Only read str_size bytes from offset 30 of extradata if the extradata is
indeed at least 30 + str_size bytes long.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit ff3fad6b0e)
mov_read_custom tries to read three strings belonging to three different
tags. When an already encountered tag is encountered again, a new buffer
for the string to be read is allocated and stored in the pointer
destined for this particular tag. But in this scenario, said pointer
already holds the address of the string read earlier, leading to a leak.
This commit therefore aborts the reading process upon encountering
an already encountered tag.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit dfef1d5e3c)
The argument pertaining to a printf %s conversion specifier must not
be NULL, even if the precision (i.e. the number of characters to write)
is zero. If it is NULL, it is undefined behaviour.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 6de6ce7bc8)
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit c784fe8b86)
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 67434afa7f)
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if allocating the AVStream for the subtitles fails.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 337783b118)
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit a708f65273)
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 9751d75152)
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit e13874b9ea)
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit c70409957c)
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit f3c63e67bb)
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle
or when creating extradata.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit f161f8e4ad)
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 9df560e898)
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon creating an AVStream.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit a5ed8aeea4)
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 331799747e)
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle
or when allocating extradata.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit b12014a5b8)
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit d38694cea9)
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit c13a752733)
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle
or if creating the extradata failed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 5ab39c2d8c)
The already parsed subtitles (contained in an FFDemuxSubtitlesQueue)
would leak if an error happened upon reading a subsequent subtitle.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit a86a5d06d8)
By default, a demuxer's read_close function is not called automatically
if an error happens when reading the header; instead it is up to the
demuxer to clean up after itself in this case. The mov demuxer did this
by calling its read_close function when it encountered some errors when
reading the header. Yet for other errors (mostly adding side-data to
streams) this has been forgotten, so that all the internal structures
of the demuxer leak.
This commit fixes this by making sure mov_read_close is called when
necessary.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit ac378c535b)
Fixes possible leaks of id3v2 metadata as well as an AVDES struct in
case the content is encrypted and an error happens lateron.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 3d3ba43bc6)
In certain error scenarios, the underlying Matroska demuxer was not
properly closed, causing leaks.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 0841063ce6)
When demuxing a Matroska/WebM file, streams are added for tracks and for
attachments, so that the array containing the former can be NULL even
when the corresponding AVFormatContext has streams. So check for there
to be tracks in the MatroskaDemuxContext instead of just streams in the
AVFormatContext before dereferencing the pointer to the tracks.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 1ef30571a0)
matroska_parse_block currently asserts that the duration is not equal to
AV_NOPTS_VALUE, but there is nothing that actually guarantees this. It
is easy to create (spec-compliant) files which run into this assert;
so replace it and instead cap the duration to INT64_MAX, as the duration
field of an AVPacket is an int64_t.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 3714d452b8)
EBML binary elements are already made reference-counted when read;
so when populating the AVStream.attached_pic, one does not need to
allocate a new buffer for the data; instead the current code just
creates a new reference to the underlying AVBuffer. But this can be
improved even further: Just move the already existing reference.
This also fixes a memleak that happens upon error because
matroska_read_close has not been called in this scenario.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit cbe336c9e8)
c801ab43c3 caused a regression: The stream
number is now parsed with strtoll without a fixed basis; as a
consequence, the "010" in a variant stream mapping like "a:010" is now
treated as an octal number (i.e. as eight, not ten). This was not
intended and may break some scripts, so this commit restores the old
behaviour.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit 19a876fd69)
The hevc_mp4toannexb bsf does not explicitly check whether a NAL unit
is so big that it extends beyond the end of the input packet; it does so
only implicitly by using the checked version of the bytestream2 API.
But this has downsides compared to real checks: It can lead to huge
allocations (up to 2GiB) even when the input packet is just a few bytes.
And furthermore it leads to uninitialized data being output.
So add a check to error out early if it happens.
Also check directly whether there is enough data for the length field.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
(cherry picked from commit ea1b71e82f)
Alternatively these conditions could be treated as errors
Fixes: 23147/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_FFWAVESYNTH_fuzzer-5639254549200896
Fixes: signed integer overflow: 9223372036854775807 + 1 cannot be represented in type 'int64_t' (aka 'long')
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 584d334afd)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If this happens, it's a sign of parsing issues earlier in the process, or
misuse by the calling module.
Prevents writing invalid bitstreams.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 318a1a383d)
If this happens, it's a sign of parsing issues earlier in the process, or
misuse by the calling module.
Prevents writing invalid bitstreams.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit ef13fafe22)
Max region ID is 87. Also the region affects not only the G0 charset but G2 and
the national subset as well.
Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit 16d29c1be8)
Previously, prompeg_write() would only report to caller that bytes we
written when a FEC packet was actually created. Not all RTP packets are
expected to generate a FEC packet however, so this behavior was causing
avio to retry writing the RTP packet, eventually forcing the FEC state
machine to send a FEC packet erroneously (and so breaking out of the
retry loop).
This was resulting in incorrect FEC data being generated, and far too
many FEC packets to be sent (~100% FEC overhead).
fix#7863
Signed-off-by: David Holroyd <david.holroyd@m2amedia.tv>
(cherry picked from commit ffc1208266)
fix ticket: 8673
set the default EXTINF duration to 1ms if duration is smaller than 1ms
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
(cherry picked from commit 9dfb19baeb)
fix ticket: 8688
should goto failed workflow if cannot get usable string by ff_make_absolute_url
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
(cherry picked from commit ea1940c6e2)
fix ticket: 8687
workflow should return if there have no value of strchr
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
(cherry picked from commit 029ff31af6)
It is a constant known at codec init, so set it in
ff_frame_thread_init(). Also, only set it for video, since the meaning
of this field is not well-defined for audio with frame threading.
Fixes availability of delay in callbacks invoked from the per-thread
contexts after 1f4cf92cfb.
(cherry picked from commit 6943ab688d)
current_picture was not writable here because a reference existed in
at least avctx->coded_frame, and potentially elsewhere if the caller
created new ones from it.
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 1ee3c984b9)
This causes regressions in end to end timestamps with mp3s and ffmpeg.
The revert is to avoid this regression in the 4.3 release
See: [FFmpeg-devel] [PATCH] Don't adjust start time for MP3 files; packets are not adjusted.
This reverts commit 460132c998.
This makes got_output consistent with the code in slice_end() which sets the output
in slice_end()
if (s->pict_type == AV_PICTURE_TYPE_B || s->low_delay) {
int ret = av_frame_ref(pict, s->current_picture_ptr->f);
...
} else {
Fixes: assertion failure
Fixes: 22178/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MPEG1VIDEO_fuzzer-5664234440753152
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This causes indexes into scale_conversion_table to wrap around, alternatively they
could be clipped, the table be enlarged or we can error out. I have not found a document that specifies
what is the correct way to handle this
Fixes: out of array access
Fixes: 21727/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_HCA_fuzzer-5752477891952640.fuzz
Fixes: 22438/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_HCA_fuzzer-5640717790871552
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Inside a function, the second ; in a double ;; is a null statement, but
outside of functions a double ;; is simply invalid C that compilers
happen to accept. v4l2_m2m_enc.c contained several ;; as a result of
macro-expansion. So change the underlying macro so that it doesn't
happen any longer.
This fixes warnings when compiling with -pedantic: "ISO C does not allow
extra ‘;’ outside of a function".
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
I have attempted to write a JPEG2000 Parser. Have tested
by generating a file containing 14 frames, as mentioned
by Micheal. Have also tried testing with various packet
sizes by setting -frame_size option. Additionally,
fixed a few formatting issues as pointed out by Micheal.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
the target_dec_fuzzer is checking for the avpkt.data pointer but if the
bmp parser cannot combine the frame, the poutbuf is not set.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
the target_dec_fuzzer is checking for the avpkt.data pointer but if the
mlp parser cannot combine the frame, the poutbuf is not set.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
the target_dec_fuzzer is checking for the avpkt.data pointer but if the
png parser cannot combine the frame, the poutbuf is not set.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 2147483636 + 2048 cannot be represented in type 'int'
Fixes: 22016/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMALOSSLESS_fuzzer-5109395618004992
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 1080285923 - -1130879337 cannot be represented in type 'int'
Fixes: 22002/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_VP9_fuzzer-6260237310099456
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Prevent codecpar->codec_id from getting out of sync with the codec instantiated for probing.
Signed-off-by: Samuel Foss <sfoss@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
make checkheaders will get error as follow:
CC libavutil/hwcontext_vulkan.h.o
In file included from libavutil/hwcontext_vulkan.h.c:1:
./libavutil/hwcontext_vulkan.h:130:23: error: ‘AV_NUM_DATA_POINTERS’ undeclared here (not in a function)
130 | void *alloc_pnext[AV_NUM_DATA_POINTERS];
| ^~~~~~~~~~~~~~~~~~~~
./libavutil/hwcontext_vulkan.h:199:43: warning: ‘enum AVPixelFormat’ declared inside parameter list will not be visible outside of this definition or declaration
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
There are two different ways KLV is used in MISB specs - sync and async.
The corresponding text (in ST1401) says:
ISO/IEC 13818-1 Table-34 defines a stream_type = 0x15 for “Metadata carried in PES packets,”
and Table 2-22 defines a stream_id = 0xFC for “metadata stream.”
and
In ISO/IEC 13818-1, Table-34 defines a stream_type = 0x06 for “PES packets containing private
data,” and Table 2-22 defines a stream_id = 0xBD for “private_stream_1.”
These constants allow us to distinguish the two cases, as codec profiles.
Signed-off-by: Marton Balint <cus@passwd.hu>
Then one doesn't need to free the frame in case the length turns out to
be insufficient.
Reviewed-by: Peter Ross <pross@xvid.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
e,g: the command:
ffprobe -show_format -i fate-suite/aac/foo.aac -loglevel 99 will
dump the trace message as follow when start_time is AV_NOPTS_VALUE
[aac @ 0x55bf8e1f3dc0] stream 0: start_time: -326791809695.818 duration: 2.174
[aac @ 0x55bf8e1f3dc0] format: start_time: -9223372036854.775 duration: 2.174 bitrate=120 kb/s
after this fix, will dump the start_time with "NOPTS".
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
Fix vpad.name leak in error path, move the vpad related operation
only if enabled show IR frequency response.
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
We need at least a few bits of entropy to determine the start index of each
queue, in order to let filters run in parallel as much as possible, and
rand() is not thread safe and disrupts any external API's usage of rand,
so instead replace it with av_get_random_seed.
While it has more overhead than rand, we only run it once per filter upon init.
Up until now, the HLS muxer uses av_strtok() to split an input string
controlling parameters of the VariantStreams and then duplicates
parts of this string containing parameters such as the language or the
name of the VariantStream. But these parts are proper zero-terminated
strings of their own that are never modified lateron, so one can simply
use the substring as-is without creating a copy. This commit implements
this.
The same also happened for the string controlling the closed caption
groups.
Furthermore, add const to indicate that the pointers to these substrings
are not used to modify them and also to indicate that these strings are
not allocated on their own.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until now, the HLS muxer duplicated a string for every VariantStream,
although neither the original nor the copies are ever modified. So use
the original directly and stop copying.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
more math unary operations will be added here
It can be tested with the model file generated with below python scripy:
import tensorflow as tf
import numpy as np
import imageio
in_img = imageio.imread('input.jpeg')
in_img = in_img.astype(np.float32)/255.0
in_data = in_img[np.newaxis, :]
x = tf.placeholder(tf.float32, shape=[1, None, None, 3], name='dnn_in')
x1 = tf.subtract(x, 0.5)
x2 = tf.abs(x1)
y = tf.identity(x2, name='dnn_out')
sess=tf.Session()
sess.run(tf.global_variables_initializer())
graph_def = tf.graph_util.convert_variables_to_constants(sess, sess.graph_def, ['dnn_out'])
tf.train.write_graph(graph_def, '.', 'image_process.pb', as_text=False)
print("image_process.pb generated, please use \
path_to_ffmpeg/tools/python/convert.py to generate image_process.model\n")
output = sess.run(y, feed_dict={x: in_data})
imageio.imsave("out.jpg", np.squeeze(output))
Signed-off-by: Ting Fu <ting.fu@intel.com>
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
high resolutions with only small blocks appear to be rather
slow with the fuzzer + sanitizers.
A solution which makes this run faster is welcome.
Fixes: Timeout (did not wait -> 17sec)
Fixes: 21006/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_HEVC_fuzzer-6002552539971584
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The old resync logic had some bugs, for example the packet size could stuck
into 192 bytes, because pos47_full was not updated for every packet, and for
unseekable inputs the resync logic simply skipped some 0x47 sync bytes,
therefore the calculated distance between sync bytes was a multiple of 188
bytes.
AVIO only buffers a single packet (for UDP/mpegts, that usually means 1316
bytes), so for every ten consecutive 188-byte MPEGTS packets there was always a
seek failure, and that caused the old code to not find the 188 byte pattern
across 10 consecutive packets.
This patch changes the custom logic to the one which is used when probing to
determine the packet size. This was already proposed as a FIXME a long time
ago...
then ff_h264_free_tables() and h264_decode_end() can be removed
in h264_decode_init() if it's failed.
The FF_CODEC_CAP_INIT_CLEANUP flag is need for single thread, For multithread,
it'll be cleanup still by AV_CODEC_CAP_FRAME_THREADS flag if have.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
then ff_mpv_encode_end() will be unnecessary in ff_mpv_encode_init()
if it's failed.
The FF_CODEC_CAP_INIT_CLEANUP flag is need for single thread, For multithread,
it'll be cleanup still by AV_CODEC_CAP_FRAME_THREADS flag if have.
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
then we can remove adpcm_encode_close() in adpcm_encode_init() if have failed.
so the goto error lable will be unnecessary and can be removed later.
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
7546ac2fee made it so that the start_time for mp3 files is
adjusted for skip_samples. However, this appears incorrect because
subsequent packet timestamps are not adjusted and skip_samples are
applied by deleting data from a packet without changing the timestamp.
E.g., we are told the start_time is ~25ms and we get a packet with a
timestamp of 0 that has had the skip_samples discarded from it. As such
rendering engines may incorrectly discard everything prior to the
25ms thinking that is where playback should officially start. Since the
samples were deleted without adjusting timestamps though, the true
start_time is still 0.
Other formats like MP4 with edit lists will adjust both the start
time and the timestamps of subsequent packets to avoid this issue.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Currently find_ref_idx() would trigger 2 scans in DPB to find the
requested POC:
1. Firstly, ignore MSB of ref->poc and search for the requested POC;
2. Secondly, compare the entire ref->poc with requested POC;
For long term reference, we are able to only check LSB if MSB is not
presented(e.g. delta_poc_msb_present_flag == 0). However, for short
term reference, we should never ignore poc's MSB and it should be
kind of bit-exact. (Details in 8.3.2)
Otherwise this leads to decoding failures like:
[hevc @ 0x5638f4328600] Error constructing the frame RPS.
[hevc @ 0x5638f4328600] Error parsing NAL unit #2.
[hevc @ 0x5638f4338a80] Could not find ref with POC 21
Error while decoding stream #0:0: Invalid data found when processing input
Search the requested POC based on whether MSB is used, and avoid
the 2-times scan for DPB buffer. This benefits both native HEVC
decoder and integrated HW decoders.
Signed-off-by: Xu Guangxin <guangxin.xu@intel.com>
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
delta_poc_msb_present_flag is needed in find_ref_idx() to
indicate whether MSB of POC should be taken into account.
Details in 8.3.2.
Signed-off-by: Xu Guangxin <guangxin.xu@intel.com>
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Including codecapi.h and uuids.h in UWP mode doesn't define all defines
properly, ending up with constructs that MSVC silently tolerates, but
that clang errors out on, like this:
DEFINE_GUIDEX(CODECAPI_AVEncCommonFormatConstraint);
Just avoid including codecapi.h completely and hardcode the last few
enum values we use from there. We already use local versions of most
enums from there, due to older mingw-w64 headers being incomplete.
Signed-off-by: Martin Storsjö <martin@martin.st>
This might have been used originally for the decoder parts of
the MediaFoundation wrapper, which aren't merged yet.
Signed-off-by: Martin Storsjö <martin@martin.st>
We want to copy the lowest amount of bytes per line, but while the buffer
stride is sanitized, the src/dst stride can be negative, and negative numbers
of bytes do not make a lot of sense.
By default now, if AV_EF_CRCCHECK or AV_EF_IGNORE_ERR are enabled the decoder
will skip the chunk and carry on with the next one. This should make the
decoder able to decode more corrupt files because the functions which decode
individual chunks will very likely error out if fed invalid data and stop the
decoding of the entire image.
A lot of files have CRC included.
The CRC only covers 34 bytes at most from the frame but it should still be
enough for some amount of error detection.
The CRC flag is only signalled once every few minutes but CRC is still
always present so the patch uses the file version instead.
CRC on 24-bit files wants non-padded samples so skip such files.
Some corrupt samples may have been output before the final check
depending on the -max_samples setting.
