avcodec/tiff: Check frame parameters before blit for DNG
Fixes: out of array access Fixes: 23888/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TIFF_fuzzer-6021365974171648.fuzz Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
avformat/smacker: Don't read only one byte at a time
Instead use ffio_read_size to read data into a buffer. Also check that
the desired size was actually successfully read and combine the check
with the check for reading the extradata.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
A Smacker file can contain up to seven audio tracks. Up until now,
the pts for the i. audio packet contained in a Smacker frame was
simply the end timestamp of the last i. audio packet contained in
an earlier Smacker frame.
The problem with this is that a Smacker stream need not contain data in
every Smacker frame and so the current i. audio packet present may come
from a different underlying stream than the last i. audio packet
contained in an earlier frame.
The sample hypnotix.smk* exhibits this. It has three audio tracks and
the first of the three has a longer first packet, so that the audio for
the first track is contained in only 235 packets contained in the first
235 Smacker frames; the end timestamp of this track is 166696 (about 7.56s
at a timebase of 1/22050); the other two audio tracks both have 253 packets
contained in the first 253 Smacker frames. Up until now, the 236th
packet of the second track being the first audio packet in the 236th
Smacker frame would get the end timestamp of the last first audio packet
from the last Smacker frame containing a first audio packet and said
last audio packet is the first audio packet from the 235th Smacker frame
from the first audio track, so that the timestamp is 166696. In contrast,
the 236th packet from the third track (whose packets contain the same number
of samples as the packets from the second track) has a timestamp of
156116 (because its timestamp is derived from the end timestamp of the
235th packet of the second audio track). In the end, the second track
ended up being 177360/22050 s = 8.044s long; in contrast, the third
track was 166780/22050 s = 7.56s long which also coincided with the
video.
This commit fixes this by not using timestamps from other tracks for
a packet's pts.
avformat/smacker: Stop caching and copying audio frames
The layout of a Smacker frame is as follows: For some frames, the
beginning of the frame contained a palette for the video stream; then
there are potentially several audio frames, followed by the data for the
video stream.
The Smacker demuxer used to read the palette, then cache every audio frame
into a buffer (that gets reallocated to the desired size every time a
frame is read into this buffer), then read and return the video frame
(together with the palette). The cached audio frames are then returned
by copying the data into freshly allocated buffers; if there are none
left, the next frame is read.
This commit changes this: At the beginning of a frame, the palette is
read and cached as now. But audio frames are no longer cached at all;
they are returned immediately. This gets rid of copying and also allows
to remove the code for the buffer-to-AVStream correspondence.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
avformat/smacker: Avoid potential inifinite loop on error
When reading a new frame, the Smacker demuxer seeks to the next frame
position where it excepts the next frame; then it (potentially) reads
the palette, the audio packets associated with the frame and finally the
actual video frame. It is only at the end that the frame counter as well
as the position where the next frame is expected get updated.
This has a downside: If e.g. invalid data is encountered when reading
the palette, the demuxer returns immediately (with an error) and if the
caller calls av_read_frame again, the demuxer seeks to the position where
it already was, reads the very same palette data again and therefore will
return an error again. If the caller calls av_read_frame repeatedly
(say, until a packet is received or until EOF), this meight become an
infinite loop.
This could also happen if e.g. the size of one of the audio frames was
invalid or if the frame size was gigantic.
This commit changes this by skipping a frame if it turns out to be
invalid or an error happens otherwise. This ensures that EOF will be
returned eventually in the above scenario.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The Smacker demuxer buffers audio packets before it outputs them, but it
increments the counter of buffered packets prematurely: If allocating
the audio buffer fails, an error (most likely AVERROR(ENOMEM)) is returned.
If the caller decides to call av_read_frame() again, the next call will
take the codepath for returning already buffered audio packets and it
will fail (because the buffer that ought to be allocated isn't) without
decrementing the number of supposedly buffered audio packets (it doesn't
matter whether there would be enough memory available in subsequent calls).
Depending on the caller's behaviour this is potentially an infinite loop.
This commit fixes this by only incrementing the number of buffered audio
packets after having successfully read them and unconditionally reducing
said number when outputting one of them. It also changes the semantics
of the curstream variable: It is now the number of currently buffered
audio packets whereas it used to be the index of the last audio stream
to be read. (Index refers to the index in the array of buffers, not to
the stream index.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Lu Jiao [Tue, 16 Jun 2020 14:10:10 +0000 (22:10 +0800)]
avcodec/amfenc_hevc: set default gops_per_idr 1
Previously gops_per_idr default 60 would make amd gpu encoding hevc
has erratic seek bar behaviour as descripted in this ticket:
https://trac.ffmpeg.org/ticket/7272
This bad default is already fixed in newer AMF headers:
https://github.com/GPUOpen-LibrariesAndSDKs/AMF/blob/master/amf/public/include/components/VideoEncoderHEVC.h
Derek Buitenhuis [Fri, 26 Jun 2020 13:08:37 +0000 (14:08 +0100)]
ffprobe: Allow unknown format private AVOptions
This useful, because by ffprobe's very nature, you use it to probe
a file and find out what it is. Requiring every format private option
to be known to the demuxer forces one to run ffprobe twice, if one
wants to use ffprobe in a generic way.
