Merge remote-tracking branch 'newdev/master'

* newdev/master:
  mov: set audio service type for AC-3 from bitstream mode in the 'dac3' atom.
  Get audio_service_type for AC-3 based on bitstream mode in the AC-3 parser and decoder, and vice-versa for the AC-3 encoder.
  Use audio_service_type to set stream disposition.
  Add APIchanges entry for audio_service_type.
  Add audio_service_type field to AVCodecContext for encoding and reporting of the service type in the audio bitstream.
  configure: in check_ld, place new -l flags before existing ones
  support @heading, @subheading, @subsubheading, and @subsubsection in texi2pod.pl
  doc: update build system documentation
  aacenc: indentation
  aacenc: fix the side calculation in search_for_ms
  vp8.c: rename EDGE_* to VP8_EDGE_*.

Conflicts:
	doc/APIchanges
	libavcodec/avcodec.h
	libavcodec/version.h
	libavcodec/vp8.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>

configure

@@ -642,7 +642,7 @@ check_ld(){
         test "${f}" = "${f#-l}" && flags="$flags $f" || libs="$libs $f"
     done
     check_cc $($filter_cflags $flags) || return
-    check_cmd $ld $LDFLAGS $flags -o $TMPE $TMPO $extralibs $libs
+    check_cmd $ld $LDFLAGS $flags -o $TMPE $TMPO $libs $extralibs
 }
 
 check_cppflags(){

doc/APIchanges

@@ -12,6 +12,9 @@ libavutil:   2009-03-08
 
 API changes, most recent first:
 
+2011-03-25 - 34b47d7 - lavc 52.115.0 - AVCodecContext.audio_service_type
+  Add audio_service_type field to AVCodecContext.
+
 2011-XX-XX - XXXXXXX - lavu  XX.XXX.X - pixfmt.h
   Add PIX_FMT_BGR48LE and PIX_FMT_BGR48BE pixel formats
 

doc/build_system.txt

 FFmpeg currently uses a custom build system, this text attempts to document
 some of its obscure features and options.
 
-Options to make:
-make V=1 <target>
-    Build target with verbosity 1, instead of 1, 2 can be used too
-make SAMPLES=<path to the fate samples> <target>
-    specify the path to the fate samples at make time
+Makefile variables:
+
+V
+    Disable the default terse mode, the full command issued by make and its
+    output will be shown on the screen.
+
+DESTDIR
+    Destination directory for the install targets, useful to prepare packages
+    or install Libav in cross-environments.
+
+Makefile targets:
+
+all
+    Default target, builds all the libraries and the executables.
+
+fate
+    Run the fate test suite, note you must have installed it
+
+fate-list
+    Will list all fate/regression test targets
+
+install
+    Install headers, libraries and programs.
+
+libavformat/output-example
+    Build the libavformat basic example.
+
+libavcodec/api-example
+    Build the libavcodec basic example.
+
+libswscale/swscale-test
+    Build the swscale self-test (useful also as example).
 
 
 Useful standard make commands:
@@ -21,17 +48,3 @@ make -k
     continue build in case of errors, this is useful for the regression tests
     sometimes but note it will still not run all reg tests.
 
-
-Targets to make:
-fate-list
-    Will list all fate/regression test targets
-
-fate
-    Run the fate test suite, note you must have installed it
-
-
-Setting up local fate:
-use the following command to get the fate test samples
-rsync -aL rsync://rsync.mplayerhq.hu:/samples/fate-suite/ fate/fate-suite
-pass --samples=<path to the samples> to configure or pass the path with the
-SAMPLES variable to make

doc/fate.txt

+FATE Automated Testing Environment
+
+FATE provides a regression testsuite that can be run locally or configured
+to send reports to fate.ffmpeg.org.
+In order to run, it needs a large amount of data (samples and references)
+that is provided separately from the actual source distribution.
+
+Use the following command to get the fate test samples
+
+# rsync -aL rsync://rsync.mplayerhq.hu:/samples/fate-suite/ fate/fate-suite
+
+To inform the build system about the testsuite location, pass
+`--samples=<path to the samples>` to configure or set the SAMPLES Make
+variable or the FATE_SAMPLES environment variable to a suitable value.
+
+For information on how to set up FATE to send results to the official FFmpeg
+testing framework, please refer to the following wiki page:
+http://wiki.multimedia.cx/index.php?title=FATE
+
+FATE Makefile targets:
+
+fate-list
+    Will list all fate/regression test targets.
+
+fate
+    Run the FATE test suite (requires the fate-suite dataset).
+
+Fate Makefile variables:
+
+V
+    Verbosity level, can be set to 0, 1 or 2.
+    * 0: show just the test arguments
+    * 1: show just the command used in the test
+    * 2: show everything
+
+SAMPLES
+    Specify or override the path to the FATE samples at make time, it has a
+    meaning only while running the regression tests.
+
+THREADS
+    Specify how many threads to use while running regression tests, it is
+    quite useful to detect thread-related regressions.
+
+Example:
+    make V=1 SAMPLES=/var/fate/samples THREADS=2 fate

