To avoid allocating ridiculous amounts of memory for corrupted files,
read the input in chunks limited to filesize or an arbitrary large
amount when that is not known (chosen to be 50M).
As far as I can tell the code should not change behaviour
depending on locale in any of these places.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Fix trac ticket #2300 because the duration of the segments
was computed using the timestamp of the last packet plus its
duration using the 1/90000 default time base instead of using
the chained muxer time base.
* qatar/master:
lavf: Add a fate test for the noproxy pattern matching
lavf: Handle the environment variable no_proxy more properly
Conflicts:
libavformat/Makefile
libavformat/internal.h
libavformat/tls.c
libavformat/utils.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This reverts commit 4a8fc1d83b.
The commit caused null pointer derefernces when using udp://
after i fixed that it caused ffmpeg to get stuck and remapped
arguments like ?ttl=255 -> ?ttl%3d255
I dont want to leave this broken thus temporary revert so we all
have some time to look at this without half the network protocols
being broken in the meantime
The handling of the environment variable no_proxy, present since
one of the initial commits (de6d9b6404), is inconsistent with
how many other applications and libraries interpret this
variable. Its bare presence does not indicate that the use of
proxies should be skipped, but it is some sort of pattern for
hosts that does not need using a proxy (e.g. for a local network).
As investigated by Rudolf Polzer, different libraries handle this
in different ways, some supporting IP address masks, some supporting
arbitrary globbing using *, some just checking that the pattern matches
the end of the hostname without regard for whether it actually is
the right domain or a domain that ends in the same string.
This simple logic should be pretty similar to the logic used by
lynx and curl.
Signed-off-by: Martin Storsjö <martin@martin.st>
Followup to http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/151321
patch by Reimar and Thomas Mundt fixes some AVC-Intra files from
different tickets.
It does not fix http://samples.ffmpeg.org/ffmpeg-
bugs/trac/ticket524/AVCI50.mov
Authors of this commit are: Reimar and Thomas Mundt
Patch and commit message mostly taken from ffmpeg-devel, mail by Carl
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
"que" sounds like a slang word to me. This commit renames a few
variables, fix the comments and the logging messages (sometimes along
with small other typo fixes).
it causes problems (incorrectly detect TS discontinuities)
with a brokan TS file (test-audio-broken.ts)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The values compared here can be more than INT64_MAX apart. Since the
difference is always positive, converting to uint64_t before subtracting
gives the correct result without overflows.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The previous code computes the offset by reversing the growth
of the allocated buffer size: it is complex and did lead to
inconsistencies when the size limit is reached.
Fix trac ticket #1991.
This fixes a regression where this count became 1 with
HPM-GC\ EXPORT\ FCP-1A-AVCI100-1080i25-001.mxf
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '5c7bf2dddee5bdfa247ff0d57cb8a37d19077f66':
lavf: move nuv fourcc audio tags from riff to nuv
lavf: add a common function for selecting a pcm codec from parameters
Conflicts:
libavformat/internal.h
libavformat/mov.c
libavformat/riff.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Scaling the denominator instead of the numerator if it is too large
loses precision. Fixes an assert caused by a negative frame duration in
the fuzzed sample nasa-8s2.ts_s202310.
CC: libav-stable@libav.org
I guess the user expects to see the stream with the highest bitrate, not with
the most frames, this is especially useful for multi bitrate streams.
This patch changes av_find_best_stream to select the stream based on a number
of conditions, the first condition has the highest priority, the last condition
has the lowest:
1) Select the stream with the highest FFMIN(5, codec_info_nb_frames) value
2) Select the stream with the highest bitrate
3) Select the stream with the highest codec_info_nb_frames
4) Select the first stream
Signed-off-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'cbcd497f384f0f8ef3f76f85b29b644b900d6b9f':
adxdec: use planar sample format
adpcmdec: use planar sample format for adpcm_thp
adpcmdec: use planar sample format for adpcm_ea_xas
adpcmdec: use planar sample format for adpcm_ea_r1/r2/r3
adpcmdec: use planar sample format for adpcm_xa
adpcmdec: use planar sample format for adpcm_ima_ws for vqa version 3
adpcmdec: use planar sample format for adpcm_4xm
adpcmdec: use planar sample format for adpcm_ima_wav
adpcmdec: use planar sample format for adpcm_ima_qt
pcmdec: use planar sample format for pcm_lxf
mace: use planar sample format
atrac1: use planar sample format
build: non-x86: Only compile mpegvideo optimizations when necessary
rtpdec_mpeg4: au_headers is a single array, simple av_free is enough
avcodec: free extended_data instead address of it
fate: Add tests of the ff_make_absolute_url function
url: Handle relative urls starting with two slashes
url: Handle relative urls being just a new query string
url: Don't treat slashes in query parameters as directory separators
Conflicts:
libavcodec/adxdec.c
libavcodec/mips/Makefile
libavcodec/pcm.c
libavcodec/utils.c
libavformat/Makefile
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Previously we had ignored the past dts and just filled in from the
point where we have had sufficient information.
