
+                if (num_core_channels + !!s->lfe >
                     FF_ARRAY_ELEMS(dca_default_coeffs[0])) {
                     avpriv_request_sample(s->avctx, "Downmixing %d channels",
                                           s->prim_channels + !!s->lfe);
                     return AVERROR_PATCHWELCOME;
                 }
-                for (i = 0; i < s->prim_channels + !!s->lfe; i++) {
+                for (i = 0; i < num_core_channels + !!s->lfe; i++) {
                     s->downmix_coef[i][0] = dca_default_coeffs[am][i][0];
                     s->downmix_coef[i][1] = dca_default_coeffs[am][i][1];
                 }
             }
             av_dlog(s->avctx, "Stereo downmix coeffs:\n");
-            for (i = 0; i < s->prim_channels + !!s->lfe; i++) {
+            for (i = 0; i < num_core_channels + !!s->lfe; i++) {
                 av_dlog(s->avctx, "L, input channel %d = %f\n", i,
                         s->downmix_coef[i][0]);
                 av_dlog(s->avctx, "R, input channel %d = %f\n", i,
@@ -2329,6 +2329,17 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
     { /* xxx should also do MA extensions */
         if (s->amode < 16) {
             avctx->channel_layout = dca_core_channel_layout[s->amode];
+
+            if (s->prim_channels + !!s->lfe > 2 &&
+                avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
+                /*
+                 * Neither the core's auxiliary data nor our default tables contain
+                 * downmix coefficients for the additional channel coded in the XCh
+                 * extension, so when we're doing a Stereo downmix, don't decode it.
+                 */
+                s->xch_disable = 1;
+            }
+
 #if FF_API_REQUEST_CHANNELS
 FF_DISABLE_DEPRECATION_WARNINGS
             if (s->xch_present && !s->xch_disable &&
@@ -2366,7 +2377,7 @@ FF_ENABLE_DEPRECATION_WARNINGS
                 return AVERROR_INVALIDDATA;
             }
 
-            if (s->prim_channels + !!s->lfe > 2 &&
+            if (num_core_channels + !!s->lfe > 2 &&
                 avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
                 channels = 2;
                 s->output = s->prim_channels == 2 ? s->amode : DCA_STEREO;
@@ -2415,7 +2426,7 @@ FF_ENABLE_DEPRECATION_WARNINGS
          * masks in some sense -- unfortunately some channels could overlap */
         if (av_popcount(channel_mask) != av_popcount(channel_layout)) {
             av_log(avctx, AV_LOG_DEBUG,
-                   "DTS-XXCH: Inconsistant avcodec/dts channel layouts\n");
+                   "DTS-XXCH: Inconsistent avcodec/dts channel layouts\n");
             return AVERROR_INVALIDDATA;
         }
 
@@ -2433,6 +2444,7 @@ FF_ENABLE_DEPRECATION_WARNINGS
                         s->xxch_order_tab[j++] = posn;
                     }
                 }
+
             }
 
             s->lfe_index = av_popcount(channel_layout & (AV_CH_LOW_FREQUENCY-1));

libavcodec/diracdec.c

@@ -201,6 +201,7 @@ typedef struct DiracContext {
 
     uint16_t *mctmp;            /* buffer holding the MC data multipled by OBMC weights */
     uint8_t *mcscratch;
+    int buffer_stride;
 
     DECLARE_ALIGNED(16, uint8_t, obmc_weight)[3][MAX_BLOCKSIZE*MAX_BLOCKSIZE];
 
@@ -343,22 +344,44 @@ static int alloc_sequence_buffers(DiracContext *s)
             return AVERROR(ENOMEM);
     }
 
-    w = s->source.width;
-    h = s->source.height;
-
     /* fixme: allocate using real stride here */
-    s->sbsplit  = av_malloc(sbwidth * sbheight);
-    s->blmotion = av_malloc(sbwidth * sbheight * 16 * sizeof(*s->blmotion));
-    s->edge_emu_buffer_base = av_malloc((w+64)*MAX_BLOCKSIZE);
+    s->sbsplit  = av_malloc_array(sbwidth, sbheight);
+    s->blmotion = av_malloc_array(sbwidth, sbheight * 16 * sizeof(*s->blmotion));
 