doc/ffmpeg.texi

@@ -586,6 +586,28 @@ Set the ISO 639 language code (3 letters) of the current audio stream.
 @table @option
 @item -atag @var{fourcc/tag}
 Force audio tag/fourcc.
+@item -audio_service_type @var{type}
+Set the type of service that the audio stream contains.
+@table @option
+@item ma
+Main Audio Service (default)
+@item ef
+Effects
+@item vi
+Visually Impaired
+@item hi
+Hearing Impaired
+@item di
+Dialogue
+@item co
+Commentary
+@item em
+Emergency
+@item vo
+Voice Over
+@item ka
+Karaoke
+@end table
 @item -absf @var{bitstream_filter}
 Bitstream filters available are "dump_extra", "remove_extra", "noise", "mp3comp", "mp3decomp".
 @end table

doc/texi2pod.pl

@@ -231,10 +231,12 @@ while(<$inf>) {
 
     # Single line command handlers.
 
-    /^\@(?:section|unnumbered|unnumberedsec|center)\s+(.+)$/
+    /^\@(?:section|unnumbered|unnumberedsec|center|heading)\s+(.+)$/
         and $_ = "\n=head2 $1\n";
-    /^\@subsection\s+(.+)$/
+    /^\@(?:subsection|subheading)\s+(.+)$/
         and $_ = "\n=head3 $1\n";
+    /^\@(?:subsubsection|subsubheading)\s+(.+)$/
+        and $_ = "\n=head4 $1\n";
 
     # Block command handlers:
     /^\@itemize\s*(\@[a-z]+|\*|-)?/ and do {

ffmpeg.c

@@ -2123,6 +2123,7 @@ static int transcode(AVFormatContext **output_files,
                 codec->sample_rate = icodec->sample_rate;
                 codec->channels = icodec->channels;
                 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->codec_id == CODEC_ID_MP3)
                     codec->block_align= 0;

libavcodec/aac_ac3_parser.c

@@ -94,6 +94,7 @@ get_next:
             avctx->channel_layout = s->channel_layout;
         }
         avctx->frame_size = s->samples;
+        avctx->audio_service_type = s->service_type;
     }
 
     avctx->bit_rate = s->bit_rate;

libavcodec/aac_ac3_parser.h

@@ -49,6 +49,7 @@ typedef struct AACAC3ParseContext {
     int bit_rate;
     int samples;
     int64_t channel_layout;
+    int service_type;
 
     int remaining_size;
     uint64_t state;

libavcodec/aaccoder.c

@@ -817,7 +817,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
                 int prevsc = sce->sf_idx[w*16+g];
                 if (dists[w*16+g] > uplims[w*16+g] && sce->sf_idx[w*16+g] > 60) {
                     if (find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]-1))
-                    sce->sf_idx[w*16+g]--;
+                        sce->sf_idx[w*16+g]--;
                     else //Try to make sure there is some energy in every band
                         sce->sf_idx[w*16+g]-=2;
                 }
@@ -1057,7 +1057,7 @@ static void search_for_ms(AACEncContext *s, ChannelElement *cpe,
                     for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
                         M[i] = (sce0->coeffs[start+w2*128+i]
                               + sce1->coeffs[start+w2*128+i]) * 0.5;
-                        S[i] =  sce0->coeffs[start+w2*128+i]
+                        S[i] =  M[i]
                               - sce1->coeffs[start+w2*128+i];
                     }
                     abs_pow34_v(L34, sce0->coeffs+start+w2*128, sce0->ics.swb_sizes[g]);

libavcodec/ac3.h

@@ -87,6 +87,7 @@ typedef struct {
     uint16_t crc1;
     uint8_t sr_code;
     uint8_t bitstream_id;
+    uint8_t bitstream_mode;
     uint8_t channel_mode;
     uint8_t lfe_on;
     uint8_t frame_type;