This should fix Ticket1734
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '7751e4693dd10ec98c20fbd9887233b575034272':
ogg: check that the expected number of headers had been parsed
libx264: change default to closed gop to match x264cli
Use avcodec_free_frame() to free AVFrames.
lavf: use a malloced AVFrame in try_decode_frame().
lavc: add avcodec_free_frame().
lavc: ensure extended_data is set properly on decoding
lavc: initialize AVFrame.extended_data in avcodec_get_frame_defaults()
lavc: use av_mallocz to allocate AVFrames.
lavc: rename the argument of avcodec_alloc_frame/get_frame_defaults
Conflicts:
doc/APIchanges
doc/examples/decoding_encoding.c
libavcodec/utils.c
libavcodec/version.h
libavfilter/src_movie.c
libavformat/oggdec.c
libavformat/oggdec.h
libavformat/oggparsetheora.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
libx264: add forgotten ;
matroskadec: fix a sanity check.
matroskadec: only return corrupt packets that actually contain data
lavf: zero data/size of the packet passed to read_packet().
ARM: use 2-operand syntax for ADD Rd, PC in Apple PIC code
ARM: align PIC offset pools to 4 bytes
ARM: swap source operands in some add instructions
configure: update tms470 detection for latest version
lavf probe: prevent codec probe with no data at all seen
motion_est: fix use of inline on extern functions
Conflicts:
libavcodec/motion_est_template.c
libavformat/matroskadec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This occurs with fuzzed mpeg-ts files. set_codec_from_probe_data() is
called with a zeroed AVProbeData since no packet made through for
specific stream.
* commit '3f7fd59d151a2773f0e2e93e56b6b13ec6e5334b':
avformat: fix typo in avformat_close_input
mp3enc: write Xing TOC
mp3enc: support MPEG-2 and MPEG-2.5 in Xing header.
mp3enc: downgrade some errors in writing Xing frame to warnings
lavf: flush the output AVIOContext in av_write_trailer().
lavf: cosmetics, reformat av_write_trailer().
avio: flush the internal buffer in avio_close()
Enhance doc on asyncts audiofilter
cmdutils: avoid setting data pointers to invalid values in alloc_buffer()
libavcodec: remove av_destruct_packet_nofree()
Conflicts:
libavcodec/avpacket.c
libavformat/mp3enc.c
libavformat/nutenc.c
libavformat/utils.c
libavformat/version.h
tests/ref/lavf/voc
tests/ref/lavf/voc_s16
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is consistent with stdio and is what we want to do in all cases.
Fixes a bug in the voc muxer which didn't flush in write_trailer()
previously. This is the cause of the change in the test results.
Make internal small_strptime() function public, and use it in place of
strptime().
This allows to avoid a dependency on strptime() on systems which do not
support it.
In particular, fix trac ticket #992.
This improves dts validity checks and consequently fps detection of files with invalid dts
Fixes Ticket1681
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The resolution is in the packets, so decoding must happen.
Since most other formats do not set the dimension, make it
a special case for PGS. If other codecs were to have the
same requirement, using a CODEC_CAP would be preferred.
This also changes behavior as the descriptor table is more complete than
the switch/case it replaces. As well as considering all non video as
intra only
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
lavf: Detect discontinuities in timestamps for framerate/analyzeduration calculation
lavf: Initialize the stream info timestamps in avformat_new_stream
id3v2: Match PIC mimetype/format case-insensitively
configure: Rename check_asm() to more fitting check_inline_asm()
fate: Only test enabled filters
avresample: De-doxygenize some comments where Doxygen is not appropriate
rtmp: split chunk_size var into in_chunk_size and out_chunk_size
rtmp: Factorize the code by adding find_tracked_method
Conflicts:
configure
Merged-by: Michael Niedermayer <michaelni@gmx.at>
These are normally initialized to AV_NOPTS_VALUE at the start
of avformat_find_stream_info, but if a new stream is found while
this function is running (e.g. like in mpegts), the newly added
AVStreams didn't have these values properly initalized, leading
to avformat_find_stream_info terminating too soon (when the
first timestamps are far from 0).
Signed-off-by: Martin Storsjö <martin@martin.st>
This adds a function to retrieve the number of entries in a
dictionary and updates the places directly accessing what should
be an opaque struct to use this new function instead.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This is limited to the chars that arent filtered by av_log() already
we might filter more aggressively if theres some case where this becomes
needed.
Fixes Ticket1181
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
At this place, the normal way of initializing a struct works
fine, there's no need for a struct literal.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
flvdec: remove spurious use of stream id
lavf: deprecate r_frame_rate.
lavf: round estimated average fps to a "standard" fps.