-    s->mctmp     = av_malloc((w+64+MAX_BLOCKSIZE) * (h+MAX_BLOCKSIZE) * sizeof(*s->mctmp));
-    s->mcscratch = av_malloc((w+64)*MAX_BLOCKSIZE);
-
-    if (!s->sbsplit || !s->blmotion || !s->mctmp || !s->mcscratch)
+    if (!s->sbsplit || !s->blmotion)
         return AVERROR(ENOMEM);
     return 0;
 }
 
+static int alloc_buffers(DiracContext *s, int stride)
+{
+    int w = s->source.width;
+    int h = s->source.height;
+
+    av_assert0(stride >= w);
+    stride += 64;
+
+    if (s->buffer_stride >= stride)
+        return 0;
+    s->buffer_stride = 0;
+
+    av_freep(&s->edge_emu_buffer_base);
+    memset(s->edge_emu_buffer, 0, sizeof(s->edge_emu_buffer));
+    av_freep(&s->mctmp);
+    av_freep(&s->mcscratch);
+
+    s->edge_emu_buffer_base = av_malloc_array(stride, MAX_BLOCKSIZE);
+
+    s->mctmp     = av_malloc_array((stride+MAX_BLOCKSIZE), (h+MAX_BLOCKSIZE) * sizeof(*s->mctmp));
+    s->mcscratch = av_malloc_array(stride, MAX_BLOCKSIZE);
+
+    if (!s->edge_emu_buffer_base || !s->mctmp || !s->mcscratch)
+        return AVERROR(ENOMEM);
+
+    s->buffer_stride = stride;
+    return 0;
+}
+
 static void free_sequence_buffers(DiracContext *s)
 {
     int i, j, k;
@@ -382,6 +405,7 @@ static void free_sequence_buffers(DiracContext *s)
         av_freep(&s->plane[i].idwt_tmp);
     }
 
+    s->buffer_stride = 0;
     av_freep(&s->sbsplit);
     av_freep(&s->blmotion);
     av_freep(&s->edge_emu_buffer_base);
@@ -1854,6 +1878,9 @@ static int dirac_decode_data_unit(AVCodecContext *avctx, const uint8_t *buf, int
         s->plane[1].stride = pic->avframe->linesize[1];
         s->plane[2].stride = pic->avframe->linesize[2];
 
+        if (alloc_buffers(s, FFMAX3(FFABS(s->plane[0].stride), FFABS(s->plane[1].stride), FFABS(s->plane[2].stride))) < 0)
+            return AVERROR(ENOMEM);
+
         /* [DIRAC_STD] 11.1 Picture parse. picture_parse() */
         if (dirac_decode_picture_header(s))
             return -1;

libavcodec/golomb-test.c

@@ -58,7 +58,7 @@ int main(void)
         }
     }
 
-#define EXTEND(i) (i << 3 | i & 7)
+#define EXTEND(i) ((i) << 3 | (i) & 7)
     init_put_bits(&pb, temp, SIZE);
     for (i = 0; i < COUNT; i++)
         set_ue_golomb(&pb, EXTEND(i));

libavcodec/h264.h

@@ -61,10 +61,10 @@
 #define MAX_SLICES 16
 
 #ifdef ALLOW_INTERLACE
-#define MB_MBAFF(h)    h->mb_mbaff
-#define MB_FIELD(h)    h->mb_field_decoding_flag
-#define FRAME_MBAFF(h) h->mb_aff_frame
-#define FIELD_PICTURE(h) (h->picture_structure != PICT_FRAME)
+#define MB_MBAFF(h)    (h)->mb_mbaff
+#define MB_FIELD(h)    (h)->mb_field_decoding_flag
+#define FRAME_MBAFF(h) (h)->mb_aff_frame
+#define FIELD_PICTURE(h) ((h)->picture_structure != PICT_FRAME)
 #define LEFT_MBS 2
 #define LTOP     0
 #define LBOT     1
@@ -84,12 +84,12 @@
 #define FIELD_OR_MBAFF_PICTURE(h) (FRAME_MBAFF(h) || FIELD_PICTURE(h))
 
 #ifndef CABAC
-#define CABAC(h) h->pps.cabac
+#define CABAC(h) (h)->pps.cabac
 #endif
 
