Some received packets can have size 0. The return value from
av_malloc(0) may be NULL, which is ok if the size was 0. On
OS X, however, the returned pointer is non-null but leads to
crashes when trying to free it.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (22 commits)
configure: enable memalign_hack automatically when needed
swscale: unbreak the build on non-x86 systems.
swscale: remove if(bitexact) branch from functions.
swscale: remove if(canMMX2BeUsed) conditional.
swscale: remove swScale_{c,MMX,MMX2} duplication.
swscale: use emms_c().
Move emms_c() from libavcodec to libavutil.
tiff: set palette in the context when specified in TIFF_PAL tag
rtsp: use strtoul to parse rtptime and seq values.
pgssubdec: fix incorrect colors.
dvdsubdec: fix incorrect colors.
ape: Allow demuxing of files with metadata tags.
swscale: remove dead macro WRITEBGR24OLD.
swscale: remove AMD3DNOW "optimizations".
swscale: remove duplicate code in ppc/ subdirectory.
swscale: remove duplicated x86/ functions.
swscale: force --enable-runtime-cpudetect and remove SWS_CPU_CAPS_*.
vsrc_buffer.h: add file doxy
vsrc_buffer: tweak error message in init()
msmpeg4: reindent.
...
Merged-by: Michael Niedermayer <michaelni@gmx.at>
strtol could return negative values, leading to various error messages,
mainly "non-monotonically increasing dts".
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The new function accepts a slightly more intuitive order of paramters,
and returns an error code, thus allowing applications to report a
meaningful error message.
* qatar/master:
ffmpeg: get rid of the -vglobal option.
dct32: Add AVX implementation of 32-point DCT
dct32: Change pass 6 permutation to allow for AVX implementation
dct32: port SSE 32-point DCT to YASM
multiple inclusion guard cleanup
avio: document buffer must created with av_malloc() and friends
avio: check AVIOContext malloc failure
swscale: point out an alternative to sws_getContext
svq3: Do initialization after parsing the extradata
add changelog entries for 0.7_beta2
mp3lame: add #include required for AV_RB32 macro.
Conflicts:
Changelog
libavcodec/svq3.c
libavcodec/x86/dct32_sse.c
libavfilter/vsrc_buffer.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
configure: make executable again
LATM/AAC: Free previously initialized context on reinit.
configure: Do not unconditionally add -Wall to host CFLAGS.
configure: Set OS/2 objformat to a.out.
Add support for a.out object format to assembler macros.
fate: disable threading for encoding
fate: add comment field
fate: allow overriding default build and install dirs
mpegtsenc: Add an AVClass pointer to the private data
mpegaudio: clean up #includes
mpegaudio: move all header parsing to mpegaudiodecheader.[ch]
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Since a private class is set for this muxer, the callers will
assume that the private data starts with an AVClass pointer.
If no such member exists, the first few bytes of the struct
will be overwritten, and the class pointer may be broken at
any later time.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
qdm2: Use floating point synthesis filter.
h264: correct border check.
h264: fix loopfilter with threading at slice boundaries.
Fix ff_mpa_synth_filter_fixed() prototype
Rename costablegen.c ---> cos_tablegen.c.
Collapse tableprint.c into tableprint.h.
Simplify trig table rules
Remove potentially unstable filenames from comments in generated files.
Ignore generated tables and generated table generator programs.
Simplify CLEANFILES make variable by using wildcards.
Remove silly insults from avformat_version() Doxygen documentation.
mpegaudiodsp: fix x86 and ppc makefiles
configure: Adjust AVX assembler check.
mpegaudio: remove unused version of SAME_HEADER_MASK
mpegaudio: remove useless #undef at end of file
asfdec: add missing #include for av_bswap32()
mpegaudio: merge two #if CONFIG_FLOAT blocks
mpegaudio: move some struct definitions from mpegaudio.h
Move some mpegaudio functions to new mpegaudiodsp subsystem
Conflicts:
libavcodec/h264.c
libavcodec/x86/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In ff_id3v2_parse(), prevent unsigned integer overflow if data length
indicator is skipped and tlen is < 4.
Fix crash decoding file Allaby_cut.mp3, fix trac issue #182.
* qatar/master:
Fix compilation of iirfilter-test.
libx264: handle closed GOP codec flag
lavf: remove duplicate assignment in avformat_alloc_context.
lavf: use designated initializers for AVClasses.
flvdec: clenup debug code
asfdec: fix possible overread on broken files.
asfdec: do not fall back to binary/generic search
asfdec: reindent after previous commit c7bd5ed
asfdec: fallback to binary search internally
mpegaudio: add _fixed suffix to some names
Modify x86util.asm to ease transitioning to 10-bit H.264 assembly.