libavcodec/ac3_parser.c

@@ -69,7 +69,7 @@ int ff_ac3_parse_header(GetBitContext *gbc, AC3HeaderInfo *hdr)
 
         skip_bits(gbc, 5); // skip bsid, already got it
 
-        skip_bits(gbc, 3); // skip bitstream mode
+        hdr->bitstream_mode = get_bits(gbc, 3);
         hdr->channel_mode = get_bits(gbc, 3);
 
         if(hdr->channel_mode == AC3_CHMODE_STEREO) {
@@ -151,6 +151,9 @@ static int ac3_sync(uint64_t state, AACAC3ParseContext *hdr_info,
     hdr_info->channels = hdr.channels;
     hdr_info->channel_layout = hdr.channel_layout;
     hdr_info->samples = hdr.num_blocks * 256;
+    hdr_info->service_type = hdr.bitstream_mode;
+    if (hdr.bitstream_mode == 0x7 && hdr.channels > 1)
+        hdr_info->service_type = AV_AUDIO_SERVICE_TYPE_KARAOKE;
     if(hdr.bitstream_id>10)
         hdr_info->codec_id = CODEC_ID_EAC3;
     else if (hdr_info->codec_id == CODEC_ID_NONE)

libavcodec/ac3dec.c

@@ -273,6 +273,7 @@ static int parse_frame_header(AC3DecodeContext *s)
 
     /* get decoding parameters from header info */
     s->bit_alloc_params.sr_code     = hdr.sr_code;
+    s->bitstream_mode               = hdr.bitstream_mode;
     s->channel_mode                 = hdr.channel_mode;
     s->channel_layout               = hdr.channel_layout;
     s->lfe_on                       = hdr.lfe_on;
@@ -1399,6 +1400,10 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
         if(s->out_channels < s->channels)
             s->output_mode  = s->out_channels == 1 ? AC3_CHMODE_MONO : AC3_CHMODE_STEREO;
     }
+    /* 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;
 
     /* decode the audio blocks */
     channel_map = ff_ac3_dec_channel_map[s->output_mode & ~AC3_OUTPUT_LFEON][s->lfe_on];

libavcodec/ac3dec.h

@@ -79,6 +79,7 @@ typedef struct {
     int bit_rate;                           ///< stream bit rate, in bits-per-second
     int sample_rate;                        ///< sample frequency, in Hz
     int num_blocks;                         ///< number of audio blocks
+    int bitstream_mode;                     ///< bitstream mode                         (bsmod)
     int channel_mode;                       ///< channel mode                           (acmod)
     int channel_layout;                     ///< channel layout
     int lfe_on;                             ///< lfe channel in use

libavcodec/ac3enc.c

@@ -1657,6 +1657,18 @@ static av_cold int validate_options(AVCodecContext *avctx, AC3EncodeContext *s)
     if (s->cutoff > (s->sample_rate >> 1))
         s->cutoff = s->sample_rate >> 1;
 
+    /* validate audio service type / channels combination */
+    if ((avctx->audio_service_type == AV_AUDIO_SERVICE_TYPE_KARAOKE &&
+         avctx->channels == 1) ||
+        ((avctx->audio_service_type == AV_AUDIO_SERVICE_TYPE_COMMENTARY ||
+          avctx->audio_service_type == AV_AUDIO_SERVICE_TYPE_EMERGENCY  ||
+          avctx->audio_service_type == AV_AUDIO_SERVICE_TYPE_VOICE_OVER)
+         && avctx->channels > 1)) {
+        av_log(avctx, AV_LOG_ERROR, "invalid audio service type for the "
+                                    "specified number of channels\n");
+        return AVERROR(EINVAL);
+    }
+
     return 0;
 }
 
@@ -1799,7 +1811,9 @@ static av_cold int ac3_encode_init(AVCodecContext *avctx)
         return ret;
 
     s->bitstream_id   = 8 + s->bit_alloc.sr_shift;
-    s->bitstream_mode = 0; /* complete main audio service */
+    s->bitstream_mode = avctx->audio_service_type;
+    if (s->bitstream_mode == AV_AUDIO_SERVICE_TYPE_KARAOKE)
+        s->bitstream_mode = 0x7;
 
     s->frame_size_min  = 2 * ff_ac3_frame_size_tab[s->frame_size_code][s->bit_alloc.sr_code];
     s->bits_written    = 0;

libavcodec/avcodec.h

@@ -546,6 +546,19 @@ enum AVLPCType {
     AV_LPC_TYPE_NB              , ///< Not part of ABI
 };
 