The size of a single allocation performed by av_malloc() or av_realloc()
is supposed to be bounded by max_alloc_size, which defaults to INT_MAX
and can be set by the user; yet currently this is not completely
honoured: The actual value used is max_alloc_size - 32. How this came
to be can only be understood historically:
a) 0ecca7a49f disallowed allocations
> INT_MAX. At that time the size parameter of av_malloc() was an
unsigned and the commentary added ("lets disallow possible ambiguous
cases") indicates that this was done as a precaution against calling the
functions with negative int values. Genuinely limiting the size of
allocations to INT_MAX doesn't seem to have been the intention given
that at this time the memalign hack introduced in commit
da9b170c6f (which when enabled increased
the size of allocations slightly so that one can return a correctly
aligned pointer that actually does not point to the beginning of the
allocated buffer) was already present.
b) Said memalign hack allocated 17 bytes more than actually desired, yet
allocating 16 bytes more is actually enough and so this was changed in
a9493601638b048c44751956d2360f215918800c; this commit also replaced
INT_MAX by INT_MAX - 16 (and made the limit therefore a limit on the size
of the allocated buffer), but kept the comment, although there is nothing
ambiguous about allocating (INT_MAX - 16)..INT_MAX.
c) 13dfce3d44 then increased 16 to 32 for
AVX, 6b4c0be558 replaced INT_MAX by
MAX_MALLOC_SIZE (which was of course defined to be INT_MAX) and
5a8e994287 added max_alloc_size and made
it user-selectable.
d) 4fb311c804 then dropped the memalign
hack, yet it kept the -32 (probably because the comment about ambiguous
cases was still present?), although it is no longer needed at all after
this commit. Therefore this commit removes it and uses max_alloc_size
directly.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
It avoids leaving dangling pointers behind in memory.
Also remove redundant checks for whether the URLContext to be closed is
already NULL.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Some flac muxers write truncated metadata picture size if the picture
data do not fit in 24 bits. Detect this by truncting the size found inside
the picture block and if it matches the block size use it and read rest
of picture data.
This workaround is only for flac files and not ogg files with flac
METADATA_BLOCK_PICTURE comments and it can be disabled with strict level
above normal. Currently there is a 500MB limit on truncate size to protect
from large memory allocations.
The truncation bug in lavf flacenc was fixed in e447a4d112
but based on existing broken files other unknown flac muxers seems to truncate also.
Before the fix a broken flac file for reproduction could be generated with:
ffmpeg -f lavfi -i sine -f lavfi -i color=red:size=2400x2400 -map 0:0 -map 1:0 -c:v:0 bmp -disposition:1 attached_pic -t 1 test.flac
Fixes ticket 6333
Signed-off-by: Anton Khirnov <anton@khirnov.net>
ff_id3v2_parse_apic/chapters/priv/priv_dict all had a parameter
extra_meta of type ID3v2ExtraMeta ** as if the functions wanted to make
*extra_meta point to something else. But they don't, so just use an
ID3v2ExtraMeta *.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until now, the ID3v2ExtraMeta structure (which is used when parsing
ID3v2 tags containing attached pictures, chapters etc.) contained a
pointer to separately allocated data that depended on the type of the
tag. Yet the difference of the sizes of the largest and the smallest of
these structures is fairly small, so that it is better to simply include
a union of all the possible types of tag-dependent structures in
ID3v2ExtraMeta. This commit implements this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
If the write_id3v2 option is set, the aiff muxer would write id3v2 tags
if there is global metadata or if there are attached pics to write.
Chapters are ignored in this check that precedes writing id3v2 tags.
Yet 47ac344970 added support for writing
chapters as id3v2 tags, so one should check for the existence of chapters,
too; otherwise the chapters would only be written in case there is
global metadata or an attached pic.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Just including icodecapi.h + codecapi.h apparently misses some bits,
that MSVC doesn't complain about, but Clang does. Thus add a missing
header to the custom hacky include path (required for keeping
ICodecAPI visible when building in UWP mode, as it is intended to).
The issue in Windows SDK headers about ICodecAPI availability in
UWP/app mode has been reported upstream at
https://developercommunity.visualstudio.com/content/problem/1037125/icodecapi-not-visible-when-compiling-in-uwp-mode.html,
and later also filed in Feedback Hub (which can't be linked to
externally).
Signed-off-by: Martin Storsjö <martin@martin.st>
This was never actually used, likely due to confusion, as the device context
also had one used for uploads and downloads.
Also, since we're only using it for very quick image barriers (which are
practically free on all hardware), use the compute queue instead of the
transfer queue.
This commit makes full use of the enabled queues to provide asynchronous
uploads of images (downloads remain synchronous).
For a pure uploading use cases, the performance gains can be significant.
With this, the puzzle of making libplacebo, ffmpeg and any other Vulkan
API users interoperable is complete.
Users of both libraries can initialize one another's contexts without having
to create a new one.
This allows for users who derive devices to set options for the
new device context they derive.
The main use case of this is to allow users to enable extensions
(such as surface drawing extensions) in Vulkan while deriving from
the device their frames are on. That way, users don't need to write
any initialization code themselves, since the Vulkan spec invalidates
mixing instances, physical devices and active devices.
Apart from Vulkan, other hwcontexts ignore the opts argument since they
don't support options at all (or in VAAPI and OpenCL's case, options are
currently only used for device selection, which device_derive overrides).
The Xiph foundation never standardized either Daala nor its mapping in Ogg,
and all files that were created are undecodable without knowledge of the
git hash.
The description of AVOutputFormat.init contains the statement that "this
method must not write output". Due to this, the webm_chunk muxer defers
opening the AVIOContext for the child muxer until avformat_write_header(),
i.e. there is no AVIOContext when the sub-muxer's avformat_init_output()
is called. But this violates the documentation of said function which
requires the AVFormatContext to have an already opened AVIOContext.
This commit fixes this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Don't use the functions for searching substrings when all one is
looking for is a char anyway. Given that there is already a standard
library function for "find last occurence of a char in a string" also
allows one to remove a custom loop.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The current parsing process for adaptation_sets does not guarantee
every adaptation set to contain at least one stream, because the loop
exits immediately as soon as the end of the string has been reached,
without checking whether the currently active adaptation set group is
lacking a stream. This would lead to segfaults lateron as the rest of
the code presumed that every adaptation set contains a stream. This
commit fixes this by erroring out when the last adaptation set group
is incomplete.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The WebM DASH manifest muxer uses a loop to parse the adaptation_sets
string (which is given by the user and governs which AVStreams are
mapped to what adaptation set) and the very beginning of this loop is
"if (*p == ' ') continue;". This of course leads to an infinite loop if
the condition is true. It is true if e.g. the string begins with ' ' or
if there are more than one ' ' between different adaptation set groups.
To fix this, the parsing process has been modified to consume the space
if it is at a place where it can legitimately occur, i.e. when a new
adaptation set group is expected. The latter restriction implies that an
error is returned if a space exists where none is allowed to exist.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The syntax of the adaptation_sets string by which the user determines
the mapping of AVStreams to adaptation sets is
"id=x,streams=a,b,c id=y,streams=d,e" (means: the streams with the
indices a, b and c belong to the adaptation set with id x). Yet there
was no check for whether these indices were actual numbers and if there
is a number whether it really extends to the next ',', ' ' or to the
end of the string or not. This commit adds a check for this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
In order to parse a number from a string, the WebM DASH manifest muxer
would duplicate (via heap-allocation) the part of the string that
contains the number, then read the number via atoi() and then free the
duplicate again. This has been replaced by simply using strtoll() (which
in contrast to atoi() has defined behaviour when the number is not
representable).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Since commit c5324d92c5 all custom
interleave_packet() functions always return clean packets (even on
error), so that unreferencing manually can be removed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
AVStream.request_probe as well as AVStream.mux_ts_offset are below the
separator of public and private fields, so that a further "NOT PART OF
PUBLIC API" is redundant.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This change makes it possible for child encoders to define custom profile
option names which can be used for setting the AVCodecContext->profile.
Also rename unit name to something rather unique, so it won't be used elsewhere.
Signed-off-by: Marton Balint <cus@passwd.hu>
This will be used for AVCodecContext->profile. By specifying constants in the
encoders we won't have to use the common AVCodecContext options table and
different encoders can use the same profile name even with different values.
Signed-off-by: Marton Balint <cus@passwd.hu>
The framerate field is the one users are supposed to set, but not
all users might be setting it, so it might be good to fall back
time_base in that case.
Signed-off-by: Martin Storsjö <martin@martin.st>
Move the copying of the frame to vos_data further up in the function,
so that when writing the actual frame data for the first frame, it's
clear that the stream really is in annex b format, for the cases where
we create extradata from the first frame.
Alternatively - we could invert the checks for bitstream format. If
extradata is missing, we can't pretend that the bitstream is in
mp4 form, because we can't even know the NAL unit length prefix size
in that case.
Also avoid creating extradata for AVC intra. If the track tag is
an AVC intra tag, don't copy the frame into vos_data - this matches
other existing cases of how vos_data and TAG_IS_AVCI interact in
other places.
Signed-off-by: Martin Storsjö <martin@martin.st>
EAGAIN is returned when input is provided but can't be consumed. The filtering
process is unaffected in this case, and the function will be able to consume
new input after retrieving filtered packets with av_bsf_receive_packet().
Remove the line about empty packets never failing added in
41b05b849f while at it. Even if it's currently
the case, it unnecessarily constrains the API and could be changed in the future
in case it needs to be extended.
The user should always check for errors and never expect a call to never fail.
Reviewed-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
av_stream_get_side_data() tells the caller whether a stream has side
data of a specific type; if present it can also tell the caller the size
of the side data via an optional argument. The Matroska muxer always
used this optional argument, although it doesn't really need the size,
as the relevant side-data are not buffers, but structures. So change
this.
Furthermore, relying on the size also made the code susceptible to
a quirk of av_stream_get_side_data(): It only sets the size argument if
it found side data of the desired type. mkv_write_video_color() checks
for side-data twice with the same variable for the size without resetting
the size in between; if the second type of side-data isn't present, the
size will still be what it was after the first call. This was not
dangerous in practice, as the check for the existence of the second
side-data compared the size with the expected size, so it would only be
problematic if lots of elements were to be added to AVContentLightMetadata.
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Some real-world sites use an authorization header with a bearer token; when
combined with lengthy request parameters to identify the video segment,
it's rather trivial these days to have a request body of more than 4k bytes.
MAX_URL_SIZE is hard-coded to 4k bytes in libavformat/internal.h, and
HTTP_HEADERS_SIZE is 4k as well in libavformat/http.h, so this patch increases
the buffer size to 8k, as that is the default request body limit in Apache, and
most other httpds seem to support at least as much, if not more.
Signed-off-by: Marton Balint <cus@passwd.hu>
Both are codec properties and not encoder capabilities. The relevant
AVCodecDescriptor.props flags exist for this purpose.
Signed-off-by: James Almer <jamrial@gmail.com>
TIFF 6 spec: "Do not use both strip-oriented and tile-oriented fields in the same TIFF file."
Fixes: null pointer use, crash
Fixes: crash-762680f9d1b27f9b9085e12887ad44893fb2b020
Found-by: Shiziru <lunasl@protonmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: division by zero
Fixes: 21677/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DSD_MSBF_fuzzer-5712547983654912
Fixes: 21751/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DSD_LSBF_fuzzer-5197097180856320
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -1238335488 + -1003634688 cannot be represented in type 'int'
Fixes: 21649/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MJPEG_fuzzer-5112005765890048
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -717241856 + -1434459904 cannot be represented in type 'int'
Fixes: 21405/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMALOSSLESS_fuzzer-5677143666458624
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -9223372036854775808 - 45000 cannot be represented in type 'long'
Fixes: ticket8187
Found-by: Suhwan
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 30000299 * 256 cannot be represented in type 'int'
Fixes: ticket8184
Found-by: Suhwan
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: out of array access
Fixes: stack-buffer-overflow-READ-0x0831fff1
Found-by: GalyCannon <galycannon@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The TTA muxer writes a seektable in a dynamic buffer as it receives
packets and when writing the trailer, closes the dynamic buffer using
avio_close_dyn_buf(), writes the seektable and frees the buffer. But
the TTA muxer already has a deinit function which unconditionally
calls ffio_free_dyn_buf() on the dynamic buffer, so switching to
avio_get_dyn_buf() means that one can remove the code to free the
buffer; furthermore, it also might save an allocation if the seektable
is so small that it fits into the dynamic buffer's write buffer or if
adding the padding that avio_close_dyn_buf() adds necessitated
reallocating of the underlying buffer.
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
ff_id3v2_free_extra_meta() takes a ID3V2ExtraMeta ** so that it can
already reset the pointer.
Reviewed-by: Jun Zhao <mypopy@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
A buffer whose size is not a multiple of four has been initialized using
consecutive writes of 32bits. This results in a stack-buffer-overflow
reported by ASAN in the checkasm-sw_scale FATE-test.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
If the RPS we are predicting from has maximum size then at least one of
the pictures in it must be discarded before adding the current one.
Also revert 588114cea4, which added
now-redundant checks for the special case of a too-large RPS with all
pictures being in the same direction from the current one.
Sticking a full frame in the extradata works, as the code for writing
the avcC/hvcC extracts the relevant parameter set NAL units - provided
that they actually exist in the frame.
Some encoders don't provide split out extradata directly on init (or
at all). In particular, the MediaFoundation encoder wrapper doesn't
always (depending on the actual encoder device) - this is the case for
Qualcomm's HEVC encoder on SD835, and also on some QSV H264 encoders).
This only works for cases where the moov hasn't already been written
(e.g. when not writing fragmented mp4 with empty_moov, unless using
the delay_moov option).
Signed-off-by: Martin Storsjö <martin@martin.st>
2d8d554f15 added a new error condition
to mov_read_stsz() but forgot to free a temporary buffer when it
occurs.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
changes since v1
- default behavior, no longer hidden behind decoder parameter
- updated tests to reflect change
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This contains encoder wrappers for H264, HEVC, AAC, AC3 and MP3.
This is based on top of an original patch by wm4
<nfxjfg@googlemail.com>. The original patch supported both encoding
and decoding, but this patch only includes encoding.
The patch contains further changes by Paweł Wegner
<pawel.wegner95@gmail.com> (primarily for splitting out the encoding
parts of the original patch) and further cleanup, build compatibility
fixes and tweaks for use with Qualcomm encoders by Martin Storsjö.
Signed-off-by: Martin Storsjö <martin@martin.st>
because it need be check for success, is should not
change the old way if it failure.
fix ticket: 8674
Signed-off-by: Steven Liu <liuqi05@kuaishou.com>
The Matroska muxer has always mapped the title tag to the FileDescription
element for attachments streams since support for writing attachments
was added in commit c7a63a521b. This
commit merely documents this fact.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Each AttachedFile in Matroska can have a FileDescription element that
contains a human-friendly name for the attached file; yet this element
has been ignored up until now. This commit changes this and exports it
as title tag instead (the Matroska muxer mapped the title tag to the
AttachedFile element since support for Attachments was added).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The Matroska muxer writes the Chapters early when chapters were already
available when writing the header; in this case any tags pertaining to
these chapters get written, too.
Yet if no chapters had been supplied before writing the header, Chapters
can also be written when writing the trailer if any are supplied. Tags
belonging to these chapters were up until now completely ignored.
This commit changes this: Writing the tags belonging to chapters has
been moved to mkv_write_chapters(). If mkv_write_tags() has not been
called yet (i.e. when chapters are written when writing the header),
the AVIOContext for writing the ordinary Tags element is used, but not
output, as this is left to mkv_write_tags() in order to only write one
Tags element. Yet if mkv_write_tags() has already been called,
mkv_write_chapters() will output a Tags element of its own which only
contains the tags for chapters.
When chapters are available initially, the corresponding tags will now
be the first tags in the Tags element; but the ordering of tags in Tags
is irrelevant anyway.
This commit also makes chapter_id_offset local to mkv_write_chapters()
as it is used only there and not reused at all.
Potentially writing a second Tags element means that the maximum number
of SeekHead entries had to be incremented. All the changes to FATE
result from the ensuing increase in the amount of space reserved for the
SeekHead (21 bytes more).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This is needed so that it can access mkv_write_tag() and mkv_check_tag()
without using forward declarations (which are unnecessary here).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until now, the Matroska muxer writes only one Tags level 1 element
and therefore using a certain place to store the dynamic buffer used for
writing it was hardcoded; yet the Matroska specifications allow an
unlimited amount of Tags elements and we have reason to write a second
one: If chapters are provided after writing the header, they are written
when writing the trailer; yet the corresponding tags are ignored. This
can be fixed by writing them in a second Tags element.
Also use a MatroskaMuxContext * instead of an AVFormatContext * as
parameter in mkv_write_tag() and mkv_write_tag_targets() as that is all
these functions use.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Mostly reindentation after the last commit. Also remove a variable that
is always zero; move another variable to a more local scope and don't
assign a value to a local variable immediately before leaving the function.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Mainly reindentation plus some reordering in MatroskaMuxContext;
moreover, use the IS_SEEKABLE() macro troughout the code.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
EBML numbers are variable length numbers: Only seven bits of every byte
are available to encode the number, the other bits encode the length of
the number itself. So an eight byte EBML number can only encode numbers
in the range 0..(2^56 - 1). And when using EBML numbers to encode the
length of an EBML element, the EBML number corresponding to 2^56 - 1 is
actually reserved to mean that the length of the corresponding element
is unknown.
