* release/0.7: (33 commits)
Update for 0.7.8
svq1dec: call avcodec_set_dimensions() after dimensions changed. Fixes NGS00148
vp3dec: Check coefficient index in vp3_dequant() Fixes NGS00145
qdm2dec: fix buffer overflow. Fixes NGS00144
h264: Fix invalid interlaced progressive MB combinations for direct mode prediction. Fixes Ticket312
mpegvideo: dont use ff_mspel_motion() for vc1 Fixes Ticket655
imgutils: Fix illegal read.
ac3probe: Detect Sonic Foundry Soft Encode AC3 as raw AC3. Our ac3 code chain can handle it fine. More ideal would be to write a demuxer that actually extracts what can be from the additional headers and uses it for whatever it can be used for.
mjpeg: support mpo Fixes stereoscopic_photo.mpo
Add a version bump and APIchanges entry for avcodec_open2 and avformat_find_stream_info.
lavf: fix multiplication overflow in avformat_find_stream_info()
lavf: fix invalid reads in avformat_find_stream_info()
lavf: add avformat_find_stream_info()
lavc: fix parentheses placement in avcodec_open2().
lavc: introduce avcodec_open2() as a replacement for avcodec_open().
rawdec: use a default sample rate if none is specified. Fixes "ffmpeg -f s16le -i /dev/zero"
rawdec: add check on sample_rate
qdm2dec: check remaining input bits in the mainloop of qdm2_fft_decode_tones() This is neccessary but likely not sufficient to prevent out of array reads.
cinepak: check strip_size
wma: Check channel number before init. Fixes Ticket240
...
Conflicts:
RELEASE
doc/APIchanges
libavcodec/avcodec.h
libavcodec/utils.c
libavcodec/version.h
libavdevice/v4l2.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* release/0.7: (290 commits)
nuv: Fix combination of size changes and LZO compression.
av_lzo1x_decode: properly handle negative buffer length.
Do not call parse_keyframes_index with NULL stream.
update versions for 0.7 branch
Version numbers for 0.8.6
snow: emu edge support Fixes Ticket592
imc: validate channel count
imc: check for ff_fft_init() failure (cherry picked from commit 95fee70d6773fde1c34ff6422f48e5e66f37f263)
libgsmdec: check output buffer size before decoding (cherry picked from commit b03761b1309293bbf30edef767503875277b01cf)
configure: fix arch x86_32
mp3enc: avoid truncating id3v1 tags by one byte
asfdec: Check packet_replic_size earlier
cin audio: validate the channel count
binkaudio: add some buffer overread checks.
atrac1: validate number of channels (cherry picked from commit bff5b2c1ca1290ea30587ff2f76171f9e3854872)
atrac1: check output buffer size before decoding (cherry picked from commit 33684b9c12b74c0140fb91e8150263db4a48d55e)
vp3: fix oob read for negative tokens and memleaks on error. (cherry picked from commit 8370e426e42f2e4b9d14a1fb8107ecfe5163ce7f)
apedec: set s->currentframeblocks after validating nblocks
apedec: use unsigned int for 'nblocks' and make sure that it's within int range
apedec: check for data buffer realloc failure (cherry picked from commit 11ca8b2d7486e879926488404b3b79af774f0f2d)
...
Conflicts:
Changelog
Makefile
RELEASE
configure
libavcodec/error_resilience.c
libavcodec/mpegvideo.c
libavformat/matroskaenc.c
tests/ref/lavf/mxf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows skipping past unsupported RTCP packet types, as
RFC 3550 section 6.1 mandates.
Currently this only has any practical effect if a sender puts
an unrecognized type before RTCP_BYE in a compounded packet, or
(incorrectly) does not put RTCP_SR first.
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 07b77fe3871f86b87e35876d38f1969da5ece4b2)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
We actually read 20 bytes of these packets.
Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 5d6ecf5345c0913e2b66427ea062e7989201a139)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
dnxhddec: optimise dnxhd_decode_dct_block()
rtp: remove disabled code
eac3enc: use different numbers of blocks per frame to allow higher bitrates
dnxhd: add regression test for 10-bit
dnxhd: 10-bit support
dsputil: update per-arch init funcs for non-h264 high bit depth
dsputil: template get_pixels() for different bit depths
dsputil: create 16/32-bit dctcoef versions of some functions
jfdctint: add 10-bit version
mov: add clcp type track as Subtitle stream.
mpeg4: add Mpeg4 Profiles names.
mpeg4: decode Level Profile for MPEG4 Part 2.
ffprobe: display bitstream level.
imgconvert: remove unused glue and xglue macros
Conflicts:
libavcodec/dsputil_template.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (40 commits)
H.264: template left MB handling
H.264: faster fill_decode_caches
H.264: faster write_back_*
H.264: faster fill_filter_caches
H.264: make filter_mb_fast support the case of unavailable top mb
Do not include log.h in avutil.h
Do not include pixfmt.h in avutil.h
Do not include rational.h in avutil.h
Do not include mathematics.h in avutil.h
Do not include intfloat_readwrite.h in avutil.h
Remove return statements following infinite loops without break
RTSP: Doxygen comment cleanup
doxygen: Escape '\' in Doxygen documentation.
md5: cosmetics
md5: use AV_WL32 to write result
md5: add fate test
md5: include correct headers
md5: fix test program
doxygen: Drop array size declarations from Doxygen parameter names.
doxygen: Fix parameter names to match the function prototypes.
...
Conflicts:
libavcodec/x86/dsputil_mmx.c
libavformat/flvenc.c
libavformat/oggenc.c
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (21 commits)
build: simplify commands for clean target
swscale: split swscale.c in unscaled and generic conversion routines.
swscale: cosmetics.
swscale: integrate (literally) swscale_template.c in swscale.c.
swscale: split out x86/swscale_template.c from swscale.c.
swscale: enable hScale_altivec_real.
swscale: split out ppc _template.c files from main swscale.c.
swscale: remove indirections in ppc/swscale_template.c.
swscale: split out unscaled altivec YUV converters in their own file.
mpegvideoenc: fix multislice fate tests with threading disabled.
mpegts: Wrap #ifdef DEBUG and av_hex_dump_log() combination in a macro.
build: Simplify texi2html invocation through the --output option.
Mark some variables with av_unused
Replace avcodec_get_pix_fmt_name() by av_get_pix_fmt_name().
svq3: Check negative mb_type to fix potential crash.
svq3: Move svq3-specific fields to their own context.
rawdec: initialize return value to 0.
Remove unused get_psnr() prototype
rawdec: don't leak option strings.
bktr: get default framerate from video standard.
...
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Most of these variables are only used in av_dlog statements, some
are required but not used by other macros.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* 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>
Adding _POSIX_C_SOURCE to CPPFLAGS globally produces all sorts of problems
since it causes certain system functions to be hidden on some (BSD) systems.
The solution is to only add the flag on systems that really require it, i.e.
glibc-based ones.
This change makes BSD systems compile out-of-the-box without the need for
adding specific flags manually. It also allows dropping a number of flags
set manually on a file-per-file basis, but were only present to work around
breakage introduced by the presence of _POSIX_C_SOURCE.
Also add _XOPEN_SOURCE to CPPFLAGS for glibc systems. We use XSI extensions
in several places already, so it is preferable to define it globally instead
of littering source files with individual #defines only needed for glibc.
* qatar/master: (22 commits)
ac3enc: move extract_exponents inner loop to ac3dsp
avio: deprecate url_get_filename().
avio: deprecate url_max_packet_size().
avio: make url_get_file_handle() internal.
avio: make url_filesize() internal.
avio: make url_close() internal.
avio: make url_seek() internal.
avio: cosmetics, move AVSEEK_SIZE/FORCE declarations together
avio: make url_write() internal.
avio: make url_read_complete() internal.
avio: make url_read() internal.
avio: make url_open() internal.
avio: make url_connect internal.
avio: make url_alloc internal.
applehttp: Merge two for loops
applehttp: Restructure the demuxer to use a custom AVIOContext
applehttp: Move finished and target_duration to the variant struct
aacenc: reduce the number of loop index variables
avio: deprecate url_open_protocol
avio: deprecate url_poll and URLPollEntry
...
