* commit '04d2f9ace3fb6e880f3488770fc5a39de5b63cbb':
mvi: Add sanity checking for the audio frame size
alac: Do bounds checking of lpc_order read from the bitstream
xwma: Avoid division by zero
avidec: Make sure a packet is large enough before reading its data
vqf: Make sure the bitrate is in the valid range
vqf: Make sure sample_rate is set to a valid value
electronicarts: Check packet sizes before reading
lavf: Avoid setting avg_frame_rate if delta_dts is negative
vc1dec: Undo mpegvideo initialization if unable to allocate tables
vc1dec: Fix leaks in ff_vc1_decode_init_alloc_tables on errors
wnv1: Make sure the input packet is large enough
dcadec: Validate the lfe parameter
Conflicts:
libavcodec/dcadec.c
libavcodec/wnv1.c
libavformat/avidec.c
libavformat/electronicarts.c
libavformat/utils.c
libavformat/xwma.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'fa6eef4210c2fd7f7324d558b09311c75987a31e':
wtv: Mark attachment with a negative stream id
avconv: do not use lavfi direct rendering with -deinterlace
avidec: Let the inner dv demuxer take care of discarding
Update Changelog
kmvc: Clip pixel position to valid range
kmvc: use fixed sized arrays in the context
indeo: reject negative array indexes
indeo: Cosmetic formatting
indeo: Refactor ff_ivi_init_tiles and ivi_decode_blocks
indeo: Refactor ff_ivi_dec_huff_desc
indeo: use a typedef for the mc function pointer
indeo: use proper error code
Conflicts:
Changelog
ffmpeg.c
libavcodec/ivi_common.c
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In some case for aac in AVI, avidec extracts wrong PTS value.
(www.ffmpeg.org/trac/ffmpeg/ticket/1755)
I found additional case(ss=4096) and add condition.
Problematic file link : https://docs.google.com/open?id=0B6r7ZfWFIypCOTdZQUtGVEdJUUE
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The special cases in demuxers and decoders are a mess otherwise (and more
would be needed to support it fully)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mpc8: return more meaningful error codes.
mpc: return more meaningful error codes.
wv,mpc8: don't return apetag data in packets.
rtmp: do not warn about receiving metadata packets
x86: h264dsp: Adjust YASM #ifdefs
x86: yadif: Mark mmxext optimizations as such
h264: convert loop filter strength dsp function to yasm.
Improve descriptiveness of a number of codec and container long names
Conflicts:
libavcodec/flvdec.c
libavcodec/libopenjpegdec.c
libavformat/apetag.c
libavformat/mp3dec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
movenc: Write chan atom for all audio tracks in mov mode movies.
mpegtsenc: use avio_open_dyn_buf(), zero pointers after freeing
doc/avconv: add some details about the transcoding process.
avidec: make scale and rate unsigned.
avconv: check output stream recording time before each frame returned from filters
avconv: split selecting input file out of transcode().
avconv: split checking for active outputs out of transcode().
avfiltergraph: make some functions static.
Conflicts:
ffmpeg.c
libavfilter/avfiltergraph.c
libavfilter/internal.h
libavformat/mpegtsenc.c
tests/ref/fate/acodec-alac
tests/ref/fate/acodec-pcm-s16be
tests/ref/fate/acodec-pcm-s24be
tests/ref/fate/acodec-pcm-s32be
tests/ref/fate/acodec-pcm-s8
tests/ref/lavf/mov
Merged-by: Michael Niedermayer <michaelni@gmx.at>
0-sized packets are used to implement variable fps.
However there seems to be a variation where these are not
even stored in the main file but as 0-size index entries
only.
This fixes the sample in trac issue #957, it now plays both
the same ways as in MPlayer and in a way that looks correct.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master: (27 commits)
avconv: free packet in write_frame() when discarding due to frame number limit
FATE: use +/- flag option syntax for vp8 emu-edge tests
lavf: make av_interleave_packet_per_dts() private.
lavf: deprecate av_read_packet().
oggdec: output correct timestamps for Vorbis
avconv: pass input stream timestamps to audio encoders
lavc: shrink encoded audio packet size after encoding.
xa: set correct bit rate
xa: do not set bit_rate, block_align, or bits_per_coded_sample
xa: fix end-of-file handling
xa: fix timestamp calculation
bink: fix typo in FFALIGN() argument
bink: align plane width to 8 when calculating bundle sizes
doc: pass -Idoc texi2html and texi2pod
doc: texi2pod: add -I flag
movenc: Add a min_frag_duration option
rtsp: Set the default delay to 0.1 s for the RTSP/SDP/RTP demuxers
libavformat: Set the default for the max_delay option to -1
Generate manpages for AV{Format,Codec}Context AVOptions.
doc/avconv: remove entries for AVOptions.
...
Conflicts:
doc/Makefile
doc/ffmpeg.texi
doc/muxers.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/options.c
libavcodec/vp8.c
libavformat/options.c
tests/fate/demux.mak
tests/ref/fate/truemotion1-15
tests/ref/fate/truemotion1-24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