And therefore put_ebml_length() asserted that the length it should
represent is < 2^56 - 1. Yet there was nothing that actually guaranteed
this to be true for the Segment (the main/root EBML element of a
Matroska file that encompasses nearly the whole file). This commit
changes this by checking in advance how big the length is and only
updating the number if it is representable at all; if not, the unknown
length element is not touched.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The Matroska muxer has a pair of functions designed to write master
elements whose exact length is not known in advance: start_ebml_master()
and end_ebml_master(). The first one of these would write the EBML ID of
the master element that is about to be started, reserve some bytes for
the length field and record the current position as well as how many
bytes were used for the length field. When writing the master's contents
is finished, end_ebml_master() gets the current position (at the end of
the master element), seeks to the length field using the recorded
position, writes the length field and seeks back to the end of the
master element so that one can continue writing other elements.
But if one wants to modify the content of the master element itself,
then the seek back is superfluous. This is the scenario that presents
itself when writing the trailer: One wants to update several elements
contained in the Segment master element (this is the main/root master
element of a Matroska file) that were already written when writing the
header. The current approach is to seek to the beginning of the file
to update the elements, then seek to the end, call end_ebml_master()
which immediately seeks to the beginning to write the length and seeks
back. The seek to the end (which has only been performed because
end_ebml_master() uses the initial position to determine the length
of the master element) and the seek back are of course superfluous.
This commit avoids these seeks by no longer using start/end_ebml_master()
to write the segment's length field. Instead, it is now written
manually. The new approach is: Seek to the beginning to write the length
field, then update the elements (in the order they appear in the file)
and seek back to the end.
This reduces the ordinary amount of seeks of the Matroska muxer to two
(ordinary excludes scenarios where one has big Chapters or Attachments
or where one writes the Cues at the front).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
If the AVIOContext for output was unseekable when writing the header,
no space for Cues would be reserved even if the reserve_index_space
option was used (because it is reasonable to expect that one can't seek
back to the beginning to write the Cues anyway). But if the AVIOContext
was seekable when writing the trailer, it was presumed that space for
the Cues had been reserved when the reserve_index_space option indicated
so even when it was not. As a result, the beginning of the file would be
overwritten.
This commit fixes this: If the reserve_index_space option had been used
and no space has been reserved in advance because of unseekability when
writing the header, then no attempt to write Cues will be performed
when writing the trailer; after all, writing them at the front is
impossible and writing them at the end is probably undesired.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
We won't be able to seek back to write the actual duration anyway.
FATE-tests using the md5pipe command had to be updated due to this change.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The Matroska muxer behaves differently in several ways when it thinks
that it is in unseekable/livestreaming mode: It does not add Cue entries
because they won't be written anyway for a livestream and it writes some
elements only preliminarily (with the intention to overwrite them with
an updated version at the end) when non-livestreaming etc.
There are two ways to set the Matroska muxer into livestreaming mode:
Setting an option or by providing an unseekable AVIOContext. Yet the
actual checks were not consistent:
If the AVIOContext was unseekable and no AAC extradata was available
when writing the header, writing the header failed; but if the AVIOContext
was seekable, it didn't, because the muxer expected to get the extradata
via packet side-data. Here the livestreaming option has not been checked,
although one can't use the updated extradata in case it is a livestream.
If the reserve_index_space option was used, space for writing Cues would
be reserved when writing the header unless the AVIOContext was
unseekable. Yet Cues were only written if the livestreaming option was
not set and the AVIOContext was seekable (when writing the trailer), so
if the AVIOContext was seekable and the livestreaming option set, the
reserved space would never be used at all.
If the AVIOContext was unseekable and the livestreaming option was not
set, it would be attempted to update the main length field at the end.
After all, it might be possible that the file is so short that it fits
into the AVIOContext's buffer in which case the seek back would work.
Yet this is dangerous: It might be that we are not dealing with a
simple output file, but that our output gets split into chunks and that
each of these chunks is actually seekable. In this case some part of the
last chunk (namely the eight bytes that have the same offset as the
length field had in the header) will be overwritten with what the muxer
wrongly believes to be the filesize.
(The livestreaming option has been added to deal with this scenario,
yet its documentation ("Write files assuming it is a live stream.")
doesn't make this clear at all. At least the segment muxer does not
set the option for live and given that the chances of successfully
seeking when the output is actually unseekable are slim, it is best to
not attempt to update the length field in the unseekable case at all.)
All these inconsistencies were fixed by treating the output as seekable
if the livestreaming option is not set and if the AVIOContext is
seekable. A macro has been used to enforce consistency and improve code
readability.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
If the Matroska muxer's AVIOContext was unseekable when writing the
header, but is seekable when writing the trailer, the code for writing
the trailer presumes that a dynamic buffer exists and tries to update
its content in order to overwrite data that has already been
preliminarily written when writing the header, yet said buffer doesn't
exist as it has been written finally and not preliminarily when writing
the header (because of the unseekability it was presumed that one won't
be able to update the data anyway).
This commit adds a check for this and also for a similar situation
involving updating extradata with new data from packet side-data.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
It's the correct dedicated capability reported by supported encoders.
Otherwise, the frame thread path will be used for unsupported encoders
like r210 for no gain.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: James Almer <jamrial@gmail.com>
Enables HEVC Range Extension encoding support (Linux) for 4:2:2 8/10 bit
on ICL+ (gen11 +) platform.
Restricted to linux only for now.
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Enables HEVC Range Extension decoding support (Linux) for 4:2:2 8/10 bit
on ICL+ (gen11 +) platform.
Restricted to linux only for now.
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
The parsing process of the AVOpt-enabled string controlling the mapping
of input streams to variant streams is roughly as follows: Space and tab
separate variant stream group maps while the entries in each variant
stream group map are separated by ','.
The parsing process of each variant stream group proceeded as follows:
At first the number of occurences of "a:", "v:" and "s:" in each variant
stream group is calculated so that one can can allocate an array of
streams with this number of entries. Then the string is split along ','
and each substring is parsed. If such a substring starts with "a:", "s:"
or "v:" it is treated as stream specifier and (if there is a correct
number after ':') a stream of the variant stream is mapped to one of the
actual input streams.
Nothing actually guarantees that the number of streams allocated initially
equals the number of streams that are mapped to an actual input stream.
These numbers can differ if e.g. the name, the sgroup, agroup or ccgroup
of the variant stream contain "a:", "s:" or "v:".
The problem hereby is that the rest of the code presumes these numbers
to be equal and segfaults if it isn't (because the corresponding input
stream is NULL).
This commit fixes this by modifying the initial counting process to only
count occurences of "a:", "s:" or "v:" that are at the beginning or that
immediately follow a ','.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This avoids accessing an old, no longer valid buffer.
Fixes: out of array access
Fixes: crash_audio-2020
Found-by: le wu <shoulewoba@gmail.com>
Reviewed-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 2147483647 + 1 cannot be represented in type 'int'
Fixes: 19950/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_BINKAUDIO_DCT_fuzzer-5765514337189888
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Suggested-by: Paul
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Call it directly from write_packets_common() instead of indirectly
through prepare_input_packet().
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This commit stops using pkt->stream_index as index in an AVFormatContext's
streams array before actually comparing the value with the count of
streams in said array. 96e5e6abb9 used
pkt->stream_index in prepare_input_packet() before checking and
6406351222 did likewise in
write_packets_common().
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The NEON hscale function only supports X8 filter sizes and should only
be selected when these are being used. At the moment filterAlign is
set to 8 but in the future when extra NEON assembly for specific sizes is
added they will need to have checks here too.
The immediate usecase for this change is making the hscale checkasm
test easier and without NEON specific edge-cases (x86 already has these
guards).
This applies the same fix from 718c8f9aa5
on the 32 bit arm version of the function, fixing fate-checkasm-sw_scale
there.
Signed-off-by: Martin Storsjö <martin@martin.st>
If you have a file with multiple Metadata Keys, the second time you parse
the keys, you will re-alloc c->meta_keys without freeing the old one.
This change will avoid parsing all the consecutive Metadata keys.
Reviewed-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Many places are using their own custom code for handling overflow
around timestamps or other int64_t values. There are enough of these
now that having some common saturated math functions seems sound.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Also fill x8-x17 with garbage before calling the function.
Figure out the number of stack parameters and make sure that the
value on the stack after those is untouched.
Signed-off-by: Martin Storsjö <martin@martin.st>
Figure out the number of stack parameters and make sure that the
value on the stack after those is untouched.
Signed-off-by: Martin Storsjö <martin@martin.st>
We should just use a normal bl here, and the linker will add the 'x'
bit if necessary.
This fixes calling the checkasm_fail_func on windows, where the
code is built in thumb mode (and the linker doesn't clear the 'x'
bit in the blx instruction).
Signed-off-by: Martin Storsjö <martin@martin.st>
On windows and darwin (and modern android), the x18 register is reserved
and shouldn't be modified by user code, while it is freely available on
linux. Strictly avoid it, to keep the assembly code portable.
This would have helped catch the issue fixed in 872790b1f9
immediately.
Signed-off-by: Martin Storsjö <martin@martin.st>
The NEON hscale function only supports X8 filter sizes and should only
be selected when these are being used. At the moment filterAlign is
set to 8 but in the future when extra NEON assembly for specific sizes is
added they will need to have checks here too.
The immediate usecase for this change is making the hscale checkasm
test easier and without NEON specific edge-cases (x86 already has these
guards).
Signed-off-by: Josh de Kock <josh@itanimul.li>
The h264_nvenc and hevc_nvenc encoders aren't respecting the framerate in the codec context.
Instead it was using the timebase which in our use-case was 1/1000 so the encoder was behaving
as if we wanted 1000fps. This resulted in poor encoding results due to an extremely low bitrate.
Both the amf and qsv encoders already contain similar logic to first check the framerate before
falling back to the timebase.
Signed-off-by: Zachariah Brown <zachariah@renewedvision.com>
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
A temporary heap array currently stores pids from all streams. It is
used to make sure there are no duplicated pids. However, this array is
not needed because the pids from past streams are stored in the
MpegTSWriteStream structs.
Reviewed-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
Until now, we would have only attempted to utilize already decrypted
data if it was enough to fill the size of buffer requested, that could
very well be up to 32 kilobytes.
With keep-alive connections this would just lead to recv blocking
until rw_timeout had been reached, as the connection would not be
officially closed after each transfer. This would also lead to a
loop, as such timed out I/O request would just be attempted again.
By just returning the available decrypted data, keep-alive based
connectivity such as HLS playback is fixed with schannel.
The dec_buf seems to be properly managed between read calls,
and we have no logic to decrypt before attempting socket I/O.
Thus - until now - such data would not be decrypted in case of
connections such as HTTP keep-alive, as the recv call would
always get executed first, block until rw_timeout, and then get
retried by retry_transfer_wrapper.
Thus - if data is received - decrypt all of it right away. This way
it is available for the following requests in case they can be
satisfied with it.
Use this in vf_spp.c, where the get_pixels operation is done on
unaligned source addresses.
Hook up the x86 (mmx and sse) versions of get_pixels to this
function pointer, as those implementations seem to support unaligned
use.
This fixes fate-filter-spp on armv7.
Signed-off-by: Martin Storsjö <martin@martin.st>
The reference (thp.txt) uses floats so wrap around would seem incorrect.
Fixes: signed integer overflow: 1073741824 + 1073741824 cannot be represented in type 'int'
Fixes: 20658/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ADPCM_THP_fuzzer-5646302555930624
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Set DEPRECATED flag to option cabac, replace with coder. The
priority logic is:
1. s->coder; then
2. avctx->coder_type; then
3. s->cabac.
Change the default option to -1 and allow the default cabac to be
determined by profile.
Add FF_API_OPENH264_CABAC macro for cabac to remove this option after
LIBAVCODEC_VERSION_MAJOR = 59.
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
And determine the profile with following priority:
1. s->profile; then
2. avctx->profile; then
3. s->cabac; then
4. a default profile.
This seems more natural in case user somehow sets both avctx->profile and
s->profile.
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Support the profiles "constrained_baseline" and "high" for libopenh264
version >= 1.8, support "constrained_baseline" and "main" for earlier
version.
If option not supported with current version, convert to constrained
baseline with a warning for users.
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Only warn instead. API users can find out which extensions were unavailable
by using the enabled_inst_extensions and enabled_dev_extensions fields.
This eliminates having to trial-and-error to find which extensions were missing.
Due to our AVHWDevice infrastructure, where API users are offered a way
to derive contexts rather than always create new one, our filterchains,
being supported by a single hardware device context, can grow to considerable
size.
Hence, in such situations, using the maximum amount of queues the device offers
can be benefitial to eliminating bottlenecks where queue submissions on the
same family have to wait for the previous one to finish.
Fixes: Out of array access
Fixes: 20659/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_IFF_ILBM_fuzzer-5658548592967680
Fixes: 20659/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_IFF_ILBM_fuzzer-5723561177382912
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This is intended to replace the deprecated the AV_FRAME_DATA_QP_TABLE*
API and extend it to a wider range of codecs.
In the future, it may also be extended to support other encoding
parameters such as motion vectors.
Additional changes by Anton Khirnov <anton@khirnov.net> with suggestions
by Lynne <dev@lynne.ee>.
Signed-off-by: Juan De León <juandl@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Fixes: signed integer overflow: 2147483594 + 142 cannot be represented in type 'int'
Fixes: 20492/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_UTVIDEO_fuzzer-5658568101724160
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 1931744255 + 252497024 cannot be represented in type 'int'
Fixes: 20492/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TTA_fuzzer-5763348114440192
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 289082077 - -2003141111 cannot be represented in type 'int'
Fixes: 20492/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_RALF_fuzzer-5196077752123392
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This reverts commit 97b526c192.
It broke the API, and assumed no other APIs used multiple semaphores.
This also disallowed certain optimizations to happen.
Dealing with APIs that give or expect single semaphores is easier when
we use per-image semaphores.
Fixes: negation of -2147483648 cannot be represented in type 'int32_t' (aka 'int'); cast to an unsigned type to negate this value to itself
Fixes: signed integer overflow: -2147483648 - 1 cannot be represented in type 'int32_t' (aka 'int')
Fixes: 20492/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_IFF_ILBM_fuzzer-5764066459254784
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The fixed point integer reference specifies the multiplication used
to have 16bit input and clips so we need to clip the input
The floating point implementation does not seem to do that.
Fixes: signed integer overflow: 6317568 * 410 cannot be represented in type 'int'
Fixes: 20492/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_G729_fuzzer-5700189272932352
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -2147483647 - 61436 cannot be represented in type 'int'
Fixes: 20492/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ADPCM_IMA_APM_fuzzer-5092176004644864
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The specs note that images should be in the GENERAL layout when exporting
for maximum compatibility.
CUDA exported images are handled differently, and the queue is the same,
so we don't need to do that there.
As it turns out, we were already assuming and treating all images as if they had
concurrent access mode. This just changes the flag to CONCURRENT, which has less
restrictions than EXCLUSIVE, and fixed validation messages on machines with
multiple queues.
The validation layer didn't pick this up because the machine I was testing on
had only a single queue.
This solves a huge oversight - it lets users reliably use their own
AVVulkanDeviceContext. Otherwise, the extensions supplied and enabled
are not discoverable by anything outside of hwcontext_vulkan.
Also clarifies that any user-supplied VkInstance must be at least 1.1.
Also documents all options supported by the hwdevice.
This lets users enable all extensions they need without writing their own
instance initialization code.
For every variantstream vs, vs->packets_written is set to one, only to be
set to zero a few lines below. Given that the relevant structure has
been zeroed during the allocation, this commit removes both assignments.
A redundant initialization for vs->init_range_length has been removed as
well a few lines below. Given that the relevant structure has been
zeroed during the allocation, this commit removes both assignments. A
redundant initialization for vs->init_range_length has been removed as
well.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
By the av_strtok() description:
* On the first call to av_strtok(), s should point to the string to
* parse, and the value of saveptr is ignored. In subsequent calls, s
* should be NULL, and saveptr should be unchanged since the previous
* call.
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
V4L2 api can indicate that flushing of the capture buffers is completed
by setting the V4L2_BUF_FLAG_LAST flag.
Use guards because the flag was only defined in Linux v4.2.
Reference:
linux/Documentation/media/uapi/v4l/dev-decoder.rst
"The client must continue to handle both queues independently,
similarly to normal decode operation. This includes:
...
- queuing and dequeuing CAPTURE buffers, until a buffer marked with
the V4L2_BUF_FLAG_LAST flag is dequeued"
Reviewed-by: Ming Qian <ming.qian@nxp.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
v4l2_m2m devices may send an empty packet/frame while draining
to indicate that all capture buffers have been flushed.
Currently, the empty packet/frame is not handled correctly:
When encoding, the empty packet is forwarded to the muxer, usually
creating warnings.
When decoding, a reference to the memory is created anyway. Since in
the past this memory contained a decoded frame, it results in an extra
frame being decoded.
