Or in other word, do not go to the next line when the text cannot be
rendered within the frame.

The rationale is that wrapping is hard to manage, as it depends on the
position of the text, thus making hard/impossible to compute the size
occupied by the rendered text, and makes the filter behavior
unpredictible with moving text (implemented in a pending patch).

Also the current wrapping code is very limited (for example it splits
words in the middle), and correctly handling wrapping requires much
more complexity (alignment, gravity, word parsing etc.), while
drawtext is meant to be only a thin wrapper around libfreetype, so
such advanced features don't belong here.

Fixes Ticket149
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

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

libpostproc/postprocess_template.c: In function 'dering_MMX2':
libpostproc/postprocess_template.c:1045:5: error: can't find a register in class 'GENERAL_REGS' while reloading 'asm'
libpostproc/postprocess_template.c:1045:5: error: 'asm' operand has impossible constraints

* qatar/master:
  rtp: factorize  dynamic payload type fallback
  flvdec: Ignore the index if it's from a creator known to be different
  cmdutils: move grow_array out of #if CONFIG_AVFILTER
  avconv: actually set InputFile.rate_emu
  ratecontrol: update last_qscale_for sooner
  Fix unnecessary shift with 9/10bit vertical scaling
  prores: mark prores as intra-only in libavformat/utils.c:is_intra_only()
  prores: return more meaningful error values
  prores: improve error message wording
  prores: cosmetics: prettyprinting, drop useless parentheses
  prores: lowercase AVCodec name entry

Conflicts:
	cmdutils.c
	libavcodec/proresdec_lgpl.c
	tests/ref/lavfi/pixfmts_scale
Merged-by: Michael Niedermayer <michaelni@gmx.at>

Fixes Ticket200
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

aacdec: Make aac-latm closer to the spec (not reading random data leading to random false configurations)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

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

occurd during debug. I dont know if this can happen normally but if so
it would be quite bad.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Move the identical code in rtp_write_header() and
ff_sdp_write_media() inside ff_rtp_get_payload_type()
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>

Signed-off-by: Martin Storsjö <martin@martin.st>

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

The score of 50 can probably be raised if needed
Fixes Ticket490
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

Its current maintainer is too lazy/busy to actively maintain it.

fixes programs linking with --disable-avfilter

The code before was unsafe, had a race condition and could crash.
This Fixes Ticket499
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

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

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

This fixes changing channels
It possibly might cause regressions but i cant avoid this without having a test
case that needs the locking code.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

In order to make ratecontrol converge properly update the
last_qscale_for values when populating the qscale array.

It fixes the issue reported on
http://lists.libav.org/pipermail/libav-tools/2011-September/000050.html

Signed-off-by: Diego Biurrun <diego@biurrun.de>

All other codecs use lowercase names in that field.

It may be needed to use values greater than 72, and in general greater
than an arbitrary big value. Let the user choose what's too big for
her.

Authorship of the decoders is elvis / maxim. For more details see previous
comits
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

* qatar/master:
  Add LATM demuxer
  avplay: flush audio decoder with empty packets at EOF if the decoder has CODEC_CAP_DELAY set.
  8svx/iff: fix decoding of compressed stereo 8svx files.
  8svx: log an error message if output buffer is too small
  8svx: check packet size before reading the initial sample value.
  8svx: output 8-bit samples instead of 16-bit.
  8svx: split delta decoding into a separate function.
  mp4: Don't read an empty Decoder Config Descriptor
  fate.sh: Ignore errors from rm command during cleanup.
  fate.sh: Run git-pull in quiet mode to avoid console spam.
  Apple ProRes decoder
  rtmp: Make the input FLV parser handle data cut at any point
  rv34: Check for invalid slices offsets
  eval: test isnan(sqrt(-1)) instead of just sqrt(-1)

Conflicts:
	Changelog
	libavcodec/8svx.c
	libavcodec/proresdec.c
	libavcodec/version.h
	libavformat/iff.c
	libavformat/version.h
	tests/ref/fate/eval
Merged-by: Michael Niedermayer <michaelni@gmx.at>

This change fixes a bug where seeking doesn't work properly for
matroska files that have the CUES element before the first cluster.
This bug was accidentally introduced a few months ago by my deferred CUES
loading patch<http://git.videolan.org/?p=ffmpeg.git;a=commit;h=31ad14c21e0735387ba8082c6e3436241f7ccfc8>
.

When the CUES element appears before the first cluster in the file, the data
is parsed and placed in matroska->index but that data is never added to the
seek index. Currently the transfer from matroska->index to the seek index
only happens when matroska_parse_cues() is called.
Matroska_parse_cues() only gets called on a seek if cues_parsing_deferred is
set. Cues_parsing_deferred only gets set if parsing the CUES requires
seeking past the first cluster. There is no code to handle the case where
CUES is before the first cluster.

This fix essentially restores the matroska->index processing that was
happening at the end of matroska_read_header() before I made my CUES
deferral change. In the case where CUES is before the first
cluster, matroska->index will have data and the seek index will be updated.
In the case where CUES is later in the file, matroska->index will be empty
and cues_parsing_deferred will be set so loading will happen later.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

This is a raw demuxer for the AAC LATM decoder and thus limited to
single stream LOAS.

CODEC_CAP_DELAY set.

Make the iff demuxer send the whole audio chunk to the decoder as a
single packet and move stereo interleaving from the iff demuxer to the
decoder.

Based on a patch by Stefano Sabatini.
git.videolan.org/ffmpeg.git
commit e280a4da