For example, say one wants to probe all user-uploaded files, while
also ignoring edit lists for any MP4s that are uploaded. Currently,
you'd have to run ffprobe twice: once to identify the format, and
once again to actually probe the metadata you want. After this
patch, you could set -ignore_editlist 1 on every call and only
probe once.
Fixes: signed integer overflow: 6500736 * 473 cannot be represented in type 'int' Fixes: 23259/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MV30_fuzzer-5179394271477760 Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
avcodec/hcadec: Check total_band_count against imdct_in size
Fixes: index 128 out of bounds for type 'float [128]' Fixes: 23465/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_HCA_fuzzer-5089866596745216 Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
avcodec/tiff: Do not overrun the array ends in dng_blit()
Fixes: out of array access Fixes: 23589/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TIFF_fuzzer-5110559589793792.fuzz Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
avcodec/dstdec: Replace AC overread check by sample rate check
Real files do skip coding 0 bits at the end, thus this kind of check
does not work reliable.
Fixes: Ticket 8770 Fixes: dst-256fs44-6ch-refdstencoder.dff
The samplerate is specified in ISO/IEC 14496-3:2005(E) as one of 3 fixed
values, this also can be used to limit the duration and avoid the timeout
Limin Wang [Wed, 1 Jul 2020 12:57:58 +0000 (20:57 +0800)]
FATE: fix colorbalance fate test failed on x86_32
floating point precision will cause rgb*max generate different value on
x86_32 and x86_64. have pass fate test on x86_32 and x86_64 by using
lrintf to get the nearest integral value for rgb * max before av_clip.
Reviewed-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Joe Ratterman [Thu, 28 May 2020 17:31:48 +0000 (10:31 -0700)]
Replace 'FontName' with 'Fontname' in the documentation.
This is the only use of 'FontName' with that capitalization, as both
source-code and tests use 'Fontname'. Having consistent capitalization
makes it easier to find the relevant source from the docs.
See these examples for other uses:
libavcodec/ass_split.c:68
tests/ref/fate/sub-cc:9
Guo, Yejun [Mon, 25 May 2020 08:16:22 +0000 (16:16 +0800)]
vf_dnn_processing.c: add dnn backend openvino
We can try with the srcnn model from sr filter.
1) get srcnn.pb model file, see filter sr
2) convert srcnn.pb into openvino model with command:
python mo_tf.py --input_model srcnn.pb --data_type=FP32 --input_shape [1,960,1440,1] --keep_shape_ops
See the script at https://github.com/openvinotoolkit/openvino/tree/master/model-optimizer
We'll see srcnn.xml and srcnn.bin at current path, copy them to the
directory where ffmpeg is.
I have also uploaded the model files at https://github.com/guoyejun/dnn_processing/tree/master/models
3) run with openvino backend:
ffmpeg -i input.jpg -vf format=yuv420p,scale=w=iw*2:h=ih*2,dnn_processing=dnn_backend=openvino:model=srcnn.xml:input=x:output=srcnn/Maximum -y srcnn.ov.jpg
(The input.jpg resolution is 720*480)
Also copy the logs on my skylake machine (4 cpus) locally with openvino backend
and tensorflow backend. just for your information.
$ time ./ffmpeg -i 480p.mp4 -vf format=yuv420p,scale=w=iw*2:h=ih*2,dnn_processing=dnn_backend=tensorflow:model=srcnn.pb:input=x:output=y -y srcnn.tf.mp4
…
frame= 343 fps=2.1 q=31.0 Lsize= 2172kB time=00:00:11.76 bitrate=1511.9kbits/s speed=0.0706x
video:1973kB audio:187kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.517637%
[aac @ 0x2f5db80] Qavg: 454.353
real 2m46.781s
user 9m48.590s
sys 0m55.290s
$ time ./ffmpeg -i 480p.mp4 -vf format=yuv420p,scale=w=iw*2:h=ih*2,dnn_processing=dnn_backend=openvino:model=srcnn.xml:input=x:output=srcnn/Maximum -y srcnn.ov.mp4
…
frame= 343 fps=4.0 q=31.0 Lsize= 2172kB time=00:00:11.76 bitrate=1511.9kbits/s speed=0.137x
video:1973kB audio:187kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.517640%
[aac @ 0x31a9040] Qavg: 454.353
real 1m25.882s
user 5m27.004s
sys 0m0.640s
Signed-off-by: Guo, Yejun <yejun.guo@intel.com> Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
Guo, Yejun [Mon, 25 May 2020 07:38:09 +0000 (15:38 +0800)]
dnn: add openvino as one of dnn backend
OpenVINO is a Deep Learning Deployment Toolkit at
https://github.com/openvinotoolkit/openvino, it supports CPU, GPU
and heterogeneous plugins to accelerate deep learning inferencing.