This commit discards the empty packet/frame.
References:
linux/Documentation/media/uapi/v4l/dev-decoder.rst:
"The last buffer may be empty (with :c:type:`v4l2_buffer` bytesused = 0)
and in that case it must be ignored by the client, as it does not
contain a decoded frame."
linux/Documentation/media/uapi/media/v4l/vidioc-encoder-cmd.rst:
"...This buffer may be empty, indicated by the
driver setting the ``bytesused`` field to 0."
Reviewed-by: Ming Qian <ming.qian@nxp.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
Fixes: out of array access
Fixes: 21469/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_G2M_fuzzer-5199357982015488
Alternatively the arrays can be made bigger or the index can be clipped.
In case a real file with such huge tiles exist we ask the user to upload it.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When the Ogg muxer writes a page, it has to do three things: It needs to
write a page header, then it has to actually copy the page data and then
it has to calculate and write a CRC checksum of both header as well as
data at a certain position in the page header.
To do this, the muxer used a dynamic buffer for both writing as well as
calculating the checksum via an AVIOContext's feature to automatically
calculate checksums on the data it writes. This entails an allocation of
an AVIOContext, of the opaque specific to dynamic buffers and of the
buffer itself (which may be reallocated multiple times) as well as
memcopying the data (first into the AVIOContext's small write buffer,
then into the dynamic buffer's big buffer).
This commit changes this: The page header is no longer written into a
dynamic buffer any more; instead the (small) page header is written into
a small buffer on the stack. The CRC is then calculated directly via
av_crc() on both the page header as well as the page data. Then both the
page header and the page data are written.
Finally, ogg_write_page() can now no longer fail, so it has been
modified to return nothing; this also fixed a bug in the only caller of
this function: It didn't check the return value.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
After this claim was made in e34e361 kamedo2 did an in-depth ABX
test comparing these encoders:
https://hydrogenaud.io/index.php?topic=111085.0
Result: FFmpeg AAC wasn't as good as libfdk_aac on average.
I know some things have changed since then such as, "use the fast
coder as the default" (fcb681ac) for example, so maybe the situation
is different now.
However, I am unaware of any recent comparison. So without any
substantiation we shouldn't make such a blantant claim.
Signed-off-by: Lou Logan <lou@lrcd.com>
Signed-off-by: Gyan Doshi <ffmpeg@gyani.pro>
Up until now the documentation of ff_bsf_get_packet_ref() allowed return
values >= 0 in case of success, whereas av_bsf_receive_packet() only
allows 0 on success. Given that for some bitstream filters the return
value of ff_bsf_get_packet_ref() is forwarded to the caller of
av_bsf_receive_packet() without any filtering, there would be a problem
if ff_bsf_get_packet_ref() actually returned values > 0. But it
currently doesn't and there is no reason why it should ever do so.
Therefore this commit aligns the return values of these functions by
restricting ff_bsf_get_packet_ref() to always returns 0 on success.
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Mainly includes reindentation and returning directly (i.e. without
a goto fail when possible).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
hls_init() would at first allocate the vtt_basename string, then
allocate the vtt_m3u8_name string followed by several operations that
may fail and then open the subtitles' output context. Yet upon freeing,
these strings were only freed when the subtitles' output context
existed, ensuring that they leak if something goes wrong between their
allocation and the opening of the subtitles' output context. So drop the
check for whether this output context exists.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This fixes memleaks in instances such as:
a) When an allocation fails at one of the two places in hls_init() where
the error is returned immediately without goto fail first.
b) When an error happens when writing the header.
c) When an allocation fails at one of the three places in
hls_write_trailer() where the error is returned immediately without goto
fail first.
d) When one decides not to write the trailer at all (e.g. because of
errors when writing packets).
Furthermore, it removes code duplication and allows to return
immediately, without goto fail first.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Several variables which are only used when the HLS_SINGLE_FILE flag is
unset have been set even when this flag is set. This has been changed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The Matroska specification allows multiple (level 1) Tags elements per
file, yet our demuxer didn't: While it parsed any amount of Tags
elements it found in front of the Clusters (albeit with warnings because
of duplicate elements), it would treat any Tags element only referenced
via a SeekHead entry as already parsed if any Tags element has already
been parsed; therefore this Tags element would not be parsed at all.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
There can be more than one SeekHead in a Matroska file, but most of the
other level 1 elements can only occur once.* Therefore the Matroska
demuxer only allows one entry per ID in its internal list of level 1
elements known to it; the only exception to this are SeekHeads.
The only exception to this are SeekHeads: When one is encountered
(either directly or in the list of entries read from SeekHeads),
a new entry in the list of known level-1 elements is always added,
even when this entry is actually already known.
This leads to lots of seeks in case of circular SeekHeads: Each time a
SeekHead is parsed, a new entry for a SeekHead will be added to the list
of entries read from SeekHeads. The exception for SeekHeads mentioned
above now implies that this SeekHead will always appear new and unparsed
and parsing will be attempted. This continued until the list of known
level-1 elements is full.
Fixing this is pretty simple: Don't add a new entry for a SeekHead if
its position matches the position of an already known SeekHead.
*: Actually, there can be multiple Tags and several other level 1
elements are "identically recurring" which means they may be resent
multiple times, but each instance must be absolutely identical to the
previous.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
A Seek element in a Matroska SeekHead should contain a SeekID and a
SeekPosition element and upon reading, they should be sanitized:
Given that IDs are restricted to 32 bit, longer SeekIDs should be treated
as invalid. Instead currently the lower 32 bits have been used.
For SeekPosition, no checks were performed for the element to be
present and if present, whether it was excessively large (i.e. the
absolute file position described by it exceeding INT64_MAX). The
SeekPosition element had a default value of -1 which means that a check
seems to have been intended; but it was not implemented. This commit adds
a check for overflow to the calculation of the absolute file position of
the referenced level 1 elements.
Using -1 (i.e. UINT64_MAX) as default value for SeekPosition implies that
a Seek element without SeekPosition will run afoul of this check.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
x264_encoder_encode can return 0 with nnal 0. As a result, encode_nals will
return 0. In this condition, it's better to return 0 immediately to avoid
the following unneeded pict_type and flags setting.
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Generic retime functionality is replaced by a few lines of code directly in the
muxers which used it, which seems a lot easier to understand and this way the
retiming is not dependant of the input durations.
Also remove retimeinterleave, since it is not used by anything anymore.
Signed-off-by: Marton Balint <cus@passwd.hu>
And rename it to retimeinterleave, use the pcm_rechunk bitstream filter for
rechunking.
By seperating the two functions we hopefully get cleaner code.
Signed-off-by: Marton Balint <cus@passwd.hu>
Previously only 1:1 bitstream filters were supported, the end of the stream was
not signalled to the bitstream filters and time base changes were ignored.
This change also allows muxers to set up bitstream filters regardless of the
autobsf flag during write_header instead of during check_bitstream and those
bitstream filters will always be executed.
Signed-off-by: Marton Balint <cus@passwd.hu>
avformat_alloc_output_context2() already sets the oformat member, so
that there is no reason to overwrite it again with the value it already
has.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Failures of the allocations that happen under the hood when using dynamic
buffers are usually completely unchecked and the Matroska muxer is no
exception to this.
The API has its part in this, because there is no documented way to
actually check for errors: The return value of both avio_get_dyn_buf()
as well as avio_close_dyn_buf() is only documented as "the length of
the byte buffer", so that using this to return errors would be an API
break.
Therefore this commit uses the only reliable way to check for errors
with avio_get_dyn_buf(): The AVIOContext's error flag. (This is one of
the advantages of avio_get_dyn_buf(): By not destroying the AVIOContext
it is possible to inspect this value.) Checking whether the size or the
pointer vanishes is not enough as it does not check for truncated output
(the dynamic buffer API is int based and so has to truncate the buffer
even when enough memory would be available; it's current actual limit is
even way below INT_MAX).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
If one already has the contents of a master elements in a buffer of
known size, then writing a EBML master element is no different from
writing an EBML binary element. It is overtly complicated to use
start/end_ebml_master() as these functions first write an unkown-length
size field of the appropriate length, then write the buffer's contents,
followed by a seek to the length field to overwrite it with the real
size (obtained via avio_tell() although it was already known in
advance), followed by another seek to the previous position. Just use
put_ebml_binary() instead.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
There is a good upper bound for the maximum length of the Colour master
element; it is therefore unnecessary to use a dynamic buffer for it.
A simple buffer on the stack is enough. This commit implements this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The Matroska muxer updates several header elements when the output is
seekable; if unseekable, the buffer containing the contents of the element
is immediately freed after writing. Before this commit, there were three
places doing exactly the same: Checking whether the output is seekable
and calling the function that writes and frees or the function that
just writes the EBML master. This has been unified; adding SeekHead
entries for these elements has been unified, too.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until now, SeekEntries were already added before
start_ebml_master_crc32() was even called and before we were actually
sure that we really write the element the SeekHead references: After
all, we might also error out later; and given that the allocations
implicit in dynamic buffers should be checked, end_ebml_master_crc32()
will eventually have to return errors itself, so that it is the right
place to add SeekHead entries.
The earlier behaviour is of course a remnant of the time in which
start_ebml_master_crc32() really did output something, so that the
position before start_ebml_master_crc32() needed to be recorded.
Erroring out later is also not as dangerous as it seems because in
this case no SeekHead will be written (if it happened when writing
the header, the whole muxing process would abort; if it happened
when writing the trailer (when writing chapters not available initially),
writing the trailer would be aborted and no SeekHead containing the
bogus chapter entry would be written).
This commit does not change the way the SeekEntries are added for those
elements that are output preliminarily; this is so because the SeekHead
is written before those elements are finally output and doing it
otherwise would increase the amount of seeks.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The mapping of streams to the various variant streams to be created by
the HLS muxer is roughly as follows: Space and tab separate variant
stream group maps while the entries in each variant stream group map are
separated by ','.
The parsing process of each variant stream group proceeded as follows:
At first the number of occurences of "a:", "v:" and "s:" in each variant
stream group is calculated so that one can can allocate an array of
streams with this number of entries. Then each entry is checked and the
check for stream numbers was deficient: It did check that there is a
number beginning after the ":", but it did not check that the number
extends until the next "," (or until the end).
This means that an invalid variant stream group like v:0_v:1 will not be
rejected; the problem is that the variant stream in this example is
supposed to have two streams associated with it (because it contains two
"v:"), yet only one stream is actually associated with it (because there
is no ',' to start a second stream specifier). This discrepancy led to
segfaults (null pointer dereferencing) in the rest of the code (when the
nonexistent second stream associated to the variant stream was inspected).
Furthermore, this commit also removes an instance of using atoi() whose
behaviour on a range error is undefined.
Fixes ticket #8652.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
fix ticket: 8651
because the init fragment mp4 file name is without base url name,
so just modify it use the full url which splice after init function.
Tested-by: matclayton
Signed-off-by: Steven Liu <liuqi05@kuaishou.com>
This patch adds possibility to use 'periodic-rekey' option with
multi-variant streams to hlsenc muxer. All streams variants
use parameters from the same key_info_file.
There are 2 sets of encryption options that kind of overlaps and add
complexity, so I tried to do the thing without changing too much code.
There is a little duplication of the key_file, key_uri, iv_string, etc
in the VariantStream since we copy it from hls to each variant stream,
but generally all the code remains the same to minimise appearing
of unexpected bugs. Refactoring could be done as a separate patch then as needed.
Signed-off-by: Yaroslav Pogrebnyak <yyyaroslav@gmail.com>
segment duration is using vs duration which compute by frame per second,
that can not fix problem of VFR video stream, so compute the duration
when split the segment, set the segment target duration use
current packet pts minus the prev segment end pts..
Reported-by: Zhao Jun <barryjzhao@tencent.com>
Reviewed-by: Zhao Jun <barryjzhao@tencent.com>
Signed-off-by: Steven Liu <liuqi05@kuaishou.com>
and make it static again.
These functions have been moved from nutenc to aviobuf and internal.h
in f8280ff4c0 in order to use them in a
forthcoming patch in utils.c. Said patch never happened, so this commit
moves them back and makes them static, effectively reverting said
commit as well as f8280ff4c0 (which added
the ff-prefix to these functions).
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
NUT uses variable-length integers in order to for length fields.
Therefore the NUT muxer often writes data into a dynamic buffer in order
to get the length of it, then writes the length field using the fewest
amount of bytes needed. To do this, a new dynamic buffer was opened,
used and freed for each element which involves lots of allocations. This
commit changes this: The dynamic buffers are now resetted and reused.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
With the description in frame size with refs semantics (SPEC 7.2.5),
it is a requirement of bitstream conformance that for at least one
reference frame has the valid dimensions.
Modify the check to make sure the decoder works well in SINGLE_REFERENCE
mode that not all reference frames have valid dimensions.
Check and error out if invalid reference frame is used in inter_recon.
One of the failure case is a 480x272 inter frame (SINGLE_REFERENCE mode)
with following reference pool:
0. 960x544 LAST valid
1. 1920x1088 GOLDEN invalid, but not used in single reference mode
2. 1920x1088 ALTREF invalid, but not used in single reference mode
3~7 ... Unused
Identical logic in libvpx:
<https://github.com/webmproject/libvpx/blob/master/vp9/decoder/vp9_decodeframe.c#L736>
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Hard coded parameters for qmin and qmax are currently used to initialize
v4l2_m2m device. This commit uses values from avctx->{qmin,qmax} if they
are set.
Reviewed-by: Ming Qian <ming.qian@nxp.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
ff_vaapi_encode_close() is not enough to free the resources like cbs
if initialization failure happens after codec->configure (except for
vp8/vp9).
We need to call avctx->codec->close() to deallocate, otherwise memory
leak happens.
Add FF_CODEC_CAP_INIT_CLEANUP for vaapi encoders and deallocate the
resources at free_and_end inside avcodec_open2().
Reviewed-by: Timo Rothenpieler <timo@rothenpieler.org>
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Since commit 979b5b8959, reverting the
Matroska ContentCompression is no longer done inside
matroska_parse_frame() (the function that creates AVPackets out of the
parsed data (unless we are dealing with certain codecs that need special
handling)), but instead in matroska_parse_block(). As a consequence,
the data that matroska_parse_frame() receives is no longer always owned
by an AVBuffer; it is owned by an AVBuffer iff no ContentCompression needed
to be reversed; otherwise the data is independently allocated and needs
to be freed on error.
Whether the data is owned by an AVBuffer or not is indicated by a variable
buf of type AVBufferRef *: If it is NULL, the data is independently
allocated, if not it is owned by the underlying AVBuffer (and is used to
avoid copying the data when creating the AVPackets).
Because the allocation of the buffer holding the uncompressed data happens
outside of matroska_parse_frame() (if a ContentCompression needs to be
reversed), the data is passed as uint8_t ** in order to not leave any
dangling pointers behind in matroska_parse_block() should the data need to
be freed: In case of errors, said uint8_t ** would be av_freep()'ed in
case buf indicated the data to be independently allocated.
Yet there is a problem with this: Some codecs (namely WavPack and
ProRes) need special handling: Their packets are only stored in
Matroska in a stripped form to save space and the demuxer reconstructs
full packets. This involved allocating a new, enlarged buffer. And if
an error happens when trying to wrap this new buffer into an AVBuffer,
this buffer needs to be freed; yet instead the given uint8_t ** (holding
the uncompressed, yet still stripped form of the data) would be freed
(av_freep()'ed) which certainly leads to a memleak of the new buffer;
even worse, in case the track does not use ContentCompression the given
uint8_t ** must not be freed as the actual data is owned by an AVBuffer
and the data given to matroska_parse_frame() is not the start of the
actual allocated buffer at all.
Both of these issues are fixed by always freeing the current data in
case it is independently allocated. Furthermore, while it would be
possible to track whether the pointer from matroska_parse_block() needs
to be reset or not, there is no gain in doing so, as the pointer is not
used at all afterwards and the sematics are clear: If the data passed
to matroska_parse_frame() is independently allocated, then ownership
of the data passes to matroska_parse_frame(). So don't pass the data
via uint8_t **.
Fixes Coverity ID 1462661 (the issue as described by Coverity is btw
a false positive: It thinks that this error can be triggered by ProRes
with a size of zero after reconstructing the original packets, but the
reconstructed packets can't have a size of zero).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Not requiring this leads to unexpected result, since Rav1e's current
two pass API has no way to fail in such a case.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
The previous code here did not handle passing a frames context when
ffmpeg itself did not know about the device it came from (for example,
because it was created by device derivation inside a filter graph), which
would break encoders requiring that input. Fix that by checking for HW
frames and device context methods independently, and prefer to use a
frames context method if possible. At the same time, revert the encoding
additions to the device matching function because the additional
complexity was not relevant to decoding.
Also fixes#8637, which is the same case but with the device creation
hidden in the ad-hoc libmfx setup code.
Extradata included in packet side data is meant to replace the codec context
extradata. So when muxing for example to MP4 without this change and if
extradata is present in a packet side data, the result will be that the
parameter sets present in keyframes will be filtered, but the parameter sets
ultimately included in the av1C box will not.