Conflicts:
ffmpeg.c
ffprobe.c
libavformat/avformat.h
libavformat/electronicarts.c
libavformat/flvdec.c
libavformat/rawdec.c
libavformat/utils.c
tests/ref/fate/iv8-demux
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'fe1c1198e670242f3cf9e3e1eef27cff77f3ee23':
lavf: use dts difference instead of AVPacket.duration in find_stream_info()
avf: introduce nobuffer option
fate: make yadif tests consistent across systems
vf_hqdn3d: support 9 and 10bit colordepth
vf_hqdn3d: reduce intermediate precision
vf_hqdn3d: simplify and optimize
factor identical ff_inplace_start_frame out of two filters
vf_hqdn3d: cosmetics
avprobe/avconv: fix tentative declaration compile errors on MSVS.
Conflicts:
doc/APIchanges
ffmpeg.c
ffprobe.c
libavformat/avformat.h
libavformat/options_table.h
libavformat/utils.c
libavformat/version.h
tests/fate/filter.mak
tests/ref/fate/filter-yadif-mode0
tests/ref/fate/filter-yadif-mode1
Merged-by: Michael Niedermayer <michaelni@gmx.at>
According to its description, it is supposed to be the LCM of all the
frame durations. The usability of such a thing is vanishingly small,
especially since we cannot determine it with any amount of reliability.
Therefore get rid of it after the next bump.
Replace it with the average framerate where it makes sense.
FATE results for the wtv and xmv demux tests change. In the wtv case
this is caused by the file being corrupted (or possibly badly cut) and
containing invalid timestamps. This results in lavf estimating the
framerate wrong and making up wrong frame durations.
In the xmv case the file contains pts jumps, so again the estimated
framerate is far from anything sane and lavf again makes up different
frame durations.
In some other tests lavf starts making up frame durations from different
frame.
AVPacket.duration is mostly made up and thus completely useless, this is
especially true for video streams.
Therefore use dts difference for framerate estimation and
the max_analyze_duration check.
The asyncts test now needs -analyzeduration, because the default is 5
seconds and the audio stream in the sample appears at ~10 seconds.
Useful in cases where a significant analyzeduration is
still needed, while minimizing buffering before output.
An example is processing low-latency streams where all
media types won't necessarily come in if the
analyzeduration is small.
Additional changes by Josh Allmann <joshua.allmann@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master: (35 commits)
h264_idct_10bit: port x86 assembly to cpuflags.
x86inc: clip num_args to 7 on x86-32.
x86inc: sync to latest version from x264.
fft: rename "z" to "zc" to prevent name collision.
wv: return meaningful error codes.
wv: return AVERROR_EOF on EOF, not EIO.
mp3dec: forward errors for av_get_packet().
mp3dec: remove a pointless local variable.
mp3dec: remove commented out cruft.
lavfi: bump minor to mark stabilizing the ABI.
FATE: add tests for yadif.
FATE: add a test for delogo video filter.
FATE: add a test for amix audio filter.
audiogen: allow specifying random seed as a commandline parameter.
vc1dec: Override invalid macroblock quantizer
vc1: avoid reading beyond the last line in vc1_draw_sprites()
vc1dec: check that coded slice positions and interlacing match.
vc1dec: Do not ignore ff_vc1_parse_frame_header_adv return value
configure: Move parts that should not be user-selectable to CONFIG_EXTRA
lavf: remove commented out cruft in avformat_find_stream_info()
...
Conflicts:
Makefile
configure
libavcodec/vc1dec.c
libavcodec/x86/h264_deblock.asm
libavcodec/x86/h264_deblock_10bit.asm
libavcodec/x86/h264dsp_mmx.c
libavfilter/version.h
libavformat/mp3dec.c
libavformat/utils.c
libavformat/wv.c
libavutil/x86/x86inc.asm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
By moving it to a later point relative and unknown timestamps
are more likely to have been corrected
similar patch reviewed-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Conflicts:
libavformat/utils.c
If skip_samples is set and timestamps are synthesized using durations,
make them start at -skip_samples (rescaled) instead of 0,
so that the timestamp of the first undiscarded sample is 0.
This adds the minimum delay needed with the current decoder to
recognize the reorder buffer size for the reference bitstreams.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
To ensure the full range of values is still used, also adjust all uses of this function to loop from 0
instead of 1. This way only 60.00 is added and nothing lost.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
MS Screen 1 decoder
aacdec: Fix popping channel layouts.