Please refer to https://github.com/openvinotoolkit/openvino/blob/master/build-instruction.md
to build openvino (c library is built at the same time). Please add
option -DENABLE_MKL_DNN=ON for cmake to enable CPU path. The header
files and libraries are installed to /usr/local/deployment_tools/inference_engine/
with default options on my system.
To build FFmpeg with openvion, take my system as an example, run with:
$ export LD_LIBRARY_PATH=$LD_LIBRARY_PATH:/usr/local/deployment_tools/inference_engine/lib/intel64/:/usr/local/deployment_tools/inference_engine/external/tbb/lib/
$ ../ffmpeg/configure --enable-libopenvino --extra-cflags=-I/usr/local/deployment_tools/inference_engine/include/ --extra-ldflags=-L/usr/local/deployment_tools/inference_engine/lib/intel64
$ make
Here are the features provided by OpenVINO inference engine:
- support more DNN model formats
It supports TensorFlow, Caffe, ONNX, MXNet and Kaldi by converting them
into OpenVINO format with a python script. And torth model
can be first converted into ONNX and then to OpenVINO format.
see the script at https://github.com/openvinotoolkit/openvino/tree/master/model-optimizer/mo.py
which also does some optimization at model level.
- optimize at inference stage
It optimizes for X86 CPUs with SSE, AVX etc.
It also optimizes based on OpenCL for Intel GPUs.
(only Intel GPU supported becuase Intel OpenCL extension is used for optimization)
Signed-off-by: Guo, Yejun <yejun.guo@intel.com> Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
libavformat/mov: Fix memleaks when demuxing DV audio
The code for demuxing DV audio predates the introduction of refcounted
packets and when the latter was added, changes to the former were
forgotten. This meant that when avpriv_dv_produce_packet initialized the
packet containing the AVBufferRef, the AVBufferRef as well as the
underlying AVBuffer leaked; the actual packet data didn't leak: They
were directly freed, but not via their AVBuffer's free function.
https://samples.ffmpeg.org/ffmpeg-bugs/trac/ticket4671/dir1.tar.bz2
contains samples for this (enable_drefs needs to be enabled for them).
Moreover, errors in avpriv_dv_produce_packet were ignored; this has been
changed, too.
Furthermore, in the hypothetical scenario that the track has a palette,
this would leak, too, so reorder the code so that the palette code
appears after the DV audio code.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
avutil/avsscanf: Add () to avoid integer overflow in scanexp()
Fixes: signed integer overflow: 2147483610 + 52 cannot be represented in type 'int' Fixes: 23260/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_PBM_fuzzer-5187871274434560 Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
avformat/utils: reorder duration computation to avoid overflow
Fixes: signed integer overflow: 8 * 9223372036854774783 cannot be represented in type 'long' Fixes: 23381/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-4818340509122560 Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: out of array access Fixes: 23554/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APNG_fuzzer-4796622520451072.fuzz Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
avutil/common: Fix integer overflow in av_ceil_log2_c()
Fixes: left shift of 1913647649 by 1 places cannot be represented in type 'int' Fixes: 23572/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMALOSSLESS_fuzzer-5082619795734528 Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
avcodec/wmalosslessdec: fix overflow with pred in revert_cdlms
Fixes: signed integer overflow: 2048 + 2147483646 cannot be represented in type 'int' Fixes: 23538/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMALOSSLESS_fuzzer-5227567073460224 Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This patch removes a check which throws an error if
the log2 precinct width/height is 0. The standard allows
the first component to have 0 as the log2 width/height.
However, to ensure proper intialization of coding style,
an extra check has been added.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Limin Wang [Sun, 28 Jun 2020 12:46:19 +0000 (20:46 +0800)]
avcodec/mpeg12dec: Fix uninitialized data in fate-sub-cc-scte20
The issue is introduced in a705bcd763e344fa, please tested with below command line:
make V=1 fate-sub-cc-scte20 TARGET_EXEC="valgrind --error-exitcode=1"
Reported-by: Martin Storsjö <martin@martin.st> Reviewed-by: Martin Storsjö <martin@martin.st> Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Derek Buitenhuis [Tue, 23 Jun 2020 15:05:18 +0000 (16:05 +0100)]
lavf/movenc: Use a dynamic buffer when writing the mfra box
When doing streamed output, with e.g. +dash, if the mfra box ended
up being larger than the AVIOContext write buffer, the (unchecked)
seeking back to update the box size would silently fail and produce
an invalid mfra box.