This is especially important for AV1 as both currently supported encoders don't
export the Sequence Header in the codec context extradata, but as packet side
data instead.
Signed-off-by: James Almer <jamrial@gmail.com>
This has previously only been checked if the chapters were initially
available, but not if they were only written in the trailer.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until now ff_vorbiscomment_write() used the bytestream API to write
VorbisComments. Therefore the caller had to provide a sufficiently large
buffer to write the output.
Yet two of the three callers (namely the FLAC and the Matroska muxer)
actually want the output to be written via an AVIOContext; therefore
they allocated buffers of the right size just for this purpose (i.e.
they get freed immediately afterwards). Only the Ogg muxer actually
wants a buffer. But given that it is easy to wrap a buffer into an
AVIOContext this commit changes ff_vorbiscomment_write() to use an
AVIOContext for its output.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
ff_vorbiscomment_write() used an AVDictionary ** parameter for a
dictionary whose contents ought to be written; yet this can be replaced
by AVDictionary * since commit 042ca05f0fdc5f4d56a3e9b94bc9cd67bca9a4bc;
and this in turn can be replaced by const AVDictionary * to indicate
that the dictionary isn't modified; the latter also applies to
ff_vorbiscomment_length().
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
If a FLAC track uses an unconventional channel layout, the Matroska
muxer adds a WAVEFORMATEXTENSIBLE_CHANNEL_MASK VorbisComment to the
CodecPrivate to preserve this information. And given that FLAC uses
24bit length fields, the muxer checks if the length is more than this
and errors out if it is.
Yet this can never happen, because we create the AVDictionary that is
the source for the VorbisComment. It only contains exactly one entry
that can't grow infinitely large (in fact, the length of the
VorbisComment is <= 4 + 33 + 1 + 18 + strlen(LIBAVFORMAT_IDENT)).
So we can simply assert the size to be < (1 << 24) - 4.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Commit 6fd300ac6c added support for WebM
Chunk livestreaming; in this case, both the header as well as each
Cluster is written to a file of its own, so that even if the AVIOContext
seems seekable, the muxer has to behave as if it were not. Yet one of
the added checks makes no sense: It ensures that no SeekHead is written
preliminarily (and hence no SeekHead is written at all) if the option
for livestreaming is set, although one should write the SeekHead in this
case when writing the Header. E.g. the WebM-DASH specification [1]
never forbids writing a SeekHead and in some instances (that don't apply
here) even requires it (if Cues are written after the Clusters).
[1]: https://sites.google.com/a/webmproject.org/wiki/adaptive-streaming/webm-dash-specification
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Since commit 4aa0665f39, the dynamic
buffer destined for the contents of the current Cluster is no longer
constantly allocated, reallocated and then freed after writing the
content; instead it is reset and reused when closing a Cluster.
Yet the code in mkv_write_trailer() still checked for whether a Cluster
is open by checking whether the pointer to the dynamic buffer is NULL or
not (instead of checking whether the position of the current Cluster is
-1 or not). If a Cluster was not open, an empty Cluster would be output.
One usually does not run into this issue, because unless there are
errors, there are only three possibilities to not have an opened Cluster
at the end of writing a packet:
The first is if one sent an audio packet to the muxer. It might trigger
closing and outputting the old Cluster, but because the muxer caches
audio packets internally, it would not be output immediately and
therefore no new Cluster would be opened.
The second is an audio packet that does not contain data (such packets
are sometimes sent for side-data only, e.g. by the FLAC encoder). The
only difference to the first scenario is that such packets are not
cached.
The third is if one explicitly flushes the muxer by sending a NULL
packet via av_write_frame().
If one also allows for errors, then there is also another possibility:
Caching the audio packet may fail in the first scenario.
If one calls av_write_trailer() after the first scenario, the cached
audio packet will be output when writing the trailer, for which
a Cluster is opened and everything is fine; because flushing the muxer
does currently not output the cached audio packet (if one is cached),
the issue also does not exist if an audio packet has been cached before
flushing. The issue only exists in one of the other scenarios.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
It's based on the following specs:
RDD 45:2017 - SMPTE Registered Disclosure Doc - Interoperable Master Format - Application ProRes
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
It's based on the following specs:
RDD 36:2015 - SMPTE Registered Disclosure Doc - Apple ProRes Bitstream Syntax and Decoding Process
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
linear. Instead of mixing these in the calculations, convert the former
first to have all following calculations in the same unit.
Signed-off-by: Kyle Swanson <k@ylo.ph>
The old approach used some highly complex delta computation math and
output-delaying.
I do not remember what the initial reasoning behind that was, but given
that we can just offset the dts by the amount of bframes, it seems wholy
unnecessary.
This leaves open an issue with VFR content, for which some more complex
logic might be needed.
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
Due to a typo, it was impossible to write 0.595 / -4.5 dB
of ltrt_cmixlev, ltrt_surmixlev, loro_cmixlev, loro_surmixlev.
Without any error 0.841 / -1.5 dB was written to file.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Rav1e currently uses the time base given to it only for ratecontrol... where
the inverse is taken and used as a framerate. So, do what we do in other wrappers
and use the framerate if we can.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Sequence numbers of segments should be unique, if an encoder is using shorter
than 1 second segments and it is restarted, then future segments will be using
already used sequence numbers if initial sequence number is based on the number
of seconds since epoch and not microseconds.
Signed-off-by: Marton Balint <cus@passwd.hu>
Fixes problems when non-rational options were set using rational expressions,
causing rounding errors and the option range limits not to be enforced
properly.
ffmpeg -f lavfi -i "sine=r=96000/2"
This caused an assertion failure with assert level 2.
Signed-off-by: Marton Balint <cus@passwd.hu>
Reindentation as well as marking several variables used for demuxing
RealAudio as const to clearly see that they don't change during
demuxing.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The Matroska demuxer has three functions for creating packets out of
the data read: One for certain RealAudio codecs (ATRAC3, cook, sipr,
RealAudio 28.8), one for WebVTT (actually, the WebM flavour of it) and
one for all the others. Only the last function supported Matroska's
ContentCompression (e.g. it reversed zlib compression or added the
removed headers to the packets). But in Matroska, all tracks are allowed
to be compressed. This commit adds support for this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Matroska is built around the principle that a reader does not need to
understand everything in a file in order to be able to make use of it;
it just needs to ignore the data it doesn't know about.
Our demuxer typically follows this principle, but there is one important
instance where it does not: A Block belonging to a TrackEntry with no
associated stream is treated as invalid data (i.e. the demuxer will try
to resync to the next level 1 element because it takes this as a sign
that it has lost sync). Given that we do not create streams if we don't
know or don't support the type of the TrackEntry, this impairs this
demuxer's forward compability.
Furthermore, ignoring Blocks belonging to a TrackEntry without
corresponding stream can (in future commits) also be used to ignore
TrackEntries with obviously bogus entries without affecting the other
TrackEntries (by not creating a stream for said TrackEntry).
Finally, given that matroska_find_track_by_num() already emits its own
error message in case there is no TrackEntry with a given TrackNumber,
the error message (with level AV_LOG_INFO) for this can be removed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
A Block (meaning both a Block in a BlockGroup as well as a SimpleBlock)
must have at least three bytes after the field containing the encoded
TrackNumber. So if there are <= 3 bytes, the Matroska demuxer would
skip this block, believing it to be an empty, but valid Block.
This might discard valid nonempty Blocks, namely if the track uses header
stripping. And certain definitely spec-incompliant Blocks don't raise
errors: Those with two or less bytes left after the encoded TrackNumber
and those with three bytes left, but with flags indicating that the Block
uses lacing as then there has to be further data describing the lacing.
Furthermore, zero-sized packets were still possible because only the
size of the last entry of a lace was checked.
This commit fixes this. All spec-compliant Blocks that contain data
(even if side data only) are now returned to the caller; spec-compliant
Blocks that don't contain anything are not returned.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Some conditions which don't change and which can therefore be checked
in read_header() were instead rechecked upon parsing each block. This
has been changed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The Matroska demuxer splits every sequence of h Matroska Blocks into
h * w / cfs packets of size cfs; here h (sub_packet_h), w (frame_size)
and cfs (coded_framesize) are parameters from the track's CodecPrivate.
It does this by splitting the Block's data in h/2 pieces of size cfs each
and putting them into a buffer at offset m * 2 * w + n * cfs where
m (range 0..(h/2 - 1)) indicates the index of the current piece in the
current Block and n (range 0..(h - 1)) is the index of the current Block
in the current sequence of Blocks. The data in this buffer is then used
for the output packets.
The problem is that there is currently no check to actually guarantee
that no uninitialized data will be output. One instance where this is
trivially so is if h == 1; another is if cfs * h is so small that the
input pieces do not cover everything that is output. In order to
preclude this, rmdec.c checks for h * cfs == 2 * w and h >= 2. The
former requirement certainly makes much sense, as it means that for
every given m the input pieces (corresponding to the h different values
of n) form a nonoverlapping partition of the two adjacent frames of size w
corresponding to m. But precluding h == 1 is not enough, other odd
values can cause problems, too. That is because the assumption behind
the code is that h frames of size w contain data to be output, although
the real number is h/2 * 2. E.g. for h = 3, cfs = 2 and w = 3 the
current code would output four (== h * w / cfs) packets. although only
data for three (== h/2 * h) packets has been read.
(Notice that if h * cfs == 2 * w, h being even is equivalent to
cfs dividing w; the latter condition also seems very reasonable:
It means that the subframes are a partition of the frames.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
RealAudio 28.8 (like other RealAudio codecs) uses a special demuxing
mode in which the data of the existing Matroska Blocks is not simply
forwarded as-is. Instead data from several Blocks is recombined
together to output several packets. The parameters governing this
process are parsed from the CodecPrivate: Coded framesize (cfs), frame
size (w) and sub_packet_h (h).
During demuxing, h/2 pieces of data of size cfs each are read from every
Matroska (Simple)Block and put at offset m * 2 * w + n * cfs of a buffer
of size h * w, where m ranges from 0 to h/2 - 1 for each Block while n
is initially zero and incremented after a Block has been parsed until it
is h, at which poin the assembled packets are output and n reset.
The highest offset is given by (h/2 - 1) * 2 * w + (h - 1) * cfs + cfs
while the destination buffer's size is given by h * w. For even h, this
leads to a buffer overflow (and potential segfault) if h * cfs > 2 * w;
for odd h, the condition is h * cfs > 3 * w.
This commit adds a check to rule this out.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
RealAudio 28.8 does not need or use sub_packet_size for its demuxing
and this field is therefore commonly set to zero. But since 18ca491b
the Real Audio specific demuxing is no longer applied if sub_packet_size
is zero because the codepath for cook and ATRAC3 divide by it; this made
these files undecodable.
Furthermore, since 569d18aa (merged in 2c8d876d) sub_packet_size being
zero is used as an indicator for invalid data, so that a file containing
such a track was completely skipped.
This commit fixes this by not checking sub_packet_size for RealAudio
28.8 at all.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
They need a special parsing mode and in order to find out whether this
mode is in use, several checks have to be performed. They can all be
combined into one: If the buffer that is only used to assemble their
packets has been allocated, use the RealAudio parsing mode.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Only flavors 0..3 seem to exist. E.g. rmdec.c treats any flavor > 3
as invalid data. Furthermore, we do not know how big the packets to
create ought to be given that for sipr these values are not read from
the bitstream, but from a table.
Furthermore, flavor is only used for sipr, so only check it for sipr;
rmdec.c does the same. (The old check for flavor being < 0 was
always wrong given that flavor is an int that is read via avio_rb16(),
so it has been removed completely.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This makes decoding far more robust, since OggS, the ogg magic,
can be commonly found randomly in streams, which previously made
the demuxer think there's a new stream or a change in such.
v4l2_m2m uses device memory mapped buffers to store dequeued
frames/packets (reference counted by AVBufferRef). When the reference
count drops to zero, the buffer ownership is returned back to the
device, so that they can re-filled with frames/packets.
There are some cases when all the capture buffers are in userspace
(i.e. due to internal buffering in ffmpeg). On the s5p-mfc this causes
an infinite wait when polling to dequeue the buffers, which can be
prevented by increasing the total number of buffers. This commit adds a
warning when all the capture buffers are dequeued.
Reviewed-by: Ming Qian <ming.qian@nxp.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
Apparently the changes from 3c9185bf3a
aren't enough; even with that in place, I got errors like this
when trying to build for iOS:
src/libavdevice/avfoundation.m:135:5: error:
'AVCaptureDeviceTransportControlsPlaybackMode' is unavailable: not
available on iOS
AVCaptureDeviceTransportControlsPlaybackMode observed_mode;
^
The data of an AVPacket may be a part of the data of an AVBufferRef;
Therefore av_grow_packet() doesn't reallocate if the available space in
the actual buffer is sufficient for the enlargement. But given that it
also zeroes the padding it also needs to make sure that the buffer is
actually writable; this commit implements this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
hdsenc already had an explicit function to free all allocations in case
of an error, but it was not marked as deinit function, so that it was
not called automatically when the AVFormatContext for muxing gets freed.
Using an explicit deinit function also makes the code cleaner by
allowing to return immediately without "goto fail".
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Enqueue/dequeue of the capture buffers should continue while draining.
Reference: linux/Documentation/media/uapi/v4l/dev-decoder.rst
"The client must continue to handle both queues independently,
similarly to normal decode operation. This includes:
...
- queuing and dequeuing CAPTURE buffers, until a buffer marked with
the V4L2_BUF_FLAG_LAST flag is dequeued"
Signed-off-by: Ming Qian <ming.qian@nxp.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
The DNG code hardcodes plane 0 at some places, so its better to disallow cases
that have more planes.
Fixes: eg_crash
Found-by: 黄宁 <tsukimurarin@163.com>
Reviewed-by: Nick Renieris <velocityra@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
dng spec 1.5.0.0
"BitsPerSample
Supported values are from 8 to 32 bits/sample. The depth must be the same for each sample if
SamplesPerPixel is not equal to 1."
Fixes: eg_crash
Found-by: 黄宁 <tsukimurarin@163.com>
Reviewed-by: Nick Renieris <velocityra@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
"slice mode" option seems to be unnecessary since it could be
determined by -slices/max_nal_size.
default: SM_FIXEDSLCNUM_SLICE mode with cpu-number slices.
-slices N: SM_FIXEDSLCNUM_SLICE mode with N slices.
-max_nal_size: SM_SIZELIMITED_SLICE mode with limited size slices.
Add FF_API_OPENH264_SLICE_MODE macro to remove this option after
LIBAVCODEC_VERSION_MAJOR = 59.
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
Libopenh264enc would set the slice according to the number of cpu cores
if uiSliceNum equals to 0 (auto) in SM_FIXEDSLCNUM_SLICE mode.
Prompt a warning for user to catch this.
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
RC_BITRATE_MODE:
set BITS_EXCEEDED to iCurrentBitsLevel and allows QP adjust
in RcCalculatePictureQp().
RC_BUFFERBASED_MODE:
use buffer status to adjust the video quality.
RC_TIMESTAMP_MODE:
bit rate control based on timestamp, introduced in release 1.4.
Default to use RC_QUALITY_MODE.
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
It would be 200kbps bitrate with gop size = 12 by default
which generated too many IDR frames in rather low bit rate.
The quality would be poor.
Set these default values to -1 to check whether it's specified
by user explicitly.
Use 2Mbps bitrate as nvenc sugguested, and leave gop size
untouched in libopenh264.
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
Clip iMinQp/iMaxQp to (1, 51) for user specified qp range.
If not set, leave iMinQp/iMaxQp untouched and use the values (0, 51)
initialized in FillDefault(), and the QP range would be adjusted to the
defaults inside libopenh264 library according to the iUsageType, (12, 42)
for iUsageType == CAMERA_VIDEO_REAL_TIME which is default.
<https://github.com/cisco/openh264/blob/master/codec/encoder/core/src/encoder_ext.cpp#L375>
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
Because not every user know about in_pad and out_pad reasonable value range
so maybe try to set 1.0, but setting 1.0 is so hugh to get an fatal error.
Suggested-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
The current implementation allows passing levels to libavcodec as
integers (such as "31" instead of "3.1").
However, in this case, the maximum reference frame value per level was
ignored because libavcodec converted the string to 310 instead of 31.
Since libx264 has correctly parsed the level to int
(x4->params.i_level_idc), we should rely on this value instead of
attempting to parse the level string on our own.
Signed-off-by: Josh Brewster <josh.brewster@protonmail.com>
Signed-off-by: Josh de Kock <josh@itanimul.li>
Without this setting, bitrate and qmin/qmax options have no
effect on the s5p-mfc hardware encoder.
Reviewed-by: Ming Qian <ming.qian@nxp.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
Currently the user gets unhelpful warnings when some default parameters
are not supported by the device. The verbosity of these log messages has
been changed to AV_LOG_DEBUG.
Reviewed-by: Ming Qian <ming.qian@nxp.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
libswscale/vscale.c makes extensive use of function pointers and in
doing so it converts these function pointers to and from a pointer to
void. Yet this is actually against the C standard:
C90 only guarantees that one can convert a pointer to any incomplete
type or object type to void* and back with the result comparing equal
to the original which makes pointers to void generic pointers to
incomplete or object type. Yet C90 lacks a generic function pointer
type.