+enum AVAudioServiceType {
+    AV_AUDIO_SERVICE_TYPE_MAIN              = 0,
+    AV_AUDIO_SERVICE_TYPE_EFFECTS           = 1,
+    AV_AUDIO_SERVICE_TYPE_VISUALLY_IMPAIRED = 2,
+    AV_AUDIO_SERVICE_TYPE_HEARING_IMPAIRED  = 3,
+    AV_AUDIO_SERVICE_TYPE_DIALOGUE          = 4,
+    AV_AUDIO_SERVICE_TYPE_COMMENTARY        = 5,
+    AV_AUDIO_SERVICE_TYPE_EMERGENCY         = 6,
+    AV_AUDIO_SERVICE_TYPE_VOICE_OVER        = 7,
+    AV_AUDIO_SERVICE_TYPE_KARAOKE           = 8,
+    AV_AUDIO_SERVICE_TYPE_NB                   , ///< Not part of ABI
+};
+
 typedef struct RcOverride{
     int start_frame;
     int end_frame;
@@ -2875,6 +2888,13 @@ typedef struct AVCodecContext {
      */
     uint64_t vbv_delay;
 
+    /**
+     * Type of service that the audio stream conveys.
+     * - encoding: Set by user.
+     * - decoding: Set by libavcodec.
+     */
+    enum AVAudioServiceType audio_service_type;
+
     /**
      * Current statistics for PTS correction.
      * - decoding: maintained and used by libavcodec, not intended to be used by user apps

libavcodec/eac3dec.c

@@ -410,7 +410,7 @@ int ff_eac3_parse_header(AC3DecodeContext *s)
 
     /* informational metadata */
     if (get_bits1(gbc)) {
-        skip_bits(gbc, 3); // skip bit stream mode
+        s->bitstream_mode = get_bits(gbc, 3);
         skip_bits(gbc, 2); // skip copyright bit and original bitstream bit
         if (s->channel_mode == AC3_CHMODE_STEREO) {
             skip_bits(gbc, 4); // skip Dolby surround and headphone mode

libavcodec/options.c

@@ -436,6 +436,16 @@ static const AVOption options[]={
 {"slice", NULL, 0, FF_OPT_TYPE_CONST, FF_THREAD_SLICE, INT_MIN, INT_MAX, V|E|D, "thread_type"},
 {"frame", NULL, 0, FF_OPT_TYPE_CONST, FF_THREAD_FRAME, INT_MIN, INT_MAX, V|E|D, "thread_type"},
 {"vbv_delay", "initial buffer fill time in periods of 27Mhz clock", 0, FF_OPT_TYPE_INT64, 0, 0, INT64_MAX},
+{"audio_service_type", "audio service type", OFFSET(audio_service_type), FF_OPT_TYPE_INT, AV_AUDIO_SERVICE_TYPE_MAIN, 0, AV_AUDIO_SERVICE_TYPE_NB-1, A|E, "audio_service_type"},
+{"ma", "Main Audio Service", 0, FF_OPT_TYPE_CONST, AV_AUDIO_SERVICE_TYPE_MAIN,              INT_MIN, INT_MAX, A|E, "audio_service_type"},
+{"ef", "Effects",            0, FF_OPT_TYPE_CONST, AV_AUDIO_SERVICE_TYPE_EFFECTS,           INT_MIN, INT_MAX, A|E, "audio_service_type"},
+{"vi", "Visually Impaired",  0, FF_OPT_TYPE_CONST, AV_AUDIO_SERVICE_TYPE_VISUALLY_IMPAIRED, INT_MIN, INT_MAX, A|E, "audio_service_type"},
+{"hi", "Hearing Impaired",   0, FF_OPT_TYPE_CONST, AV_AUDIO_SERVICE_TYPE_HEARING_IMPAIRED,  INT_MIN, INT_MAX, A|E, "audio_service_type"},
+{"di", "Dialogue",           0, FF_OPT_TYPE_CONST, AV_AUDIO_SERVICE_TYPE_DIALOGUE,          INT_MIN, INT_MAX, A|E, "audio_service_type"},
+{"co", "Commentary",         0, FF_OPT_TYPE_CONST, AV_AUDIO_SERVICE_TYPE_COMMENTARY,        INT_MIN, INT_MAX, A|E, "audio_service_type"},
+{"em", "Emergency",          0, FF_OPT_TYPE_CONST, AV_AUDIO_SERVICE_TYPE_EMERGENCY,         INT_MIN, INT_MAX, A|E, "audio_service_type"},
+{"vo", "Voice Over",         0, FF_OPT_TYPE_CONST, AV_AUDIO_SERVICE_TYPE_VOICE_OVER,        INT_MIN, INT_MAX, A|E, "audio_service_type"},
+{"ka", "Karaoke",            0, FF_OPT_TYPE_CONST, AV_AUDIO_SERVICE_TYPE_KARAOKE,           INT_MIN, INT_MAX, A|E, "audio_service_type"},
 {NULL},
 };
 