Conflicts:
libavformat/applehttp.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fate: fix partial run when no samples path is specified
ARM: NEON fixed-point forward MDCT
ARM: NEON fixed-point FFT
lavf: bump minor version and add an APIChanges entry for avio changes
avio: simplify url_open_dyn_buf_internal by using avio_alloc_context()
avio: make url_fdopen internal.
avio: make url_open_dyn_packet_buf internal.
avio: avio_ prefix for url_close_dyn_buf
avio: avio_ prefix for url_open_dyn_buf
avio: introduce an AVIOContext.seekable field
ac3enc: use generic fixed-point mdct
lavfi: add fade filter
Change yadif to not use out of picture lines.
lavc: deprecate AVCodecContext.antialias_algo
lavc: mark mb_qmin/mb_qmax for removal on next major bump.
Conflicts:
doc/filters.texi
libavcodec/ac3enc_fixed.h
libavcodec/ac3enc_float.h
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/vf_fade.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It doesn't look fit to be a part of the public API.
Adding a temporary hack to ffserver to be able to use it, should be
cleaned up when somebody is up for it.
In the name of consistency:
put_byte -> avio_w8
put_<type> -> avio_w<type>
put_buffer -> avio_write
put_nbyte will be made private
put_tag will be merged with avio_put_str
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
For MS-RTSP, we don't always get RTCP packets (never?), so the earlier
timestamping code never wrote anything into pkt->pts. The rtpdec_asf
depacketizer just sets the dts of the packet, so if the generic RTP
timestamping is used, too, we get inconsistent timestamps.
Therefore, skip the generic RTP timestamp algorithm if the depacketizer
already has set something.
This fixes "Invalid timestamps" warnings, present since SVN rev 26187.
Originally committed as revision 26241 to svn://svn.ffmpeg.org/ffmpeg/trunk
Emitted timestamps in each stream start from 0, for the first received
RTP packet. Once an RTCP packet is received, that one is used for
sync, emitting timestamps that fit seamlessly into the earlier ones.
Originally committed as revision 26187 to svn://svn.ffmpeg.org/ffmpeg/trunk
This also reverts SVN rev 26016, which incorrectly overwrote the time base
with 90 kHz for all streams, regardless of what was set by the SDP parsing.
The stream that triggered the fix in 26016 still works after this commit.
Originally committed as revision 26022 to svn://svn.ffmpeg.org/ffmpeg/trunk
The generic default is 0/0 and that obviously triggers once the value is used.
Originally committed as revision 26016 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes cases where the RTP time base and the sample rate of the stream
differ. Previously, the AVStream time_base was unconditionally set to
the sample rate (which initially was set to one value when parsing the
rtpmap field in the SDP, but later overridden by an a=SampleRate field).
Additionally, this makes the code actually use the stream time base set
in rtpmap for video codecs, instead of hardcoding it to always be 90 kHz.
Originally committed as revision 25908 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes playback of a RealRTSP/MP3 URL from the RTSP samples on
MultimediaWiki.
Originally committed as revision 25906 to svn://svn.ffmpeg.org/ffmpeg/trunk
This indicates that there was no error that needs to be reported to the
caller, so we can move on to parse the next packet immediately, if
available. The only error code that ff_mpegts_parse_packet can return
indicates that there was no packet to return from the provided data, for
which it returns -1.
Originally committed as revision 25496 to svn://svn.ffmpeg.org/ffmpeg/trunk
It may have returned a negative number for an error (e.g. AVERROR(EAGAIN),
if more data is required for it to be able to return a complete packet).
Originally committed as revision 25458 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes roundup issue 2270.
Patch by Robert Schlabbach, robert_s at gmx dot net
Originally committed as revision 25372 to svn://svn.ffmpeg.org/ffmpeg/trunk
Do the same change for ff_rdt_parse_packet, too, to keep the interfaces
similar.
Originally committed as revision 25289 to svn://svn.ffmpeg.org/ffmpeg/trunk