dct: build dct32 as separate object files
qdm2: include correct header for rdft
Conflicts:
ffpresets/libx264-fast.ffpreset
ffpresets/libx264-fast_firstpass.ffpreset
ffpresets/libx264-faster.ffpreset
ffpresets/libx264-faster_firstpass.ffpreset
ffpresets/libx264-medium.ffpreset
ffpresets/libx264-medium_firstpass.ffpreset
ffpresets/libx264-placebo.ffpreset
ffpresets/libx264-placebo_firstpass.ffpreset
ffpresets/libx264-slow.ffpreset
ffpresets/libx264-slow_firstpass.ffpreset
ffpresets/libx264-slower.ffpreset
ffpresets/libx264-slower_firstpass.ffpreset
ffpresets/libx264-superfast.ffpreset
ffpresets/libx264-superfast_firstpass.ffpreset
ffpresets/libx264-ultrafast.ffpreset
ffpresets/libx264-ultrafast_firstpass.ffpreset
ffpresets/libx264-veryfast.ffpreset
ffpresets/libx264-veryfast_firstpass.ffpreset
ffpresets/libx264-veryslow.ffpreset
ffpresets/libx264-veryslow_firstpass.ffpreset
libavformat/flvdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Moving the search and parsing of the 'fmt ' info the main loop of wav_read_header() allows tags that precede it to be parsed.
Creating wav_parse_fmt_tag() makes wav_read_header() easier to read.
asf_read_seek() inside the asf demuxer already does the
right thing, it tries the index and if that fails it uses
binary search. If binary search is called from outside of asfdec.c
it will fail because the asf code cannot clean up after itself.
Therefore introduce AVFMT_NOBINSEARCH that prevents the seek
code to fallback to binary search and AVFMT_NOGENSEARCH that
prevents the seek code to fallback to generic search.
Make the iff demuxer send the whole audio chunk to the decoder as a
single packet, move stereo interleaving from the iff demuxer to the
decoder, and introduce an 8svx_raw decoder which performs
stereo interleaving.
This is required for handling stereo data correctly, indeed samples
are stored like:
LLLLLL....RRRRRR
that is all left samples are at the beginning of the chunk, all right
samples at the end, so it is necessary to store and process the whole
buffer in order to decode each frame. Thus the decoder needs all the
audio chunk before it can return interleaved data.
Fix decoding of files 8svx_exp.iff and 8svx_fib.iff, fix trac issue #169.
Create separate fields 8svx_compression (for audio compression), and
bitmap_compression (for video compression), and perform minor related
logging tweaks.
Improve clarity, also simplify the case when both types of compression
are employed in the same file.
* qatar/master: (33 commits)
rtpdec_qdm2: Don't try to parse data packet if no configuration is received
ac3enc: put the counting of stereo rematrixing bits in the same place to make the code easier to understand.
ac3enc: clean up count_frame_bits() and count_frame_bits_fixed()
mpegvideo: make FF_DEBUG_DCT_COEFF output coeffs via av_log() instead of just via AVFrame.
srtdec: make sure we don't write past the end of buffer
wmaenc: improve channel count and bitrate error handling in encode_init()
matroskaenc: make sure we don't produce invalid file with no codec ID
matroskadec: check that pointers were initialized before accessing them
lavf: fix function name in compute_pkt_fields2 av_dlog message
lavf: fix av_find_best_stream when providing a wanted stream.
lavf: fix av_find_best_stream when decoder_ret is given and using a related stream.
ffmpeg: factorize quality calculation
tiff: add support for SamplesPerPixel tag in tiff_decode_tag()
tiff: Prefer enum TiffCompr over int for TiffContext.compr.
mov: Support edit list atom version 1.
configure: Enable libpostproc automatically if GPL code is enabled.
Cosmetics: fix prototypes in oggdec
oggdec: fix memleak with continuous streams.
matroskaenc: add missing new line in av_log() call
dnxhdenc: add AVClass in private context.
...
swscale changes largely rewritten by me or replaced by baptsites due to lots of bugs in ronalds code.
Above code is also just in case its not obvios to a large extended duplicates that where cherry picked
from ffmpeg.
Conflicts:
configure
ffmpeg.c
libavformat/matroskaenc.c
libavutil/pixfmt.h
libswscale/ppc/swscale_template.c
libswscale/swscale.c
libswscale/swscale_template.c
libswscale/utils.c
libswscale/x86/swscale_template.c
tests/fate/h264.mak
tests/ref/lavfi/pixdesc_le
tests/ref/lavfi/pixfmts_copy_le
tests/ref/lavfi/pixfmts_null_le
tests/ref/lavfi/pixfmts_scale_le
tests/ref/lavfi/pixfmts_vflip_le
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The later parsing of payload data depends on the configuration
being present. If it hasn't been configured properly yet,
parsing a data packet may lead to a crash.
Signed-off-by: Martin Storsjö <martin@martin.st>
In the main loop, stream_number is incremented after checking the stream type,
so the search usually will not find the wanted stream.
This patch eliminates the useless stream_number variable and introduces a new
one, called real_stream_index to store the real stream index of the current
stream, no matter if we are looping through all the streams or only the streams
of a program.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Yet another fix for the code originally designed for use without related_stream.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Add an extra size validity check in asf_read_frame_header(). Without
this asf->packet_size_left may become negative, which triggers an
assertion failure later.
This avoids the creation of a new AVStream instead of replacing it when
a stream reset occurs (track change with some webradios for example).
Signed-off-by: Diego Biurrun <diego@biurrun.de>