C99 additionally guarantees that a pointer to a function of one type may
be converted to a pointer to a function of another type with the result
and the original comparing equal when converting back.
This makes any function pointer type a generic function pointer type.
Yet even this does not make pointers to void generic function pointers.
Both GCC and Clang emit warnings for this when in pedantic mode.
This commit fixes this by using a union that can hold one member of any
of the required function pointer types to store the function pointer.
This works even for C90.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Current muxers only use a single bitstream filter, so there is no need to
maintain code which operates on a list of bitstream filters. When multiple
bitstream filters are needed muxers can simply use a list bitstream filter.
If there is a use case in the future when different bitstream filters should be
added at subsequent packets then a new API possibly involving reconfiguring the
list bitstream filter can be added knowing the exact requirements.
Signed-off-by: Marton Balint <cus@passwd.hu>
This can support encoders which want frames and/or device contexts. For
the device case, it currently picks the first initialised device of the
desired type to give to the encoder - a new option would be needed if it
were necessary to choose between multiple devices of the same type.
All of these encoders can accept libmfx surfaces directly in a hardware
frames context, or they can accept software frames if a suitable device
is supplied to use.
Fixes: left shift of 1073741824 by 1 places cannot be represented in type 'int32_t' (aka 'int')
Fixes: 21245/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DIRAC_fuzzer-5683334274613248
Change to int16_t suggested by Lynne
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Lynne <dev@lynne.ee>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
bump minor version for DOVI sidedata, because added the dovi_meta.h
as lavu API part. Also update APIchanges.
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
This fixes tests on 32 bit x86 mingw with clang, which uses x87
fpu by default.
In this setup, while the get_expected function is declared to
return float, the compiler is (especially given the optimization
flags set) free to keep the intermediate values (in this case,
the return value from the inlined function) in higher precision.
This results in the situation where 7.28 (which actually, as
a float, ends up as 7.2800002098), multiplied by 100, is
728.000000 when really forced into a 32 bit float, but 728.000021
when kept with higher intermediate precision.
For the multiplication case, a more suitable epsilon would e.g.
be 2*FLT_EPSILON*fabs(expected_output), but just increase the
current hardcoded threshold for now.
Signed-off-by: Martin Storsjö <martin@martin.st>
libavutil/mem.h is unneeded since 33d18982fa,
the commit that introduced the new packet-based bsf API, because with
this switch the allocations were no longer performed directly, but by
av_new_packet().
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
With 33d18982fa, the commit introducing
the new packet-based bsf API, a new buffer was no longer allocated
directly, but via av_new_packet(), so that libavutil/mem.h was no longer
needed.
Moreover since commit dc99ee6b08
av_packet_make_writable() is employed which copies the data in case it
is unavoidable so that string.h is no longer used (it was used for
memcpy()).
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Since 33d18982fa (the commit that
introduced the new bsf API) allocating an enlarged buffer in case
extradata needs to be added to a packet is done via av_new_packet(),
so that libavutil/mem.h is no longer needed.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
mux.c was split from utils.c in 55f9037f38
and during this split all headers were simply copied without checking if
they were only needed in the part that stayed in utils.c (or whether
these haeders were needed at all). As a result quite a lot of headers
in mux.c are unnecessary. This commit removes them.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
stdarg.h has been included in 780d7897a9
for ff_url_join(). This header became unnecessary when this function was
moved into a separate file in df9f22d42b.
libavutil/pixdesc.h has been included for av_get_pix_fmt_name() in
603b8bc2a1 and is unused since commit
2fb7501938 that removed the stuff belonging
to FF_API_FORMAT_PARAMETERS. Notice that this file still uses
AV_PIX_FMT_NONE and that therefore the header libavutil/pixfmt.h has
been included (this header is included in pixdesc.h as well as also in
libavutil/internal.h which is also included).
libavutil/time_internal.h has been included for gmtime_r() in commit
e7dd97b5d8cd6ea150446591f37a5946e8ab7cfb; it is unused since commit
b72a7b96f8 which basically moved the code
making use of gmtime_r() to libavutil/dict.c to use in
avpriv_dict_set_timestamp().
audiointerleave.h has been added in c26e58e32c
because of ff_interleave_compare_dts() (at that time the muxing code
was not split from utils.c yet); said function became static in commit
101e1f6ff9, making this header redundant.
metadata.h has been mostly included for what now resides in
libavutil/dict.h. The stuff that now resides in metadata.h has only been
used briefly: From commits ed7694d8cf to
d60a9f52eb.
riff.h has been added in 45da8124a0
because riff.h once contained declarations for (ff_)codec_get_tag().
This was changed in bfe5454cd2.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The documentation of av_dict_set() states that values >= 0 indicate
success, whereas av_packet_unpack_dictionary() implies that return
values > 0 are impossible. So only forward the return value of
av_dict_set() in av_packet_unpack_dictionary() on error.
(Btw: av_dict_set() does currently not return values > 0.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This patch adds support for decoding images
with a Region of Interest. Allows decoding
samples such as p0_03.j2k. This patch should
fix ticket #4681.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Adding the support to build FFMPEG with HW accelerated decode(nvdec) and
encode on aarch64 architecture.
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
support DOVI Video Stream Descriptor from Dolby Vision Streams
Within the MPEG-2 Transport Stream Format V1.2
From the spec: https://www.dolby.com/us/en/technologies/\
dolby-vision/dolby-vision-bitstreams-in-mpeg-2-transport-\
stream-multiplex-v1.2.pdf.
export the DOVI information with sidedata.
Signed-off-by: vacingfang <vacingfang@tencent.com>
This patch adds support to skip the CRG marker.
The CRG marker, is an informational marker.
Allows samples such as p0_03.j2k to be decoded.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
We derive the destination buffer stride from the input stride,
which meant if the image was flipped with a negative stride,
we'd be FFALIGNING a negative number which ends up being huge,
thus making the Vulkan buffer allocation fail and the whole
image transfer fail.
Only found out about this as OpenGL compositors can copy an entire
image with a single call if its flipped, rather than iterate over
each line.
Reindentation, removal of { } if they contain only one statement
and moving the return statement to a line of its own in situations
like "if (ret < 0) return ret;". Moreover, several overlong lines
were made shorter and a camelCase variable received a name in line
with our naming conventions.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until now, the Matroska muxer would mark a track as default if it had
the disposition AV_DISPOSITION_DEFAULT or if there was no track with
AV_DISPOSITION_DEFAULT set; in the latter case even more than one track
of a kind (audio, video, subtitles) was marked as default which is not
sensible.
This commit changes the logic used to mark tracks as default. There are
now three modes for this:
a) In the "infer" mode the first track of every type (audio, video,
subtitles) with default disposition set will be marked as default; if
there is no such track (for a given type), then the first track of this
type (if existing) will be marked as default. This behaviour is inspired
by mkvmerge. It ensures that the default flags will be set in a sensible
way even if the input comes from containers that lack the concept of
default flags. This mode is the default mode.
b) The "infer_no_subs" mode is similar to the "infer" mode; the
difference is that if no subtitle track with default disposition exists,
no subtitle track will be marked as default at all.
c) The "passthrough" mode: Here the track will be marked as default if
and only the corresponding input stream had disposition default.
This fixes ticket #8173 (the passthrough mode is ideal for this) as
well as ticket #8416 (the "infer_no_subs" mode leads to the desired
output).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
At the end of encoding, the FLAC encoder sends a packet whose side data
contains updated extradata (e.g. a correct md5 checksum). The Matroska
muxer uses this to update the CodecPrivate.
In doing so, the stream's codecpar was copied. But given that writing
a FLAC CodecPrivate does not modify the used AVCodecParameters at all,
there is no need to do so and this commit changes this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Several EBML Master elements for which a good upper bound of the final
length was available were nevertheless written without giving an
upper bound of the final length to start_ebml_master(), so that their
length fields were eight bytes long. This has been changed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The Matroska muxer does not write every stream as a Matroska track;
some streams are written as AttachedFile. But should no stream be
written as a Matroska track, the Matroska muxer would nevertheless
write a Tracks element without a TrackEntry. This is against the spec.
This commit changes this and only writes a Tracks if there is a Matroska
track.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
As WebM doesn't support Attachments, the Matroska muxer drops them when
in WebM mode. This happened silently until this commit which adds a
warning for this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
In order to determine whether the current Cluster needs to be closed
because of the limits on clustersize and clustertime,
mkv_write_packet() would first get the size of the current Cluster by
applying avio_tell() on the dynamic buffer holding the current Cluster.
It did this without checking whether there is a dynamic buffer for
writing Clusters open right now.
In this case (which happens when writing the first packet)
avio_tell() returned AVERROR(EINVAL); yet it is not good to rely on
avio_tell() (or actually, avio_seek()) to handle the situation
gracefully.
Fixing this is easy: Only check whether a Cluster needs to be closed
if a Cluster is in fact open.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
When creating DASH streams, the TrackNumber is externally prescribed
and not derived from the number of streams in the AVFormatContext, so
if the number of tracks for a file using an explicit TrackNumber was
more than one, the resulting file would be broken (it would be impossible
to tell to which track a Block belongs if different tracks share the
same TrackNumber). So disallow this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The Matroska muxer currently only adds CuePoints in three cases:
a) For video keyframes. b) For the first audio frame in a new Cluster if
in DASH-mode. c) For subtitles. This means that ordinary Matroska audio
files won't have any Cues which impedes seeking.
This commit changes this. For every track in a file without video track
it is checked and tracked whether a Cue entry has already been added
for said track for the current Cluster. This is used to add a Cue entry
for each first packet of each track in each Cluster.
Implements #3149.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The Matroska file format has practically no limit on the number of
tracks (the current limit is 2^56 - 1); yet because they are encoded in
a variable length format in (Simple)Blocks this muxer has simply imposed
a limit on the number of tracks in order to ensure that they can always
be written on one byte in order to simplify the muxing process.
This commit removes said limit.
Also, zero is an invalid TrackNumber, so disallow this value in the
dash_track_number option.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This commit factors the ability to write ordinary EBML numbers out of
the functions for writing EBML lengths. This is in preparation for
future commits.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
EBML uses variable length integers both for the EBML IDs as well as for
the EBML lengths; Matroska also uses them for the TrackNumber in
(Simple)Blocks and for the lengths of laces when EBML lacing is used.
When encoding EBML lengths, certain encodings have a special meaning,
namely that the element has an unknown length. This is not so when
encoding general EBML variable length integers.
Yet the functions called ebml_num_size() and put_ebml_num() had this
special meaning hardcoded, i.e. they are there to write EBML lengths and
not general EBML numbers. So rename them.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Matroska (or actually EBML) uses variable-length numbers where only
seven bits of every byte is usable for the length; the other bits encode
the length of the variable-length number. So in order to find out how
many bytes one needs to encode a given number one can use a loop like
while (num >> 7 * bytes) bytes++; the Matroska muxer effectively did this.
Yet it has a disadvantage: It is impossible for the result of a single
right shift of an unsigned number with most significant bit set to be
zero, because one can only shift by 0..(width - 1). On some
architectures like x64 it is not even possible to do it with undefined
right shifts in which case this leads to an infinite loop.
This can be easily avoided by switching to a loop whose condition is
(num >>= 7). The maximum value the so modified function can return
is 10; any value > 8 is invalid and will now lead to an assert in
put_ebml_num() or in start_ebml_master() (or actually in
put_ebml_size_unknown()).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Use the mime_types of the corresponding AVCodecDescriptor instead of
tables specific to Matroska. The former are generally more encompassing:
They contain every item of the current lists except "text/plain" for
AV_CODEC_ID_TEXT and "binary" for AV_CODEC_ID_BIN_DATA.
The former has been preserved by special-casing it while the latter is
a hack added in c9212abf so that the demuxer (which uses the same tables)
sets the appropriate CodecID for broken files ("binary" is not a correct
mime type at all); using it for the muxer was a mistake. The correct
mime type for AV_CODEC_ID_BIN_DATA is "application/octet-stream" and
this is what one gets from the AVCodecDescriptor.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Found while reviewing a patch fixing a similar issue
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Decoding can be handled directly in the output frame.
Also ensure flushing cleans the reference frame in all cases.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: James Almer <jamrial@gmail.com>
For FLAC, Speex, Opus and VP8 the Ogg muxer allocates two buffers
for building the headers: The first for extradata in an Ogg-specific
format and the second contains a Vorbiscomment. These buffers are
reachable via pointers in the corresponding AVStream's priv_data.
If an error happens during building the headers, the AVStream's
priv_data would be freed. This is pointless in general as it would be
freed generically anyway, but here it is actively harmful: If the second
of the aforementioned allocations fails, the first buffer would leak
upon freeing priv_data.
This commit stops freeing priv_data manually, which allows the muxer to
properly clean up in the deinit function.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
avformat_find_stream_info() may decode some frames to get stream
information. And when it does this for subtitles, the decoded subtitles
leak.
(Decoding subtitles was added in b1511e00f6
for PGS subtitles. When PGS subtitles originate from a container that
exports every segment as a packet of its own, no output will be
generated when decoding a packet, because not enough input is available.
Yet when used with PGS subtitles in the Matroska form a single packet
contains enough data to generate output. Yet said output is not freed,
hence this leak.)
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
In this example, the difference in length between the shortest and
longest string is three, so that not using pointers to strings saves
space even on 32bit systems.
Moreover, there is no need to use a sentinel here; it can be replaced
with FF_ARRAY_ELEMS.
Reviewed-by: Ross Nicholson <phunkyfish@gmail.com>
Reviewed-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Fix an occasional crash for hevc decoder in ARM 32 platform, the
root cause is the memory over read(read cross the memory boundary)
in SAO NENO functions ff_hevc_sao_band_filter_neon_8 and
ff_hevc_sao_edge_filter_neon_8.
After this fix, the crash disapper in the massive Android phone
test.
Signed-off-by: qoroliang <qoroliang@tencent.com>
Protocol options like buffer_size need to be passed to the
underlying transport implementation for udp multicasts as well.
Signed-off-by: Marton Balint <cus@passwd.hu>
The x18 is a reserved platform register on Darwin and Windows.
x8/w8 seems to be unused in this function though (and same about
x10 and x14), so there's really no reason to use x18 here - just change
the uses of x18/w18 into x8/w8 instead without any further rewrites.
Signed-off-by: Martin Storsjö <martin@martin.st>
The context structure of the truehd_core bsf had a pointer to a const
AVClass as its first member; yet this bsf does not have any AVClass
associated with it, so that this pointer is always NULL. So remove it.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Currently Musepack allocates an array that needs to be freed later in
the demuxer's read_close-function; it is the sole reason for said
function's existence. But it is unnecessary, because one can store this
array in the stream's priv_data pointer, so that it will be freed
generically.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Chapter titles are added to the chapter's metadata since 6cb6e159,
yet since 012867f0 (the predecessor of) avpriv_new_chapter() already
adds the title to the chapter's metadata. So setting it again in
matroskadec.c is redundant and expensive.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Moreover, putting the Cues in front of the Clusters by reserving space
in advance is also tested.
The new capability of using ffprobe during a remux/transcode test are
used here for information about the chapters.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This is primarily intended to test that muxers correctly write chapters
or metadata; but given that it does this by having our demuxers read the
generated files, it also tests demuxers. And of course it may prove
useful for encoders, too.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until now, they were appended to the FATE_EXTERN-$(CONFIG_FFMPEG)
variable and were therefore activated when ffmpeg was enabled regardless
of whether ffprobe was enabled.
Also the same happened with FATE_SAMPLES_FASTSTART, although the
corresponding test (mov-faststart-4gb-overflow) only requires external
samples.
Furthermore, remove the unused FATE_FULL variable (FATE_EXTERN_FFPROBE has
taken its place).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
It is a small buffer of a known, fixed size and so it should simply be
put into the muxer's context.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
An AVStream's codecpar is supposed to be filled by the caller before
avformat_write_header(); if the CodecParameters change, the caller
should signal this via packet side data, but not touch the AVStream's
codecpar.
The FLAC muxer checks for packet side data containing updated extradata,
yet if nothing has arrived by the time the trailer is written, the
already written extradata is overwritten by the very same extradata
again, unless the output is unseekable, in which case a warning that the
FLAC header can't be rewritten is emitted.
This commit changes this by only trying to rewrite the extradata if a
new streaminfo arrived via packet side data. Only then is a warning
emitted in case the output is unseekable.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
When no packet could be output, the interleavement functions
nevertheless initialized the packet destined for output (with the
exception of the data and size fields, making the initialization
pointless), although it will not be used at all. So remove the
initializations.
Reviewed-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
write_packet() currently saves the original timestamps of the packet it
got and restores them in case writing fails. This is unnecessary as we
are no longer working directly with the user-supplied AVPacket here; and
it is also pointless because the timestamps may already have been
altered before write_packet().
So remove this and add a general comment to the function that timestamps
may be modified; also remove a long outdated comment about side data.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The documentation of av_write_frame() explicitly states that the function
doesn't take ownership of the packets sent to it; while av_write_frame()
does not directly unreference the packets after having written them, it
nevertheless modifies the packet in various ways:
1. The timestamps might be modified either by prepare_input_packet() or
compute_muxer_pkt_fields().