av_gettime: support Win32 without gettimeofday()
Use av_gettime() in various places
Move av_gettime() to libavutil
dct-test: use emms_c() from libavutil instead of duplicating it
mov: fix operator precedence bug
mathematics.h: remove a couple of math defines
Remove unnecessary inclusions of [sys/]time.h
lavf: remove unnecessary inclusions of unistd.h
bfin: libswscale: add const where appropriate to fix warnings
bfin: libswscale: remove unnecessary #includes
udp: Properly check for invalid sockets
tcp: Check the return value from getsockopt
network: Use av_strerror for getting error messages
udp: Properly print error from getnameinfo
mmst: Use AVUNERROR() to convert error codes to the right range for strerror
network: Pass pointers of the right type to get/setsockopt/ioctlsocket on windows
rtmp: Reduce the number of idle posts sent by sleeping 50ms
Conflicts:
Changelog
configure
libavcodec/aacdec.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/dct-test.c
libavcodec/version.h
libavformat/riff.c
libavformat/udp.c
libavutil/Makefile
libswscale/bfin/yuv2rgb_bfin.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
flacdec: read attached pictures.
lavf: don't segfault when a NULL filename is passed to avformat_open_input()
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This can easily happen when the caller is using a custom AVIOContext.
Behave as if the filename was an empty string in this case.
CC: libav-stable@libav.org
This fixes an issue with a crazy data track starting with a large
negative timestamp.
It could as well be solved in all user apps, but this is looking
attractively simpler ...
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
vorbis: Validate that the floor 1 X values contain no duplicates.
avprobe: Identify codec probe failures rather than calling them unsupported codecs.
avformat: Probe codecs at score 0 on buffer exhaustion conditions.
avformat: Factorize codec probing.
Indeo Audio decoder
imc: make IMDCT support stereo output
imc: move channel-specific data into separate context
lavfi: remove request/poll and drawing functions from public API on next bump
lavfi: make avfilter_insert_pad and pals private on next bump.
lavfi: make formats API private on next bump.
avplay: use buffersrc instead of custom input filter.
avtools: move buffer management code from avconv to cmdutils.
avconv: don't use InputStream in the buffer management code.
avconv: fix exiting when max frames is reached.
mpc8: fix maximum bands handling
aacdec: Turn PS off when switching to stereo and turn it to implicit when switching to mono.
Conflicts:
Changelog
cmdutils.h
ffmpeg.c
ffplay.c
ffprobe.c
libavcodec/avcodec.h
libavcodec/mpc8.c
libavcodec/v210dec.h
libavcodec/version.h
libavcodec/vorbisdec.c
libavfilter/avfilter.c
libavfilter/avfilter.h
libavfilter/buffersrc.c
libavfilter/formats.c
libavfilter/src_movie.c
libavfilter/vf_aspect.c
libavfilter/vf_blackframe.c
libavfilter/vf_boxblur.c
libavfilter/vf_crop.c
libavfilter/vf_cropdetect.c
libavfilter/vf_delogo.c
libavfilter/vf_drawbox.c
libavfilter/vf_drawtext.c
libavfilter/vf_fade.c
libavfilter/vf_fifo.c
libavfilter/vf_format.c
libavfilter/vf_frei0r.c
libavfilter/vf_gradfun.c
libavfilter/vf_hflip.c
libavfilter/vf_hqdn3d.c
libavfilter/vf_libopencv.c
libavfilter/vf_lut.c
libavfilter/vf_overlay.c
libavfilter/vf_pad.c
libavfilter/vf_scale.c
libavfilter/vf_select.c
libavfilter/vf_showinfo.c
libavfilter/vf_transpose.c
libavfilter/vf_unsharp.c
libavfilter/vf_yadif.c
libavfilter/vsrc_color.c
libavfilter/vsrc_testsrc.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
indeo: Make ivi_calc_band_checksum() static, it is only used in one file.
indeo: Drop unused debug function ivi_check_band().
avcodec/utils: cast a function argument to shut up a compiler warning
truemotion1: remove disabled code
fix typo in comment
fate: fix dependencies for non-SAMPLES avconv tests
indeo: check for invalid motion vectors
indeo: check that band output buffer exists
indeo: clear allocated band buffers
indeo: track tile macroblock size
indeo: check custom Huffman tables for errors
factor out common decoding code for Indeo 4 and Indeo 5
mp3: fix start band index for block type 2 in 8kHz audio
lavf: change some (de)muxer names to lowercase
lavf: make output format matching case insensitive
Conflicts:
libavcodec/indeo4.c
libavcodec/indeo5.c
libavcodec/ivi_common.c
libavcodec/utils.c
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Use codec aspect ratio for frame aspect ratio if AVFrame is NULL.
Signed-off-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fate: Work around non-standard wc implementations at more places
fate: work around non-standard wc implementations
x86: rv40: Mark rv40_weight functions as MMX2; they use MMX2 instructions.
ac3dsp: simplify x86 versions of ac3_max_msb_abs_int16
fate: use standard diff options
tta: Fix comment about channel number; TTA supports >2 channels.
avfilter: Move ff_get_ref_perms_string() to where it is used.
build: Add 'check' target to run all compile and test targets.
indeo3: validate new frame size before resetting decoder
indeo3: when freeing buffers, set pointers referencing them to NULL as well
indeo3: initialise pixel planes on allocation
indeo3: ensure that decoded cell data is in 7-bit range as presumed by decoder
fate: rename psx-str-v3-mdec to mdec-v3
fate: convert psx-str to a demuxer test
lavf: add mdec to is_intra_only() list
Conflicts:
doc/developer.texi
libavcodec/indeo3.c
libavfilter/video.c
libavformat/utils.c
tests/fate/demux.mak
tests/fate/video.mak
tests/lavf-regression.sh
tests/ref/vsynth1/cljr
tests/ref/vsynth1/ffvhuff
tests/ref/vsynth2/cljr
tests/ref/vsynth2/ffvhuff
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Guesses the sample aspect ratio of a frame, based on both the stream and the
frame aspect ratio.