-#define CHROMA(h)    (h->sps.chroma_format_idc)
-#define CHROMA422(h) (h->sps.chroma_format_idc == 2)
-#define CHROMA444(h) (h->sps.chroma_format_idc == 3)
+#define CHROMA(h)    ((h)->sps.chroma_format_idc)
+#define CHROMA422(h) ((h)->sps.chroma_format_idc == 2)
+#define CHROMA444(h) ((h)->sps.chroma_format_idc == 3)
 
 #define EXTENDED_SAR       255
 

libavcodec/hevc.h

@@ -81,10 +81,10 @@
 #define SAMPLE_CTB(tab, x, y) ((tab)[(y) * min_cb_width + (x)])
 #define SAMPLE_CBF(tab, x, y) ((tab)[((y) & ((1<<log2_trafo_size)-1)) * MAX_CU_SIZE + ((x) & ((1<<log2_trafo_size)-1))])
 
-#define IS_IDR(s) (s->nal_unit_type == NAL_IDR_W_RADL || s->nal_unit_type == NAL_IDR_N_LP)
-#define IS_BLA(s) (s->nal_unit_type == NAL_BLA_W_RADL || s->nal_unit_type == NAL_BLA_W_LP || \
-                   s->nal_unit_type == NAL_BLA_N_LP)
-#define IS_IRAP(s) (s->nal_unit_type >= 16 && s->nal_unit_type <= 23)
+#define IS_IDR(s) ((s)->nal_unit_type == NAL_IDR_W_RADL || (s)->nal_unit_type == NAL_IDR_N_LP)
+#define IS_BLA(s) ((s)->nal_unit_type == NAL_BLA_W_RADL || (s)->nal_unit_type == NAL_BLA_W_LP || \
+                   (s)->nal_unit_type == NAL_BLA_N_LP)
+#define IS_IRAP(s) ((s)->nal_unit_type >= 16 && (s)->nal_unit_type <= 23)
 
 /**
  * Table 7-3: NAL unit type codes

libavcodec/ivi_dsp.c

@@ -235,15 +235,15 @@ void ff_ivi_recompose_haar(const IVIPlaneDesc *plane, uint8_t *dst,
 
 /** butterfly operation for the inverse Haar transform */
 #define IVI_HAAR_BFLY(s1, s2, o1, o2, t) \
-    t  = (s1 - s2) >> 1;\
-    o1 = (s1 + s2) >> 1;\
-    o2 = t;\
+    t  = ((s1) - (s2)) >> 1;\
+    o1 = ((s1) + (s2)) >> 1;\
+    o2 = (t);\
 
 /** inverse 8-point Haar transform */
 #define INV_HAAR8(s1, s5, s3, s7, s2, s4, s6, s8,\
                   d1, d2, d3, d4, d5, d6, d7, d8,\
                   t0, t1, t2, t3, t4, t5, t6, t7, t8) {\
-    t1 = s1 << 1; t5 = s5 << 1;\
+    t1 = (s1) << 1; t5 = (s5) << 1;\
     IVI_HAAR_BFLY(t1, t5, t1, t5, t0); IVI_HAAR_BFLY(t1, s3, t1, t3, t0);\
     IVI_HAAR_BFLY(t5, s7, t5, t7, t0); IVI_HAAR_BFLY(t1, s2, t1, t2, t0);\
     IVI_HAAR_BFLY(t3, s4, t3, t4, t0); IVI_HAAR_BFLY(t5, s6, t5, t6, t0);\
@@ -485,21 +485,21 @@ void ff_ivi_dc_haar_2d(const int32_t *in, int16_t *out, uint32_t pitch,
 
 /** butterfly operation for the inverse slant transform */
 #define IVI_SLANT_BFLY(s1, s2, o1, o2, t) \
-    t  = s1 - s2;\
-    o1 = s1 + s2;\
-    o2 = t;\
+    t  = (s1) - (s2);\
+    o1 = (s1) + (s2);\
+    o2 = (t);\
 
 /** This is a reflection a,b = 1/2, 5/4 for the inverse slant transform */
 #define IVI_IREFLECT(s1, s2, o1, o2, t) \
-    t  = ((s1 + s2*2 + 2) >> 2) + s1;\
-    o2 = ((s1*2 - s2 + 2) >> 2) - s2;\
-    o1 = t;\
+    t  = (((s1) + (s2)*2 + 2) >> 2) + (s1);\
+    o2 = (((s1)*2 - (s2) + 2) >> 2) - (s2);\
+    o1 = (t);\
 