This is similar to how other boxes are written in fragmented mode.
avformat/smoothstreaming: Don't write trailer of subcontext
Nothing written in avformat_write_trailer() for the submuxers will be
output anyway because the AVIOContexts used for actual output have been
closed before the call. Writing the trailer of the subcontext has probably
only been done in order to free the memory allocated by the submuxer.
And this job has been taken over by the deinit functions.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
avformat/mvdec: Fix integer overflow with billions of channels
Fixes: signed integer overflow: 1394614304 * 2 cannot be represented in type 'int' Fixes: 23491/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5697377020411904 Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
libavcodec/jpeg2000dec.c Fixed WRITE_FRAME and tile co-ordinates:
libopenjpeg2000 uses ceiling division while dividing tile
co-ordinates with the sample separation. Also, corrections
were made to the WRITE_FRAME macro.
Improves: p1_01.j2k and p1_07.j2k
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
avformat/avc, mxfenc: Avoid allocation of H264 SPS structure, fix memleak
Up until now, ff_avc_decode_sps would parse a SPS and return some
properties from it in a freshly allocated structure. Yet said structure
is very small and completely internal to libavformat, so there is no
reason to use the heap for it. This commit therefore changes the
function to return an int and to modify a caller-provided structure.
This will also allow ff_avc_decode_sps to return better error codes in
the future.
It also fixes a memleak in mxfenc: If a packet contained multiple SPS,
only the SPS structure belonging to the last SPS would be freed, the
other ones would leak when the pointer is overwritten to point to the
new SPS structure. Of course, without allocations there are no leaks.
This is Coverity issue #1445194.
Furthermore, the SPS structure has been renamed from
H264SequenceParameterSet to H264SPS in order to avoid overlong lines.
Reviewed-by: Tomas Härdin <tjoppen@acc.umu.se> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
After parsing the end timestamp of a WebVTT cue block, the current code
skips everything after the start of the timestamp that is not a \t, ' '
or \n and treats what is next as the start of a WebVTT cue settings list.
Yet if there is no such list, but a single \r, this will skip a part of
the cue payload (namely everything until the first occurence of \t, ' '
or \n) and treat what has not been skipped as the beginning of the
WebVTT cue settings list that extends until the next \r or \n (or the
end).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
avformat/av1: Avoid using dynamic buffer when assembling av1c
Given that AV1 only has exactly one sequence header, it is unnecessary
to copy the content of said sequence header into an intermediate dynamic
buffer; instead the sequence header can be copied from where it is in
the input buffer.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Jun Zhao [Sun, 17 May 2020 04:10:05 +0000 (12:10 +0800)]
lavc/aac_ac3_parser: improve the raw AAC file bit rate calculation
Now we just use one ADTS raw frame to calculate the bit rate, it's
lead to a larger error when get the duration from bit rate, the
improvement cumulate Nth ADTS frames to get the average bit rate.
e,g used the command get the duration like:
ffprobe -show_entries format=duration -i fate-suite/aac/foo.aac
before this improvement dump the duration=2.173935
after this improvement dump the duration=1.979267
in fact, the real duration can be get by command like:
ffmpeg -i fate-suite/aac/foo.aac -f null /dev/null with time=00:00:01.97
avformat/microdvddec: skip malformed lines without frame number.
Fixes: signed integer overflow: 1 - -9223372036854775808 cannot be represented in type 'long' Fixes: 23490/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5133490093031424 Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg Reviewed-by: Nicolas George <george@nsup.org> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
exwm [Fri, 19 Jun 2020 15:58:41 +0000 (15:58 +0000)]
avfilter/zoompan: add in_time variable
Currently, the zoompan filter exposes a 'time' variable (missing from docs) for use in
the 'zoom', 'x', and 'y' expressions. This variable is perhaps better named
'out_time' as it represents the timestamp in seconds of each output frame
produced by zoompan. This patch adds aliases 'out_time' and 'ot' for 'time'.
This patch also adds an 'in_time' (alias 'it') variable that provides access
to the timestamp in seconds of each input frame to the zoompan filter.
This helps to design zoompan filters that depend on the input video timestamps.
For example, it makes it easy to zoom in instantly for only some portion of a video.
Both the 'out_time' and 'in_time' variables have been added in the documentation
for zoompan.
Example usage of 'in_time' in the zoompan filter to zoom in 2x for the
first second of the input video and 1x for the rest:
zoompan=z='if(between(in_time,0,1),2,1):d=1'
V2: Fix zoompan filter documentation stating that the time variable
would be NAN if the input timestamp is unknown.
V3: Add 'it' alias for 'in_time. Add 'out_time' and 'ot' aliases for 'time'.
Minor corrections to zoompan docs.
projects / ffmpeg / log