Since the frame aspect ratio is set by the codec but the stream aspect ratio
is set by the demuxer, these two may not be equal. This function tries to
return the value that you should use if you would like to display the frame.
Basic logic is to use the stream aspect ratio if it is set to something sane
otherwise use the frame aspect ratio. This way a container setting, which is
usually easy to modify can override the coded value in the frames.
Signed-off-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
configure: add POWER[5-7] support
arm: intreadwrite: revert 16-bit load asm to old version for gcc < 4.6
vqavideo: return error if image size is not a multiple of block size
cosmetics: indentation
avformat: only fill-in interpolated timestamps if duration is non-zero
avformat: remove a workaround for broken timestamps
Conflicts:
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Manually remove that flag again for formats that read an arbitrary
amount of data and thus truncation is not an error.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
this hunk was merged in 8b97ae64 and cbf767a8 although the check was there a
few lines above since cdced09e. I removed the first check to reduce the differences
to libav.
Signed-off-by: Jean First <jeanfirst@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This replaces the matroskadec one with the same name.
The advantage is not only easier reuse in other demuxers
but also that we can make the decisions after the parser.
This fixes seeking in files that mark the keyframes incorrectly,
for example the file in track ticket #1003.
The matroska variable is still kept to be able to complain
about such broken files.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
vsrc_buffer: fix check from 7ae7c41.
libxvid: Reorder functions to avoid forward declarations; make functions static.
libxvid: drop some pointless dead code
wmal: vertical alignment cosmetics
wmal: Warn about missing bitstream splicing feature and ask for sample.
wmal: Skip seekable_frame_in_packet.
wmal: Drop unused variable num_possible_block_size.
avfiltergraph: make the AVFilterInOut alloc/free API public
graphparser: allow specifying sws flags in the graph description.
graphparser: fix the order of connecting unlabeled links.
graphparser: add avfilter_graph_parse2().
vsrc_buffer: allow using a NULL buffer to signal EOF.
swscale: handle last pixel if lines have an odd width.
qdm2: fix a dubious pointer cast
WMAL: Do not try to read rawpcm coefficients if bits is invalid
mov: Fix detecting there is no sync sample.
tiffdec: K&R cosmetics
avf: has_duration does not check the global one
dsputil: fix optimized emu_edge function on Win64.
Conflicts:
doc/APIchanges
libavcodec/libxvid_rc.c
libavcodec/libxvidff.c
libavcodec/tiff.c
libavcodec/wmalosslessdec.c
libavfilter/avfiltergraph.h
libavfilter/graphparser.c
libavfilter/version.h
libavfilter/vsrc_buffer.c
libswscale/output.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Otherwise for muxers like e.g. latmenc that never call
avio_flush (and do not have a write_trailer function)
a part of the data will always be missing.
Also update references for the voc muxer, which was also
buggy before and did not write out all data.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
rtpdec_asf: Set the no_resync_search option for the chained asf demuxer
asfdec: Add an option for not searching for the packet markers
cosmetics: Clean up the tiffenc pix_fmts declaration to match the style of others
cosmetics: Align codec declarations
cosmetics: Convert mimic.c to utf-8
avconv: remove an unused function parameter.
avconv: remove now pointless variables.
avconv: drop support for building without libavfilter.
nellymoserenc: fix crash due to memsetting the wrong area.
libavformat: Only require first packet to be known for audio/video streams
avplay: Don't try to scale timestamps if the tb isn't set
Conflicts:
Changelog
configure
ffmpeg.c
libavcodec/aacenc.c
libavcodec/bmpenc.c
libavcodec/dnxhddec.c
libavcodec/dnxhdenc.c
libavcodec/ffv1.c
libavcodec/flacenc.c
libavcodec/fraps.c
libavcodec/huffyuv.c
libavcodec/libopenjpegdec.c
libavcodec/mpeg12enc.c
libavcodec/mpeg4videodec.c
libavcodec/pamenc.c
libavcodec/pgssubdec.c
libavcodec/pngenc.c
libavcodec/qtrleenc.c
libavcodec/rawdec.c
libavcodec/sgienc.c
libavcodec/tiffenc.c
libavcodec/v210dec.c
libavcodec/wmv2dec.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It can take a long time before subtitles or data streams show up,
so we shouldn't wait for those before assuming we have all info
for streams.