 /** This is a reflection a,b = 1/2, 7/8 for the inverse slant transform */
 #define IVI_SLANT_PART4(s1, s2, o1, o2, t) \
-    t  = s2 + ((s1*4  - s2 + 4) >> 3);\
-    o2 = s1 + ((-s1 - s2*4 + 4) >> 3);\
-    o1 = t;\
+    t  = (s2) + (((s1)*4  - (s2) + 4) >> 3);\
+    o2 = (s1) + ((-(s1) - (s2)*4 + 4) >> 3);\
+    o1 = (t);\
 
 /** inverse slant8 transform */
 #define IVI_INV_SLANT8(s1, s4, s8, s5, s2, s6, s3, s7,\
@@ -557,7 +557,7 @@ void ff_ivi_inverse_slant_8x8(const int32_t *in, int16_t *out, uint32_t pitch, c
     }
 #undef COMPENSATE
 
-#define COMPENSATE(x) ((x + 1)>>1)
+#define COMPENSATE(x) (((x) + 1)>>1)
     src = tmp;
     for (i = 0; i < 8; i++) {
         if (!src[0] && !src[1] && !src[2] && !src[3] && !src[4] && !src[5] && !src[6] && !src[7]) {
@@ -597,7 +597,7 @@ void ff_ivi_inverse_slant_4x4(const int32_t *in, int16_t *out, uint32_t pitch, c
     }
 #undef COMPENSATE
 
-#define COMPENSATE(x) ((x + 1)>>1)
+#define COMPENSATE(x) (((x) + 1)>>1)
     src = tmp;
     for (i = 0; i < 4; i++) {
         if (!src[0] && !src[1] && !src[2] && !src[3]) {
@@ -631,7 +631,7 @@ void ff_ivi_row_slant8(const int32_t *in, int16_t *out, uint32_t pitch, const ui
     int     i;
     int     t0, t1, t2, t3, t4, t5, t6, t7, t8;
 
-#define COMPENSATE(x) ((x + 1)>>1)
+#define COMPENSATE(x) (((x) + 1)>>1)
     for (i = 0; i < 8; i++) {
         if (!in[0] && !in[1] && !in[2] && !in[3] && !in[4] && !in[5] && !in[6] && !in[7]) {
             memset(out, 0, 8*sizeof(out[0]));
@@ -673,7 +673,7 @@ void ff_ivi_col_slant8(const int32_t *in, int16_t *out, uint32_t pitch, const ui
     row4 = pitch << 2;
     row8 = pitch << 3;
 
-#define COMPENSATE(x) ((x + 1)>>1)
+#define COMPENSATE(x) (((x) + 1)>>1)
     for (i = 0; i < 8; i++) {
         if (flags[i]) {
             IVI_INV_SLANT8(in[0], in[8], in[16], in[24], in[32], in[40], in[48], in[56],
@@ -710,7 +710,7 @@ void ff_ivi_row_slant4(const int32_t *in, int16_t *out, uint32_t pitch, const ui
     int     i;
     int     t0, t1, t2, t3, t4;
 
-#define COMPENSATE(x) ((x + 1)>>1)
+#define COMPENSATE(x) (((x) + 1)>>1)
     for (i = 0; i < 4; i++) {
         if (!in[0] && !in[1] && !in[2] && !in[3]) {
             memset(out, 0, 4*sizeof(out[0]));
@@ -732,7 +732,7 @@ void ff_ivi_col_slant4(const int32_t *in, int16_t *out, uint32_t pitch, const ui
 
     row2 = pitch << 1;
 
-#define COMPENSATE(x) ((x + 1)>>1)
+#define COMPENSATE(x) (((x) + 1)>>1)
     for (i = 0; i < 4; i++) {
         if (flags[i]) {
             IVI_INV_SLANT4(in[0], in[4], in[8], in[12],

libavcodec/libvorbisenc.c

@@ -349,7 +349,8 @@ static int libvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
             avctx->delay              = duration;
             av_assert0(!s->afq.remaining_delay);
             s->afq.frames->duration  += duration;
-            s->afq.frames->pts       -= duration;
+            if (s->afq.frames->pts != AV_NOPTS_VALUE)
+                s->afq.frames->pts       -= duration;
             s->afq.remaining_samples += duration;
         }
         ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration);

libavcodec/mlpdec.c

@@ -872,7 +872,7 @@ static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
     return 0;
 }
 