libavcodec/version.h

@@ -21,8 +21,8 @@
 #define AVCODEC_VERSION_H
 
 #define LIBAVCODEC_VERSION_MAJOR 52
-#define LIBAVCODEC_VERSION_MINOR 114
-#define LIBAVCODEC_VERSION_MICRO  1
+#define LIBAVCODEC_VERSION_MINOR 115
+#define LIBAVCODEC_VERSION_MICRO  0
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
                                                LIBAVCODEC_VERSION_MINOR, \

libavcodec/vp8.c

@@ -512,7 +512,7 @@ void decode_mvs(VP8Context *s, VP8Macroblock *mb, int mb_x, int mb_y)
         mb->mode = VP8_MVMODE_MV;
 
         /* If we have three distinct MVs, merge first and last if they're the same */
-        if (cnt[CNT_SPLITMV] && AV_RN32A(&near_mv[1+VP8_EDGE_TOP]) == AV_RN32A(&near_mv[1+VP8_EDGE_TOPLEFT]))
+        if (cnt[CNT_SPLITMV] && AV_RN32A(&near_mv[1 + VP8_EDGE_TOP]) == AV_RN32A(&near_mv[1 + VP8_EDGE_TOPLEFT]))
             cnt[CNT_NEAREST] += 1;
 
         /* Swap near and nearest if necessary */

libavformat/mov.c

@@ -492,16 +492,20 @@ static int mov_read_esds(MOVContext *c, AVIOContext *pb, MOVAtom atom)
 static int mov_read_dac3(MOVContext *c, AVIOContext *pb, MOVAtom atom)
 {
     AVStream *st;
-    int ac3info, acmod, lfeon;
+    int ac3info, acmod, lfeon, bsmod;
 
     if (c->fc->nb_streams < 1)
         return 0;
     st = c->fc->streams[c->fc->nb_streams-1];
 
     ac3info = avio_rb24(pb);
+    bsmod = (ac3info >> 14) & 0x7;
     acmod = (ac3info >> 11) & 0x7;
     lfeon = (ac3info >> 10) & 0x1;
     st->codec->channels = ((int[]){2,1,2,3,3,4,4,5})[acmod] + lfeon;
+    st->codec->audio_service_type = bsmod;
+    if (st->codec->channels > 1 && bsmod == 0x7)
+        st->codec->audio_service_type = AV_AUDIO_SERVICE_TYPE_KARAOKE;
 
     return 0;
 }

libavformat/utils.c

@@ -2445,6 +2445,19 @@ int av_find_stream_info(AVFormatContext *ic)
         }else if(st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
             if(!st->codec->bits_per_coded_sample)
                 st->codec->bits_per_coded_sample= av_get_bits_per_sample(st->codec->codec_id);
+            // set stream disposition based on audio service type
+            switch (st->codec->audio_service_type) {
+            case AV_AUDIO_SERVICE_TYPE_EFFECTS:
+                st->disposition = AV_DISPOSITION_CLEAN_EFFECTS;    break;
+            case AV_AUDIO_SERVICE_TYPE_VISUALLY_IMPAIRED:
+                st->disposition = AV_DISPOSITION_VISUAL_IMPAIRED;  break;
+            case AV_AUDIO_SERVICE_TYPE_HEARING_IMPAIRED:
+                st->disposition = AV_DISPOSITION_HEARING_IMPAIRED; break;
+            case AV_AUDIO_SERVICE_TYPE_COMMENTARY:
+                st->disposition = AV_DISPOSITION_COMMENT;          break;
+            case AV_AUDIO_SERVICE_TYPE_KARAOKE:
+                st->disposition = AV_DISPOSITION_KARAOKE;          break;
+            }
         }
     }