Signed-off-by: Martin Storsjö <martin@martin.st>
To reproduce the problem, using ffprobe:
./ffprobe -show_packets -print_format compact -fflags +genpts -i
fate_samples/mxf/C0023S01.mxf
You will notice that the last video frame does not have it's PTS being
set, even with using genpts.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Allows avoiding the buffer when using avio read, write and seek functions.
When using the ffmpeg executable -avioflags direct can be used to enable
this mode for input files, but has no effect on output files.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
asf: only set index_read if the index contained entries.
cabac: add overread protection to BRANCHLESS_GET_CABAC().
cabac: increment jump locations by one in callers of BRANCHLESS_GET_CABAC().
cabac: remove unused argument from BRANCHLESS_GET_CABAC_UPDATE().
cabac: use struct+offset instead of memory operand in BRANCHLESS_GET_CABAC().
h264: add overread protection to get_cabac_bypass_sign_x86().
h264: reindent get_cabac_bypass_sign_x86().
h264: use struct offsets in get_cabac_bypass_sign_x86().
h264: fix overreads in cabac reader.
wmall: fix seeking.
lagarith: fix buffer overreads.
dvdec: drop unnecessary dv_tablegen.h #include
build: fix doc generation errors in parallel builds
Replace memset(0) by zero initializations.
faandct: Remove FAAN_POSTSCALE define and related code.
dvenc: print allowed profiles if the video doesn't conform to any of them.
avcodec_encode_{audio,video}: only reallocate output packet when it has non-zero size.
FATE: add a test for vp8 with changing frame size.
fate: add kgv1 fate test.
oggdec: calculate correct timestamps in Ogg/FLAC
Conflicts:
libavcodec/4xm.c
libavcodec/cook.c
libavcodec/dvdata.c
libavcodec/dvdsubdec.c
libavcodec/lagarith.c
libavcodec/lagarithrac.c
libavcodec/utils.c
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
make av_interleaved_write_frame() flush packets when pkt is NULL
mpegts: Fix dead error checks
vc1: Do not read from array if index is invalid.
targa: convert to bytestream2.
rv34: set mb_num_left to 0 after finishing a frame
Conflicts:
libavcodec/targa.c
libavcodec/vc1data.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This patch allows the user to force flushing of all queued packets
by calling av_interleaved_write_frame() with pkt set to NULL.
Signed-off-by: Jindrich Makovicka <jindrich.makovicka@nangu.tv>
Signed-off-by: Martin Storsjö <martin@martin.st>
Currently, the duration of those packets is just discarded
when enabling parsing, thus the output of the Metal Gear Solid
demuxer breaks completely when just setting AVSTREAM_PARSE_HEADERS.
The result will not be correct if a parser creates a delay even
with PARSER_FLAG_COMPLETE_FRAMES and there might be other cases
where it does not work correct, but just discarding them as it
is done currently seems worse.
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>
* qatar/master: (35 commits)
fix space type in Changelog
ZeroCodec Decoder
RealAudio Lossless decoder
rtpenc: Use AVFormatContext.packet_size instead of a private option
url: Document the expected behaviour of url_read
libavformat: Use AVFormatContext.probesize in init_input
docs: Fix a stray reference to tags in the generic doxy on dicts
cosmetics: Align some AVInput/OutputFormat declarations
zmbv: check decompress result
zmbv: correct indentation
adpcm: convert adpcm_thp to bytestream2.
adpcm: convert adpcm_yamaha to bytestream2.
adpcm: convert adpcm_swf to bytestream2.
adpcm: convert adpcm_sbpro to bytestream2.
adpcm: convert adpcm_ct to bytestream2.
adpcm: convert adpcm_ima_amv/smjpeg to bytestream2.
adpcm: convert adpcm_ea_xas to bytestream2.
adpcm: convert adpcm_ea_r1/2/3 to bytestream2.
adpcm: convert ea_maxis_xa to bytestream2.
adpcm: convert adpcm_ea to bytestream2.
...
Conflicts:
Changelog
libavcodec/Makefile
libavcodec/adpcm.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/version.h
libavcodec/zerocodec.c
libavcodec/zmbv.c
libavformat/riff.c
libavformat/url.h
tests/ref/fate/truemotion1-15
tests/ref/fate/truemotion1-24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This was forgotten in the transition from av_open_input_file to
avformat_open_input, see 603b8bc2a1.
This doesn't change anything for the default case where the
option isn't set, since PROBE_BUF_MAX is 1048576 (which was
used as max probe size earlier) while the default value for
the probesize option is 5000000, which for the probe function
is clipped to PROBE_BUF_MAX anyway.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
fate: make compare() function compatible with POSIX bc
Update Janne's email address.