-#define MSB_MASK(bits)  (-1u << bits)
+#define MSB_MASK(bits)  (-1u << (bits))
 
 /** Generate PCM samples using the prediction filters and residual values
  *  read from the data stream, and update the filter state. */

libavcodec/mss34dsp.c

@@ -84,8 +84,8 @@ void ff_mss34_gen_quant_mat(uint16_t *qmat, int quality, int luma)
     blk[6 * step] = (-(t3 + t7) + t8 + tA) >> shift;                \
     blk[7 * step] = (-(t1 + t6) + t9 + tB) >> shift;                \
 
-#define SOP_ROW(a) ((a) << 16) + 0x2000
-#define SOP_COL(a) ((a + 32) << 16)
+#define SOP_ROW(a) (((a) << 16) + 0x2000)
+#define SOP_COL(a) (((a) + 32) << 16)
 
 void ff_mss34_dct_put(uint8_t *dst, int stride, int *block)
 {

libavcodec/mss4.c

@@ -364,7 +364,7 @@ static int get_value_cached(GetBitContext *gb, int vec_pos, uint8_t *vec,
     return prev[component];
 }
 
-#define MKVAL(vals)  (vals[0] | (vals[1] << 3) | (vals[2] << 6))
+#define MKVAL(vals)  ((vals)[0] | ((vals)[1] << 3) | ((vals)[2] << 6))
 
 /* Image mode - the hardest to comprehend MSS4 coding mode.
  *

libavcodec/ppc/asm.S

@@ -36,12 +36,20 @@
 
 .macro extfunc name
     .global X(\name)
+#if _CALL_ELF == 2
+    .text
+X(\name):
+    addis %r2, %r12, .TOC.-X(\name)@ha
+    addi %r2, %r2, .TOC.-X(\name)@l
+    .localentry X(\name), .-X(\name)
+#else
     .section .opd, "aw"
 X(\name):
     .quad L(\name), .TOC.@tocbase, 0
     .previous
     .type X(\name), STT_FUNC
 L(\name):
+#endif
 .endm
 
 .macro movrel rd, sym, gp

libavcodec/svq3.c

@@ -1187,7 +1187,7 @@ static int svq3_decode_frame(AVCodecContext *avctx, void *data,
     if (h->pict_type != AV_PICTURE_TYPE_I) {
         if (!s->last_pic->f.data[0]) {
             av_log(avctx, AV_LOG_ERROR, "Missing reference frame.\n");
-            av_frame_unref(s->last_pic);
+            av_frame_unref(&s->last_pic->f);
             ret = get_buffer(avctx, s->last_pic);
             if (ret < 0)
                 return ret;
@@ -1200,7 +1200,7 @@ static int svq3_decode_frame(AVCodecContext *avctx, void *data,
 
         if (h->pict_type == AV_PICTURE_TYPE_B && !s->next_pic->f.data[0]) {
             av_log(avctx, AV_LOG_ERROR, "Missing reference frame.\n");
-            av_frame_unref(s->next_pic);
+            av_frame_unref(&s->next_pic->f);
             ret = get_buffer(avctx, s->next_pic);
             if (ret < 0)
                 return ret;

libavfilter/af_volume.c

@@ -389,6 +389,8 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
         }
     }
 
+    emms_c();
+
     if (buf != out_buf)
         av_frame_free(&buf);
 

libavfilter/vf_hqdn3d.c

@@ -147,6 +147,7 @@ static void denoise_depth(HQDN3DContext *s,
     else
         denoise_temporal(src, dst, frame_ant,
                          w, h, sstride, dstride, temporal, depth);
+    emms_c();
 }
 
 #define denoise(...) \

libavformat/avidec.c

@@ -864,7 +864,11 @@ fail:
     if (!avi->index_loaded && pb->seekable)
         avi_load_index(s);
     avi->index_loaded    |= 1;
-    avi->non_interleaved |= guess_ni_flag(s) | (s->flags & AVFMT_FLAG_SORT_DTS);
+
+    if ((ret = guess_ni_flag(s)) < 0)
+        return ret;
+
+    avi->non_interleaved |= ret | (s->flags & AVFMT_FLAG_SORT_DTS);
 
     dict_entry = av_dict_get(s->metadata, "ISFT", NULL, 0);
     if (dict_entry && !strcmp(dict_entry->value, "PotEncoder"))
@@ -1455,14 +1459,69 @@ static int avi_read_idx1(AVFormatContext *s, int size)
     return 0;
 }
 