2. If a bitstream filter gets applied, it takes ownership of the
reference and the side-data in the packet sent to it.
In case of do_packet_auto_bsf(), the end result is that the returned packet
contains the output of the last bsf in the chain. If an error happens,
a blank packet will be returned; a packet may also simply not lead to
any output (vp9_superframe).
This also implies that side data needs to be really copied and can't be
shared with the input packet.
The method choosen here minimizes copying of data: When the input isn't
refcounted and no bitstream filter is applied, the packet's data will
not be copied.
Notice that packets that contain uncoded frames are exempt from this
because these packets are not owned by and returned to the user. This
also moves unreferencing the packets containing uncoded frames to
av_write_frame() in the noninterleaved codepath; in the interleaved
codepath, these packets are already freed in av_interleaved_write_frame(),
so that unreferencing the packets in write_uncoded_frame_internal() is
no longer needed. It has been removed.
Reviewed-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Now that ff_interleave_add_packet() always returns blank packets, the
input packet to ff_interleave_packet_per_dts() will always be blank on
return as well (if supplied) and the same goes for interleave_packet()
in mux.c. Document these facts and remove the redundant resetting that
happened in av_interleaved_write_frame().
The last reference to the (long removed) destruct field that AVPackets
once had has been removed as well when updating the documentation of
ff_interleave_packet_per_dts().
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
When an error happened in ff_interleave_add_packet() when adding
a packet to the packet queue, said packet would not be unreferenced
in ff_interleave_add_packet(), but would be zeroed in
av_interleaved_write_frame(), which results in a memleak.
This has been fixed: ff_interleave_add_packet() now always unreferences
the input packet on error; as a result, it always returns blank packets
which has been documented. Relying on this a call to av_packet_unref()
in ff_audio_rechunk_interleave() can be removed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
If writing uncoded frames in noninterleaved mode fails at the preparatory
steps (i.e. before it reaches write_packet()), the packet would not be
unreferenced and the frame would leak. This is fixed by unreferencing
the packet in write_uncoded_frame_internal() instead.
This also makes it possible to remove the unreferencing in
write_packet() itself: In noninterleaved mode frames are now freed in
write_uncoded_frame_internal(), while they are freed in interleaved
mode when their containing packet gets unreferenced (like normal
packets).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Currently uncoded frames (i.e. packets whose data actually points to an
AVFrame) are not refcounted. As a consequence, calling av_packet_unref()
on them will not free them, but may simply make sure that they leak by
losing the pointer to the frame.
This commit changes this by actually making uncoded frames refcounted.
In order not to rely on sizeof(AVFrame) (which is not part of the public
API and so must not be used here in libavformat) the packet's data is
changed to a (padded) buffer containing just a pointer to an AVFrame.
Said buffer is owned by an AVBuffer with a custom free function that
frees the frame as well as the buffer. Thereby the pointer/the AVBuffer
owns the AVFrame.
Said ownership can actually be transferred by copying and resetting
the pointer, as might happen when actually writing the uncoded frames
in AVOutputFormat.write_uncoded_frame() (although currently no muxer
makes use of this possibility).
This makes packets containing uncoded frames compatible with
av_packet_unref(). This already has three advantages in interleaved mode:
1. If an error happens at the preparatory steps (before the packet is
put into the interleavement queue), the frame is properly freed.
2. If the trailer is never written, the frames still in the
interleavement queue will now be properly freed by
ff_packet_list_free().
3. The custom code for moving the packet to the packet list in
ff_interleave_add_packet() can be removed.
It will also simplify fixing further memleaks in future commits.
Suggested-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The data of an attachment file is put into an AVCodecParameter's
extradata. The corresponding size field has type int, yet there was no
check for the size to fit into an int. As a consequence, it was possible
to create extradata with negative size (by using a big enough max_alloc).
Other errors were also possible: If SIZE_MAX < INT64_MAX (e.g. on 32bit
systems) then the file size might be truncated before the allocation;
and avio_read() takes an int, too, so one would not have read as much
as one desired.
Furthermore, the extradata is now padded as is required.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Fixes: Timeout
Fixes: out of array access
Fixes: 20274/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_FLAC_fuzzer-5649631988154368
Fixes: 19275/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_FLAC_fuzzer-5757535722405888
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 2145417478 + 76702564 cannot be represented in type 'int'
Fixes: 20313/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_RA_144_fuzzer-5734487724130304
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Previously, there was no way to flush an encoder such that after
draining, the encoder could be used again. We generally suggested
that clients teardown and replace the encoder instance in these
situations. However, for at least some hardware encoders, the cost of
this tear down/replace cycle is very high, which can get in the way of
some use-cases - for example: segmented encoding with nvenc.
To help address that use case, we added support for calling
avcodec_flush_buffers() to nvenc and things worked in practice,
although it was not clearly documented as to whether this should work
or not. There was only one previous example of an encoder implementing
the flush callback (audiotoolboxenc) and it's unclear if that was
intentional or not. However, it was clear that calling
avocdec_flush_buffers() on any other encoder would leave the encoder in
an undefined state, and that's not great.
As part of cleaning this up, this change introduces a formal capability
flag for encoders that support flushing and ensures a flush call is a
no-op for any other encoder. This allows client code to check if it is
meaningful to call flush on an encoder before actually doing it.
I have not attempted to separate the steps taken inside
avcodec_flush_buffers() because it's not doing anything that's wrong
for an encoder. But I did add a sanity check to reject attempts to
flush a frame threaded encoder because I couldn't wrap my head around
whether that code path was actually safe or not. As this combination
doesn't exist today, we'll deal with it if it ever comes up.
This generates a potential memory leak, and mixes side data from the last
packet with other properties from the first.
Keep all the properties from the first packet only in the output packet
instead.
Signed-off-by: James Almer <jamrial@gmail.com>
When flushing the capture buffers, the driver may send a V4L2_EVENT_EOS
to notify that draining is completed. Currently, v4l2_m2m does not
subscribe to this event, which can cause some devices (i.e. imx8qm) to
hang at the end of encoding/decoding. Support for handling the event is
added in this commit.
Some devices may not signal V4L2_EVENT_EOS. This is logged as a warning
message during initialization and not treated as a fatal error.
Signed-off-by: Ming Qian <ming.qian@nxp.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
The only difference of the currently used write_packet()-function to
ff_raw_write_packet() is that the former also counts the number of
frames. Yet doing so in the muxer itself is unnecessary as this is
already done generically in write_packet() in libavformat/mux.c.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
These muxers don't depend on the WebM Chunk or the WebM DASH Manifest
muxers.
Furthermore, remove some #if checks in webm_chunk.c and webmdashenc.c.
They are always true now that webm_chunk.c and webmdashenc.c are only
compiled when their corresponding muxers are enabled.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
avio_internal.h has been included in this muxer since the beginning and
was never needed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
libavutil/avstring.h is unnecessary since 8a632b3e. The other
unnecessary headers were never used.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The webm_chunk muxer requires the WebM muxer, yet it does not directly
require anything from libavformat/matroska.c (it does not even include
the corresponding header). So remove the dependency from the Makefile
and add a _select to configure.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
When QSV is enabled in FFmpeg, the command "ffmpeg -hwaccels" shows a
duplicate entry in acceleration methods for QSV:
Hardware acceleration methods:
vaapi
qsv
drm
opencl
qsv
Reviewed-by: Mark Thompson <sw@jkqxz.net>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
This has happened when writing chapters: Both editions as well as
chapters are by default not hidden and given that we don't support
writing hidden chapters at all, we don't need to write said elements at
all. The same goes for ChapterFlagEnabled.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The switch cases were missing:
- Primaries: bt470m, film, smpte428, and ebu3213.
- TRCs: gamma22, gamma28, linear, log, log_sqrt, iec61966_2_4, bt1361,
iec61966_2_1, bt2020_10bit, and bt2020_12bit.
- Space: rgb, fcc, ycgco, bt2020_cl, smpte2085, chroma-derived-nc,
chroma-derived-c, and ictcp.
They also annoyingly remapped the following (which are functionally
equivalent but can be treated differently by clients):
- smpte240m primaries to smpte170m.
- smpte170m TRC to bt709.
- bt470bg color space to smpte170m.
The enum values in FFmpeg are the same values as ITU-T H.273 and
ISO/IEC 23001-8 so we can just use them directly, which is both simpler
and preserves the user intent.
Signed-off-by: Michael Bradshaw <mjbshaw@google.com>
Up until now, mkv_write_track() received the index of the stream whose
header data it is about to write as parameter; this index has until
recently been explicitly used to generate both TrackNumber and TrackUID.
But this is no longer so and as there is no reason why the function
for writing a single TrackEntry should even know the index of the
TrackEntry it is about to write, said index is replaced in the list of
function parameters by the corresponding AVStream and mkv_track.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
mkv_cuepoint (the structure used to store the index entries in the
Matroska muxer) currently contains fields for both the index of the
packet's stream in the AVFormatContext.streams array and for the
Matroska TrackNumber; correspondingly, mkv_add_cuepoint() has parameters
for both. But these two numbers can't be chosen independently, so get
rid of the TrackNumber.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Attachments are streams in FFmpeg, but they are not tracks in Matroska.
Yet they were counted when checking a limit for the number of tracks that
the Matroska muxer imposes. This is unnecessary and has been changed.
Also use unsigned variables for the variables denoting TrackNumbers as
negative TrackNumbers are impossible.
(The Matroska file format actually has practically no limit on the
number of tracks and this is purely what our muxer supports. But even if
this limit were removed/relaxed in the future, it still makes sense to
use small TrackNumbers as this patch does, because greater numbers need
more bytes to encode.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Using random values for TrackUID and FileUID (as happens when the
AVFMT_FLAG_BITEXACT flag is not set) has the obvious downside of making
the output indeterministic. This commit mitigates this by writing the
potentially random values with a fixed size of eight byte, even if their
actual values would fit into less than eight bytes. This ensures that
even in non-bitexact mode, the differences between two files generated
with the same settings are restricted to a few bytes in the header.
(Namely the SegmentUID, the TrackUIDs (in Tracks as well as when
referencing them via TagTrackUID), the FileUIDs (in Attachments as
well as in TagAttachmentUID) as well as the CRC-32 checksums of the
Info, Tracks, Attachments and Tags level-1-elements.) Without this
patch, there might be an offset/a size difference between two such
files.
The FATE-tests had to be updated because the fixed-sized UIDs are also
used in bitexact mode.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
If there are Attachments to write, the Matroska muxer currently
allocates two objects: An array that contains an entry for each
AttachedFile containing just the stream index of the corresponding
stream and the FileUID used for this AttachedFile; and a structure with
a pointer to said array and a counter for said array. These uids are
generated via code special to Attachments: It uses an AVLFG in the
normal and a sha of the attachment data in the bitexact case. (Said sha
requires an allocation, too.)
But now that an uid is generated for each stream in mkv_init(), there is
no need any more to use special code for generating the FileUIDs of
AttachedFiles: One can simply use the uid already generated for the
corresponding stream. And this makes the whole allocations of the
structures for AttachedFiles as well as the structures itself superfluous.
They have been removed.
In case AVFMT_FLAG_BITEXACT is set, the uids will be different from the
old ones which is the reason why the FATE-test lavf-mkv_attachment
needed to be updated. The old method had the drawback that two
AttachedFiles with the same data would have the same FileUID.
The new one doesn't.
Also notice that the dynamic buffer used to write the Attachments leaks
if an error happens when writing the buffer. By removing the
allocations potential sources of errors have been removed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This commit reuses the random seed generated in mkv_init() (to determine
the TrackUIDs) for the SegmentUID in order to avoid a potentially
expensive call to av_get_random_seed().
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until now, the TrackUID of a Matroska track which is supposed to be
random was not random at all: It always coincided with the TrackNumber
which is usually the 1-based index of the corresponding stream in the
array of AVStreams. This has been changed: It is now set via an AVLFG
if AVFMT_FLAG_BITEXACT is not set. Otherwise it is set like it is set
now (the only change happens if an explicit track number has been
chosen via dash_track_number, because the system used in the normal
situation is now used, too). In particular, no FATE tests need to be
updated.
This also fixes a bug in case the dash_track_number option was used:
In this case the TrackUID was set to the provided number, but the tags
were written with a TagTrackUID simply based upon the index, so that
the tags didn't apply to the track they ought to apply to.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Tags in the Matroska file format can be summarized as follows: There is
a level 1-element called Tags containing one or many Tag elements each
of which in turn contain a Targets element and one or many SimpleTags.
Each SimpleTag roughly corresponds to a single key-value pair similar to
an AVDictionaryEntry. The Targets meanwhile contains information to what
the metadata contained in the SimpleTags contained in the containing Tag
applies (i.e. to the file as a whole or to an individual track).
The Matroska muxer writes such metadata. It puts the metadata of every
stream into a Tag whose Targets makes it point to the corresponding
track. And if the output is seekable, then it also adds another Tag for
each track whose Targets corresponds to the track and where it reserves
space in a SimpleTag to write the duration at the end of the muxing
process into.
Yet there is no reason to write two Tag elements for a track and a few
bytes (typically 24 bytes per track) can be saved by adding the duration
SimpleTag to the other Tag of the same track (if it exists).
FATE has been updated because the output files changed. (Tests that
write to unseekable output (pipes) needn't be updated (no duration tag
has ever been written for them) and the same applies to tests without
further metadata.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
AVChapters have an int as id field and therefore this value can appear
<= 0. When remuxing from Matroska, this value actually contains
the lower 32 bits of the original ChapterUID (which can be 64 bits).
In order to ensure that the ChapterUID is always > 0, they were offset
as follows (since 07704c61): First max(0, 1LL - chapter[i].id) was computed
and stored in an uint32_t. And then the IDs were offset using this value.
This has two downsides:
1. It does not ensure that the UID is actually != 0: Namely if there is
a chapter with id == INT_MIN, then the offset will be 2^31 + 1 and a
chapter with id == INT_MAX will become 2^31 - 1 + 2^31 + 1 = 2^32 = 0,
because the actual calculation was performed in 32 bits.
2. As soon as a chapter id appears to be negative, a nontrivial offset
is used, so that not even a ChapterUID that only uses 32 bits is
preserved.
So change this by treating the id as an unsigned value internally and
only offset (by 1) if an id vanishes. The actual offsetting then has to
be performed in 64 bits in order to make sure that no UINT32_MAX wraps
around.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
According to C99, there has to be at least one argument for every ...
in a variadic function-like macro. In practice most (all?) compilers also
allow to leave it completely out, but it is nevertheless required: In a
variadic macro "there shall be more arguments in the invocation than there
are parameters in the macro definition (excluding the ...)." (C99,
6.10.3.4).
CBS (not the framework itself, but the macros used in the
cbs_*_syntax_template.c files) relies on the compiler allowing to leave
a variadic macro argument out. This leads to warnings when compiling in
-pedantic mode, e.g. "warning: must specify at least one argument for
'...' parameter of variadic macro [-Wgnu-zero-variadic-macro-arguments]"
from Clang.
Most of these warnings can be easily avoided: The syntax_templates
mostly contain helper macros that expand to more complex variadic macros
and these helper macros often omit an argument for the .... Modifying
them to always expand to complex macros with an empty argument for the
... at the end fixes most of these warnings: The number of warnings went
down from 400 to 0 for cbs_av1, from 1114 to 32 for cbs_h2645, from 38 to
0 for cbs_jpeg, from 166 to 0 for cbs_mpeg2 and from 110 to 8 for cbs_vp9.
These eight remaining warnings for cbs_vp9 have been fixed by switching
to another macro in cbs_vp9_syntax_template: The fixed values for the
sync bytes as well as the trailing bits for byte-alignment are now read
via the fixed() macro (this also adds a check to ensure that trailing
bits are indeed zero as they have to be).
Reviewed-by: Mark Thompson <sw@jkqxz.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Do not limit the array allocation functions and av_calloc() to allocations
of INT_MAX, instead depend on max_alloc_size like av_malloc().
Allows a workaround for ticket #7140.
On other platforms, the functions are named get_cabac_inline_xxx but not
this one. There's also a define.
Signed-off-by: Rosen Penev <rosenp@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
dts would start over at the beginning of each trun when they should be
computed contiguously for each trun in a traf
Fixes ticket 8070
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If some but not all moof's are referenced in an sidx, whole fragments
were being skipped.
Fixes tickets 7377, 7389, and 8502
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The calculation of precinct boundaries has been
fixed. The precinct boundaries were calculated
as an offset to the band boundary, but must
instead be calculated as an offset from the
reslevel. This patch fixes#4669 and #4679.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This patch fixes an error where the COC marker
overrides all data of the SPcod field of the
COD marker. It must override only one bit of
SPcod field. This now allows p0_08.j2k to be
decoded correctly (mentioned in #4679).
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This is required to use it as an AVCodec.supported_samplerates array.
Adding the sentinel has been forgotten in 4679a474.
Without it e.g. the FATE-test ffmpeg-filter_complex_audio fails with ASAN.