APIchanges: Replace Subversion revision numbers by Git hashes.
bytestream: Eliminate one level of pointless macro indirection.
xwd: convert to bytestream2.
vqavideo: port to bytestream2 API
Read preset files with suffix .avpreset
prores: allow user to set fixed quantiser
lavf: remove some disabled code.
lavf: only set average frame rate for video.
lavf: remove a pointless check.
avcodec: add XBM encoder
Conflicts:
Changelog
cmdutils.c
cmdutils.h
doc/APIchanges
libavcodec/Makefile
libavcodec/avcodec.h
libavcodec/version.h
libavcodec/vqavideo.c
libavformat/img2enc.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
dxa: remove useless code
lavf: don't select an attached picture as default stream for seeking.
avconv: remove pointless checks.
avconv: check for get_filtered_frame() failure.
avconv: remove a pointless check.
swscale: convert hscale() to use named arguments.
x86inc: add *mp named argument support to DEFINE_ARGS.
swscale: convert hscale to cpuflags().
Conflicts:
ffmpeg.c
libswscale/x86/scale.asm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
With this we can always know if a timestamp is based on added durations
from an unknown origin or if it is based on a correct timestamp (and possibly
added durations)
This should fix some bugs where this distinction was mixed up.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (31 commits)
cdxl demux: do not create packets with uninitialized data at EOF.
Replace computations of remaining bits with calls to get_bits_left().
amrnb/amrwb: Remove get_bits usage.
cosmetics: reindent
avformat: do not require a pixel/sample format if there is no decoder
avformat: do not fill-in audio packet duration in compute_pkt_fields()
lavf: Use av_get_audio_frame_duration() in get_audio_frame_size()
dca_parser: parse the sample rate and frame durations
libspeexdec: do not set AVCodecContext.frame_size
libopencore-amr: do not set AVCodecContext.frame_size
alsdec: do not set AVCodecContext.frame_size
siff: do not set AVCodecContext.frame_size
amr demuxer: do not set AVCodecContext.frame_size.
aiffdec: do not set AVCodecContext.frame_size
mov: do not set AVCodecContext.frame_size
ape: do not set AVCodecContext.frame_size.
rdt: remove workaround for infinite loop with aac
avformat: do not require frame_size in avformat_find_stream_info() for CELT
avformat: do not require frame_size in avformat_find_stream_info() for MP1/2/3
avformat: do not require frame_size in avformat_find_stream_info() for AAC
...
Conflicts:
doc/APIchanges
libavcodec/Makefile
libavcodec/avcodec.h
libavcodec/h264.c
libavcodec/h264_ps.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/x86/dsputil_mmx.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
All colorspaces are supported.
Renamed libutvideo.cpp to libutvideodec.cpp.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Also, do not keep trying to find and open a decoder in try_decode_frame() if
we already tried and failed once.
Fixes always searching until max_analyze_duration in
avformat_find_stream_info() when demuxing codecs without a decoder.
Also, do not give AVCodecContext.frame_size priority for muxing.
Updated 2 FATE references:
dxa-feeble - adds 1 audio frame that is still within 2 seconds as specified
by -t 2 in the FATE test
wmv8-drm-nodec - durations are not needed. previously they were estimated
using the packet size and average bit rate.
Split off packet parsing into a separate function. Parse full packets at
once and store them in a queue, eliminating the need for tracking
parsing state in AVStream.
The horrible unreadable loop in read_frame_internal() now isn't weirdly
ordered and doesn't contain evil gotos, so it should be much easier to
understand.
compute_pkt_fields() now invents slightly different timestamps for two
raw vc1 tests, due to has_b_frames being set a bit later. They shouldn't
be more wrong (or right) than previous ones.
Make packet buffer a parameter, don't hardcode it to be
AVFormatContext.packet_buffer.
Also move the function higher in the file, since it will be called from
read_frame_internal().
This fixes issues when the bitrate is variable or inaccurate but the
frame size has not been determined yet.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
tiertexseq: set correct block_align for audio
tiertexseq: set audio stream start time to 0
voc/avs: Do not change the sample rate mid-stream.
segafilm: use the sample rate as the time base for audio streams
ea: fix audio pts
psx-str: fix audio pts
vqf: set packet duration
tta demuxer: set packet duration
mpegaudio_parser: do not ignore information from the first parsed frame
mpegaudio_parser: be less picky about the start position
thp: set audio packet durations
avcodec: add a Vorbis parser to get packet duration
vorbisdec: read the previous window flag for long windows
lavc: free the output packet when encoding failed or produced no output.
lavc: preserve avpkt->destruct in ff_alloc_packet().
lavc: clarify the meaning of AVCodecContext.frame_number.
mpegts: Pad the packet buffer in handle_packet().
mpegts: Do not call read_sl_header() when no bytes remain in the buffer.
Conflicts:
libavcodec/mpegaudio_parser.c
libavcodec/version.h
libavformat/mpegts.c
tests/ref/fate/pva-demux
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes duplicate timestamps on mp2 in ts with non seekable input.