+/* Scan the index and consider any file with streams more than
+ * 2 seconds or 64MB apart non-interleaved. */
+static int check_stream_max_drift(AVFormatContext *s)
+{
+    int64_t min_pos, pos;
+    int i;
+    int *idx = av_mallocz_array(s->nb_streams, sizeof(*idx));
+    if (!idx)
+        return AVERROR(ENOMEM);
+    for (min_pos = pos = 0; min_pos != INT64_MAX; pos = min_pos + 1LU) {
+        int64_t max_dts = INT64_MIN / 2;
+        int64_t min_dts = INT64_MAX / 2;
+        int64_t max_buffer = 0;
+
+        min_pos = INT64_MAX;
+
+        for (i = 0; i < s->nb_streams; i++) {
+            AVStream *st = s->streams[i];
+            AVIStream *ast = st->priv_data;
+            int n = st->nb_index_entries;
+            while (idx[i] < n && st->index_entries[idx[i]].pos < pos)
+                idx[i]++;
+            if (idx[i] < n) {
+                int64_t dts;
+                dts = av_rescale_q(st->index_entries[idx[i]].timestamp /
+                                   FFMAX(ast->sample_size, 1),
+                                   st->time_base, AV_TIME_BASE_Q);
+                min_dts = FFMIN(min_dts, dts);
+                min_pos = FFMIN(min_pos, st->index_entries[idx[i]].pos);
+            }
+        }
+        for (i = 0; i < s->nb_streams; i++) {
+            AVStream *st = s->streams[i];
+            AVIStream *ast = st->priv_data;
+
+            if (idx[i] && min_dts != INT64_MAX / 2) {
+                int64_t dts;
+                dts = av_rescale_q(st->index_entries[idx[i] - 1].timestamp /
+                                   FFMAX(ast->sample_size, 1),
+                                   st->time_base, AV_TIME_BASE_Q);
+                max_dts = FFMAX(max_dts, dts);
+                max_buffer = FFMAX(max_buffer,
+                                   av_rescale(dts - min_dts,
+                                              st->codec->bit_rate,
+                                              AV_TIME_BASE));
+            }
+        }
+        if (max_dts - min_dts > 2 * AV_TIME_BASE ||
+            max_buffer > 1024 * 1024 * 8 * 8) {
+            av_free(idx);
+            return 1;
+        }
+    }
+    av_free(idx);
+    return 0;
+}
+
 static int guess_ni_flag(AVFormatContext *s)
 {
     int i;
     int64_t last_start = 0;
     int64_t first_end  = INT64_MAX;
     int64_t oldpos     = avio_tell(s->pb);
-    int *idx;
-    int64_t min_pos, pos;
 
     for (i = 0; i < s->nb_streams; i++) {
         AVStream *st = s->streams[i];
@@ -1486,35 +1545,11 @@ static int guess_ni_flag(AVFormatContext *s)
             first_end = st->index_entries[n - 1].pos;
     }
     avio_seek(s->pb, oldpos, SEEK_SET);
+
     if (last_start > first_end)
         return 1;
-    idx= av_calloc(s->nb_streams, sizeof(*idx));
-    if (!idx)
-        return 0;
-    for (min_pos=pos=0; min_pos!=INT64_MAX; pos= min_pos+1LU) {
-        int64_t max_dts = INT64_MIN/2, min_dts= INT64_MAX/2;
-        min_pos = INT64_MAX;
 
-        for (i=0; i<s->nb_streams; i++) {
-            AVStream *st = s->streams[i];
-            AVIStream *ast = st->priv_data;
-            int n= st->nb_index_entries;
-            while (idx[i]<n && st->index_entries[idx[i]].pos < pos)
-                idx[i]++;
-            if (idx[i] < n) {
-                min_dts = FFMIN(min_dts, av_rescale_q(st->index_entries[idx[i]].timestamp/FFMAX(ast->sample_size, 1), st->time_base, AV_TIME_BASE_Q));
-                min_pos = FFMIN(min_pos, st->index_entries[idx[i]].pos);
-            }
-            if (idx[i])
-                max_dts = FFMAX(max_dts, av_rescale_q(st->index_entries[idx[i]-1].timestamp/FFMAX(ast->sample_size, 1), st->time_base, AV_TIME_BASE_Q));
-        }
-        if (max_dts - min_dts > 2*AV_TIME_BASE) {
-            av_free(idx);
-            return 1;
-        }
-    }
-    av_free(idx);
-    return 0;
+    return check_stream_max_drift(s);
 }
 
 static int avi_load_index(AVFormatContext *s)

libavformat/flvenc.c

@@ -536,7 +536,7 @@ static int flv_write_packet(AVFormatContext *s, AVPacket *pkt)
         sc->last_ts = ts;
 
     avio_wb24(pb, size + flags_size);
-    avio_wb24(pb, ts);
+    avio_wb24(pb, ts & 0xFFFFFF);
     avio_w8(pb, (ts >> 24) & 0x7F); // timestamps are 32 bits _signed_
     avio_wb24(pb, flv->reserved);
 

libavformat/h263dec.c

@@ -31,24 +31,37 @@ static int h263_probe(AVProbeData *p)
     int res_change=0;
     int src_fmt, last_src_fmt=-1;
     int last_gn=0;
+    int tr, last_tr = -1;
 
     for(i=0; i<p->buf_size; i++){
         code = (code<<8) + p->buf[i];
-        if ((code & 0xfffffc0000) == 0x800000) {
-            src_fmt= (code>>2)&7;
+        if ((code & 0xfffffc000000) == 0x80000000) {
+            tr = (code >> 18) & 0xFF;
+            src_fmt= (code>>10)&7;
             if(   src_fmt != last_src_fmt
                && last_src_fmt>0 && last_src_fmt<6
                && src_fmt<6)
                 res_change++;
 
-            if((code&0x300)==0x200 && src_fmt){
+            if (tr == last_tr) {
+                invalid_psc++;
+                continue;
+            }
+
+            if (src_fmt != 7 && !(code&(1<<9)) && (code&(1<<5))) {
+                invalid_psc++;
+                continue;
+            }
+
+            if((code&0x30000)==0x20000 && src_fmt){
                 valid_psc++;
                 last_gn=0;
             }else
                 invalid_psc++;
             last_src_fmt= src_fmt;
-        } else if((code & 0xffff800000) == 0x800000) {
-            int gn= (code>>(23-5)) & 0x1F;
+            last_tr = tr;
+        } else if((code & 0xffff80000000) == 0x80000000) {
+            int gn= (code>>(31-5)) & 0x1F;
             if(gn<last_gn){
                 invalid_psc++;
             }else

libavformat/matroskaenc.c

@@ -1088,7 +1088,7 @@ static int mkv_write_header(AVFormatContext *s)
     AVIOContext *pb = s->pb;
     ebml_master ebml_header, segment_info;
     AVDictionaryEntry *tag;
-    int ret, i;
+    int ret, i, version = 2;
 
     if (!strcmp(s->oformat->name, "webm")) mkv->mode = MODE_WEBM;
     else                                   mkv->mode = MODE_MATROSKAv2;
@@ -1096,7 +1096,12 @@ static int mkv_write_header(AVFormatContext *s)
     if (s->avoid_negative_ts < 0)
         s->avoid_negative_ts = 1;
 
-    for (i = 0; i < s->nb_streams; i++)
+    if (mkv->mode != MODE_WEBM ||
+        av_dict_get(s->metadata, "stereo_mode", NULL, 0) ||
+        av_dict_get(s->metadata, "alpha_mode", NULL, 0))
+        version = 4;
+
+    for (i = 0; i < s->nb_streams; i++) {
         if (s->streams[i]->codec->codec_id == AV_CODEC_ID_ATRAC3 ||
             s->streams[i]->codec->codec_id == AV_CODEC_ID_COOK ||
             s->streams[i]->codec->codec_id == AV_CODEC_ID_RA_288 ||
@@ -1108,6 +1113,11 @@ static int mkv_write_header(AVFormatContext *s)
                    avcodec_get_name(s->streams[i]->codec->codec_id));
             return AVERROR_PATCHWELCOME;
         }
+        if (s->streams[i]->codec->codec_id == AV_CODEC_ID_OPUS ||
+            av_dict_get(s->streams[i]->metadata, "stereo_mode", NULL, 0) ||
+            av_dict_get(s->streams[i]->metadata, "alpha_mode", NULL, 0))
+            version = 4;
+    }
 