It also fixed the fate pva demux timestamps.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (58 commits)
amrnbdec: check frame size before decoding.
cscd: use negative error values to indicate decode_init() failures.
h264: prevent overreads in intra PCM decoding.
FATE: do not decode audio in the nuv test.
dxa: set audio stream time base using the sample rate
psx-str: do not allow seeking by bytes
asfdec: Do not set AVCodecContext.frame_size
vqf: set packet parameters after av_new_packet()
mpegaudiodec: use DSPUtil.butterflies_float().
FATE: add mp3 test for sample that exhibited false overreads
fate: add cdxl test for bit line plane arrangement
vmnc: return error on decode_init() failure.
libvorbis: add/update error messages
libvorbis: use AVFifoBuffer for output packet buffer
libvorbis: remove unneeded e_o_s check
libvorbis: check return values for functions that can return errors
libvorbis: use float input instead of s16
libvorbis: do not flush libvorbis analysis if dsp state was not initialized
libvorbis: use VBR by default, with default quality of 3
libvorbis: fix use of minrate/maxrate AVOptions
...
Conflicts:
Changelog
doc/APIchanges
libavcodec/avcodec.h
libavcodec/dpxenc.c
libavcodec/libvorbis.c
libavcodec/vmnc.c
libavformat/asfdec.c
libavformat/id3v2enc.c
libavformat/internal.h
libavformat/mp3enc.c
libavformat/utils.c
libavformat/version.h
libswscale/utils.c
tests/fate/video.mak
tests/ref/fate/nuv
tests/ref/fate/prores-alpha
tests/ref/lavf/ffm
tests/ref/vsynth1/prores
tests/ref/vsynth2/prores
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
docs: use -bsf:[vas] instead of -[vas]bsf.
mpegaudiodec: Prevent premature clipping of mp3 input buffer.
lavf: move the packet keyframe setting code.
oggenc: free comment header for all codecs
lcl: error out if uncompressed input buffer is smaller than framesize.
mjpeg: abort decoding if packet is too large.
golomb: use HAVE_BITS_REMAINING() macro to prevent infloop on EOF.
get_bits: add HAVE_BITS_REMAINING macro.
lavf/output-example: use new audio encoding API correctly.
lavf/output-example: more proper usage of the new API.
tiff: Prevent overreads in the type_sizes array.
tiff: Make the TIFF_LONG and TIFF_SHORT types unsigned.
apetag: do not leak memory if avio_read() fails
apetag: propagate errors.
SBR DSP x86: implement SSE sbr_hf_g_filt
SBR DSP x86: implement SSE sbr_sum_square_sse
SBR DSP: use intptr_t for the ixh parameter.
Conflicts:
doc/bitstream_filters.texi
doc/examples/muxing.c
doc/ffmpeg.texi
libavcodec/golomb.h
libavcodec/x86/Makefile
libavformat/oggenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
compute_pkt_fields() is for unreliable estimates or guessing. The
keyframe information from the parser is (at least in theory) reliable,
so it should be used even when the other guessing is disabled with the
AVFMT_FLAG_NOFILLIN flag.
Therefore, move setting the packet keyframe flag based on parser
information from compute_pkt_fields() to read_frame_internal().
It is not supposed to be done outside lavc.
This is basically a revert of 818062f2f3.
It is unclear what issue this was supposed to fix, if it reappears again
it will have to be fixed in a more proper place.
The wtv-demux test change is because the sample starts with a B-frame.
* qatar/master: (36 commits)
adpcmenc: Use correct frame_size for Yamaha ADPCM.
avcodec: add ff_samples_to_time_base() convenience function to internal.h
adx parser: set duration
mlp parser: set duration instead of frame_size
gsm parser: set duration
mpegaudio parser: set duration instead of frame_size
(e)ac3 parser: set duration instead of frame_size
flac parser: set duration instead of frame_size
avcodec: add duration field to AVCodecParserContext
avutil: add av_rescale_q_rnd() to allow different rounding
pnmdec: remove useless .pix_fmts
libmp3lame: support float and s32 sample formats
libmp3lame: renaming, rearrangement, alignment, and comments
libmp3lame: use the LAME default bit rate
libmp3lame: use avpriv_mpegaudio_decode_header() for output frame parsing
libmp3lame: cosmetics: remove some pointless comments
libmp3lame: convert some debugging code to av_dlog()
libmp3lame: remove outdated comment.
libmp3lame: do not set coded_frame->key_frame.
libmp3lame: improve error handling in MP3lame_encode_init()
...
Conflicts:
doc/APIchanges
libavcodec/libmp3lame.c
libavcodec/pcxenc.c
libavcodec/pnmdec.c
libavcodec/pnmenc.c
libavcodec/sgienc.c
libavcodec/utils.c
libavformat/hls.c
libavutil/avutil.h
libswscale/x86/swscale_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