     mkv->tracks = av_mallocz(s->nb_streams * sizeof(*mkv->tracks));
     if (!mkv->tracks)
@@ -1119,7 +1129,7 @@ static int mkv_write_header(AVFormatContext *s)
     put_ebml_uint   (pb, EBML_ID_EBMLMAXIDLENGTH    ,           4);
     put_ebml_uint   (pb, EBML_ID_EBMLMAXSIZELENGTH  ,           8);
     put_ebml_string (pb, EBML_ID_DOCTYPE            , s->oformat->name);
-    put_ebml_uint   (pb, EBML_ID_DOCTYPEVERSION     ,           4);
+    put_ebml_uint   (pb, EBML_ID_DOCTYPEVERSION     ,     version);
     put_ebml_uint   (pb, EBML_ID_DOCTYPEREADVERSION ,           2);
     end_ebml_master(pb, ebml_header);
 

libavformat/mov.c

@@ -2167,7 +2167,7 @@ static void mov_build_index(MOVContext *mov, AVStream *st)
                 if (sc->keyframe_absent
                     && !sc->stps_count
                     && !rap_group_present
-                    && st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
+                    && (st->codec->codec_type == AVMEDIA_TYPE_AUDIO || (i==0 && j==0)))
                      keyframe = 1;
                 if (keyframe)
                     distance = 0;

libavformat/utils.c

@@ -3234,7 +3234,8 @@ int avformat_find_stream_info(AVFormatContext *ic, AVDictionary **options)
             }
         }
 #if FF_API_R_FRAME_RATE
-        ff_rfps_add_frame(ic, st, pkt->dts);
+        if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO)
+            ff_rfps_add_frame(ic, st, pkt->dts);
 #endif
         if (st->parser && st->parser->parser->split && !st->codec->extradata) {
             int i = st->parser->parser->split(st->codec, pkt->data, pkt->size);

libavutil/aarch64/asm.S

@@ -58,8 +58,8 @@ ELF     .size   \name, . - \name
 
 .macro  movrel rd, val
 #if CONFIG_PIC
-        adrp            \rd, #:pg_hi21:\val
-        add             \rd, \rd, #:lo12:\val
+        adrp            \rd, :pg_hi21:\val
+        add             \rd, \rd, :lo12:\val
 #else
         ldr             \rd, =\val
 #endif

libavutil/cpu.c

@@ -47,6 +47,26 @@
 static int flags, checked;
 
 void av_force_cpu_flags(int arg){
+    if (   (arg & ( AV_CPU_FLAG_3DNOW    |
+                    AV_CPU_FLAG_3DNOWEXT |
+                    AV_CPU_FLAG_SSE      |
+                    AV_CPU_FLAG_SSE2     |
+                    AV_CPU_FLAG_SSE2SLOW |
+                    AV_CPU_FLAG_SSE3     |
+                    AV_CPU_FLAG_SSE3SLOW |
+                    AV_CPU_FLAG_SSSE3    |
+                    AV_CPU_FLAG_SSE4     |
+                    AV_CPU_FLAG_SSE42    |
+                    AV_CPU_FLAG_AVX      |
+                    AV_CPU_FLAG_XOP      |
+                    AV_CPU_FLAG_FMA3     |
+                    AV_CPU_FLAG_FMA4     |
+                    AV_CPU_FLAG_AVX2     ))
+        && !(arg & AV_CPU_FLAG_MMX)) {
+        av_log(NULL, AV_LOG_WARNING, "MMX implied by specified flags\n");
+        arg |= AV_CPU_FLAG_MMX;
+    }
+
     flags   = arg;
     checked = arg != -1;
 }

libswscale/utils.c

@@ -389,7 +389,7 @@ static av_cold int initFilter(int16_t **outFilter, int32_t **filterPos,
 
         xDstInSrc = ((dstPos*(int64_t)xInc)>>7) - ((srcPos*0x10000LL)>>7);
         for (i = 0; i < dstW; i++) {
-            int xx = (xDstInSrc - ((filterSize - 2) << 16)) / (1 << 17);
+            int xx = (xDstInSrc - ((int64_t)(filterSize - 2) << 16)) / (1 << 17);
             int j;
             (*filterPos)[i] = xx;
             for (j = 0; j < filterSize; j++) {