Reviewed-by: Carl Eugen Hoyos <ceffmpeg@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Place HAVE_MIPSFPU further up so that functions that use floating point
ASM are defined away. Otherwise compilation failures result when soft
float in enabled on the toolchain.
Signed-off-by: Rosen Penev <rosenp@gmail.com>
Regression since 8d3630c540 where keys were changed
to not be touppered but the picture block strcmp was not changed to be case-insensitive.
Fixes ticket #8608.
If the video dimensions are different than the ASS play_res then the
font sizes need to be adjusted to get the same apparent render size.
Signed-off-by: Philip Langdale <philipl@overt.org>
Initializes the mov text sample description from the ASS header and
creates an mov font table from the fonts available in the ASS Styles.
Signed-off-by: Philip Langdale <philipl@overt.org>
Font sizes are relative to the subtitle frame dimensions. If the
expected frame dimensions are not known, the font sizes will most
likely be incorrect.
Signed-off-by: Philip Langdale <philipl@overt.org>
Style flags were only being turned on. If the default was on and style
record turned off, style flag remained on.
Signed-off-by: Philip Langdale <philipl@overt.org>
It's not necessary to walk the style record list twice per subtitle
character. style records are in order and do not overlap.
Signed-off-by: Philip Langdale <philipl@overt.org>
It represents the relationship between them more naturally and will be
useful in the following commits.
Allows significantly more frames in fate-h264-attachment-631 to be
decoded.
Those SEIs refer to the currently active SPS. However, since the SEI
NALUs precede the coded picture data in the bitstream, the active SPS is
in general not known when we are decoding the SEI.
Therefore, store the content of the picture timing SEIs and actually
parse it when the active SPS is known.
Such errors are not necessarily fatal and decoding might still be
possible, e.g. it happens for MVC streams where we do not handle the
subset SPS thus failing to parse its corresponding PPS.
Currently the frame pool used by the default get_buffer2()
implementation is a single struct, allocated when opening the decoder.
A pointer to it is simply copied to each frame thread and we assume that
no thread attempts to modify it at an unexpected time. This is rather
fragile and potentially dangerous.
With this commit, the frame pool is made refcounted, with the reference
being propagated across threads along with other context variables. The
frame pool is now also immutable - when the stream parameters change we
drop the old reference and create a new one.
Specifically, between the user-facing one and the first frame thread
one.
This is fragile and dangerous, allocate separate private data for each
per-thread context.
The current design, where
- proper init is called for the first per-thread context
- first thread's private data is copied into private data for all the
other threads
- a "fixup" function is called for all the other threads to e.g.
allocate dynamically allocated data
is very fragile and hard to follow, so it is abandoned. Instead, the
same init function is used to init each per-thread context. Where
necessary, AVCodecInternal.is_copy can be used to differentiate between
the first thread and the other ones (e.g. for decoding the extradata
just once).
subtitles.h has been included in order to use ff_subtitles_next_line()
to help parsing srt subtitles which at that time had their timing as
part of the payload and not as part of the AVPacket fields. When this
changed (in 55180b32) it has been forgotten to remove this header.
libavcodec/internal.h meanwhile has been added in bb47aa5850 and has
never been used at all.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
vobsub_read_packet() didn't check whether an array of AVPackets was
valid and therefore used uninitialized values.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This patch adds functional changes to support the
PPT marker. This patch fixes bug ticket #4610.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
if the directory name of the segments contains "%v".
This memleak is caused by masking the pointer that will eventually
be freed by a variable of the same name in a smaller scope.
Therefore the pointer that gets freed is always NULL when it is
freed and the allocated data leaks.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This is mainly about improving legibility of the code and getting rid of
overlong lines by using variables for st->codecpar instead of accessing
the codecparameters via st->codecpar->.
Also, some code has been moved to better fitting places.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This commit removes data that is only used during smacker_read_header()
from the demuxer's context and replaces the data that is used by local
variables. The other data is completely dropped.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The Smacker demuxer currently parses several fields that indicate
how many audio streams a file contains. This data is parsed and stored
into arrays in the demuxer's context and although the data is used only
to initialize the audio streams, it is kept for the whole lifetime of
the demuxer.
This has been changed: The data is used directly to create
the audio streams and no longer kept at all.
This also simplifies error handling in case adding a new stream fails:
Several arrays which until now have been allocated between parsing the
data determining how many audio streams to create and actually creating
them would need to be freed in this case. Now the streams are created
first, so freeing is no longer an issue.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The Smacker demuxer reads four consecutive 32bit values from the file
header into its demux context (as four uint32_t), converting it to
native endianness in the process and then writing these four values
later (after extradata has been allocated) to extradata as four 32bit
values (converting to little endian in the process).
This commit changes this: The stream and the extradata are allocated
earlier, so that the data destined for extradata can be read directly
into extradata.
Furthermore, given that these values are not needed for demuxing itself
they are now no longer kept as part of the demuxing context.
Finally, a check regarding the number of frames has been moved up,
too, in order to exit early before unnecessarily allocating the
stream and the extradata.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Fixes: signed integer overflow: 9223372036854775775 + 128 cannot be represented in type 'long'
Fixes: 20054/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_FFWAVESYNTH_fuzzer-5686385113825280
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The WebM DASH Manifest muxer only supports VP8, VP9, Vorbis and Opus,
but there was no check for this. The codec type is used to get a pointer
to a string containing the codec name or NULL if it is not one of those
four codecs. Said pointer has then been used without further checks as
string for the %s conversion specifier in an avio_printf()) call which
is undefined behaviour.
This commit adds a check for the supported codec types.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Some broken apps generate files that have a fake box named 'hoov'
instead of a proper 'moov' one. This is speculation but it seems like
this box contains data to be modified later (eg as file grows in size,
data gets re-written) and its name is supposed to be changed to 'moov'
once it can be used as a 'moov', but for some reason this step is skipped.
Since this is not the first time this happens ('moov' boxes can be found
in 'free' ones) extend the existing hacks to search for the moov in such
boxes and skip the moov_retry since it needs to be found right away.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
The idea was to allow separate planes to be filtered independently, however,
in hindsight, literaly nothing uses separate per-plane semaphores and it
would only work when each plane is backed by separate device memory.
throughout vp3_decode_frame the error code was being captured (ret) but never returned.
Signed-off-by: Peter Ross <pross@xvid.org>
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Fixes the following warnings:
libavformat/chromaprint.c:117:42: warning: passing argument 2 of ‘chromaprint_feed’ from incompatible pointer type
libavformat/chromaprint.c:132:52: warning: passing argument 2 of ‘chromaprint_get_raw_fingerprint’ from incompatible pointer type
libavformat/chromaprint.c:143:71: warning: passing argument 4 of ‘chromaprint_encode_fingerprint’ from incompatible pointer type
The code for GAB2 subtitles predates refcounting AVPackets. So in order
to transfer the ownership of a packet's data pkt->data was simply stored
and the packet zeroed; in the end (i.e. in the read_close-function) this
data was then simply freed with av_freep(). This of course leads to a leak
of an AVBufferRef and an AVBuffer. It has been fixed by keeping and
eventually unreferencing the packet's buf instead.
Additionally, the packet is now reset via av_packet_unref().
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
An AVIStream (intended to be used as private data for an AVStream) would
leak in this scenario.
Also return a more fitting error code instead of AVERROR_INVALIDDATA.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
If one uses a build without dv demuxer, an AVIStream struct that is
destined to be used as private data for an AVStream by the avi demuxer
would leak, because it has been moved from the AVStream (that is going
to be freed) and only stored in a local variable (in order to be used
for another AVStream), but if the dv demuxer is disabled, the earlier
code returned immediately instead.
Also return a better error code in this scenario (instead of
AVERROR_INVALIDDATA).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Using ff_free_stream() makes the code more readable, more future-proof
(the old code freed AVCodecContexts and AVCodecParameters and its
substructures manually, so that there is a chance that there would be a
memleak for some time if new substructures were added) and reduces
code size.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Related to this are the following changes:
* Mention the GNUmakefile that AviSynth+ provides for installing
just the headers.
* Expand on users installing AviSynth on their system a little
more.
AviSynth+ now supports non-Windows OSes, making AvxSynth
obsolete. Since we no longer support AviSynth 2.5 (which is
essentially what AvxSynth is), remove AvxSynth support and
replace it with AviSynth+.
As a result, the USING_AVISYNTH defines can be switched back
to generic _WIN32.
This brings a performance improvement when demuxing files, most of the
improvement comes from buffer pooling unbound packets.
time ffprobe -i samples/ffmpeg-bugs/trac/ticket6132/Samsung_HDR_-_Chasing_the_Light.ts -show_packets >/dev/null 2>&1
Before:
real 0m1.967s
user 0m1.471s
sys 0m0.493s
After:
real 0m1.497s
user 0m1.364s
sys 0m0.129s
Based on a patch of James Almer.
Signed-off-by: Marton Balint <cus@passwd.hu>
Fixes: signed integer overflow: 1169365504 + 981452800 cannot be represented in type 'int'
Fixes: ticket8293
Found-by: Suhwan
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 524280 * 4432 cannot be represented in type 'int'
Fixes: ticket8322
Found-by: Suhwan
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Put a buffer with a known fixed size into the demuxer's context instead
of allocating it separately. This also allows to remove the demuxer's
read_close()-function.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
8ffcc826 added support for muxing BlockAdditions with BlockAddID equal
to one. The restriction to BlockAddID == 1 probably resulted from
a limitation to what was needed; yet over time this led to three
occurences of "(side_data_size && additional_id == 1)". This commit
changes this by setting side_data_size to 0 if additional_id != 1.
It also stops hardcoding 1 for the value of BlockAddID to write;
but it still upholds the requirement that it is 1. See below.
Despite BlockAddId actually having a default value of 1, it is still
written, because until very recently (namely dbc50f8a) our demuxer
used a wrong default value of 0.
Furthermore, use put_ebml_binary() to write the BlockAdditional element.
(The Matroska specifications have evolved and now the BlockAddID 1 is
reserved for the codec (as described in the codec's codec mapping),
BlockMore elements with BlockAddID > 1 are now of a more
codec-independent nature and require a BlockAdditionalMapping in the
track's TrackEntry. Given that this muxer does not support writing said
BlockAdditionalMapping yet (actually, none have been defined yet), we
have to uphold the requirement that BlockAddID == 1.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
When updating the Tags at the end, the Matroska muxer would twice check
for whether (!mkv->is_live) is true, despite this code being only executed
if it is. Furthermore, a loop iterates over all the streams even when
there is no Tags element to update at all, because the check for whether
there are Tags is only performed later. This commit fixes this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
avio_close_dyn_buf() has a bug: When the write pointer does not point to
the end of the written data when calling it (i.e. when one has performed
a seek back to update already written data), it would not add padding to
the end of the buffer, but to the current position, overwriting other
data; furthermore the reported size would be wrong (off by the amount of
data it has overwritten with padding).
In order not to run into this when updating already written elements or
elements for which size has only been reserved, the Matroska muxer would
first record the current position of the dynamic buffer, then seek to
the desired position, perform the update and seek back to the earlier
position.
But now that end_ebml_master_crc32() does not make use of
avio_close_dyn_buf() any more, this is no longer necessary.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The Matroska muxer uses a dynamic buffer to buffer the content of
Clusters before eventually writing them. Up until now, each time a
Cluster was written, the dynamic buffer was closed, i.e. freed; now it
is only reset, saving allocations of the AVIOContext itself, its opaque
as well as most of the reallocations of the buffer.
This is advantageous performance-wise, in particular on systems where
reallocations are slow (namely Windows). The following table shows the
decicyles for writing a frame on Linux (Ubuntu 19.10) and Windows (7)
on an x64 Haswell (to /dev/null on Linux, to stdout which is discarded
on Windows (the default values of the size and duration of clusters for
seekable output have been explicitly set in this case); in all tests,
writing CRC-32 values has been disabled in all tests; calls to the muxer's
write_packet function in write_packet() in libavformat/mux.c have been
timed; each of the following tests has been repeated 50 times):
| Windows before | Windows after | Linux before | Linux after
_________________________________________________________________
A | 979437 | 192304 | 259500 | 183320
B | 715936 | 155648 | 152786 | 130879
C | 265115 | 56034 | 78496 | 53243
D | 386224 | 80307 | 128894 | 75354
E | 21732 | 10695 | 11320 | 9801
(A is a 10.2 mb/s file with a GOP length of 2s, amounting to an average
Cluster size of about 2.5 MiB; the average Cluster size of B is 1.1 MiB;
for C it is 2.35 MiB, for D it is 0.46 MiB; for E - a file with just a
single audio track of 158kb/s resulting in a Cluster size of about 100
kB, the relative gains were the smallest, probably because of the small
Cluster size.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
vf_dnn_processing.c recently changed to use swscale to trasfer data
between AVFrame and dnn model.
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
If the buffer doesn't contain enough bytes when reading a stream,
fail rather than continuing on with initialized data. Caught by
Chromium fuzzeras (crbug.com/1065731).
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Using a compiler with a different host triplet is considered
cross-compiling, even when it is for the same architecture as the
build system. With such a cross-compiler, it is still valid to
optimize builds with --cpu=host. Make the condition that aborts in
this case into a warning instead, since a cross-compiler for an
incompatible architecture will fail with -mtune=native anyway.
Signed-off-by: David Michael <fedora.dm0@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
containing updated extradata, in this case a new FLAC streaminfo.
Furthermore, it also tests that the Matroska muxer is able to preserve
uncommon channel layouts by adding Vorbis comments to the CodecPrivate.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The hnm demuxer's context struct contained lots of fields that are
write-only variables or that are not used outside of parsing the header
and that can therefore be replaced by local variables of hnm_read_header().
This commit removes all of these from the context; the second type has
been replaced by local variables.
An AVPacket (that was initialized when reading the header and for which
dead code to unreference it existed in hnm_read_close()) is among the
removed things. Removing it allowed to remove hnm_read_close()
altogether and also removes another instance of usage of sizeof(AVPacket).
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Writing the language of WebVTT in WebM proceeded differently than the
language of all other tracks: In case no language was given, it does not
write anything instead of "und" (for undefined). Because the default
value of the Language element in WebM (that inherited it from Matroska)
is "eng" (for English), any such track will actually be flagged as
English.
Doing it this way goes back to commit 509642b4 (the commit adding
support for WebVTT) and no reason for this has been given in the commit
message or in the discussion about this patch on the mailing list; the
best I can think of is this: the WebM wiki contains "The srclang attribute
is stored as the Language sub-element." Someone unfamiliar with default
values in Matroska/WebM could interpret this as meaning that no Language
element should be written if the language is unknown. And this is wrong
and this commit changes it.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
mkv_write_track() currently has three places where it checks for whether
the current codec type is audio: One in a switch and two outside of it.
These checks can be combined by moving the code after the other two checks
inside the audio-related part of the switch.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reserving space in Matroska works by writing a Void element. And until
now this worked as follows: The current position was recorded and the
EBML ID as well as the length field written; then the new position was
recorded to know how much more to write. Afterwards the actual writing
has been performed via ffio_fill().
But it is unnecessary to explicitly use the positions (obtained via
avio_tell()) to find out how much still needs to be written, because the
length of the ID and the length field are known. So rewrite the function
to no longer use them.
Also, given that ffio_fill() uses an int parameter and given that no
current caller (and no sane future caller) will want to reserve several
GB of space, make the size parameter of put_ebml_void() itself an int.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
When the Cues are written in front of the Cluster, the muxer would seek
to the beginning (to where the Cues ought to be written) and write the
Cues; afterwards it would seek back to the end of the file only to seek
to the beginning once again to update several elements there. This
commit removes the seek to the end.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The Matroska muxer has the ability to write the Cues (the index) at the
beginning of the file (in front of the Cluster): The user inputs the
amount of space that should be reserved at the beginning of the file and
if this is sufficient, the Cues will be written there and the part of the
reserved space not used up by the Cues will be filled with a "Void"
element.
There is just one problem with this: One can not fill a single byte this
way, because said Void element is minimally two bytes long (one byte ID,
one byte length field). Up until now, if one reserved one byte more than
needed, one would run into an assert when writing the Void element.
There are two solutions for this: Error out if it happens. Or adjust the
length field of the Cues in order to ensure that the above situation
can't happen (i.e. write the length on one byte more than necessary).
The first solution is very unsatisfactory, as enough space has been
reserved. Therefore this commit implements the second solution.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
When the user opted to write the Cues at the beginning, the Cues were
simply written without checking in advance whether enough space has been
reserved for them. If it wasn't enough, the data following the space
reserved for the Cues was simply overwritten, corrupting the file.
This commit changes this by checking whether enough space has been
reserved for the Cues before outputting anything. If it isn't enough,
no Cues will be output at all and the file will be finalized normally,
yet writing the trailer will nevertheless return an error to notify
the user that his wish of having Cues at the front of the file hasn't
been fulfilled.
This change opens new usecases for this option: It is now safe to use
this option to e.g. record live streams or to use it when muxing the
output of an expensive encoding, because when the reserved space turns
out to be insufficient, one ends up with a file that just lacks Cues
but is otherwise fine.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The Matroska muxer currently assumed WavPack version 4.03 in case it was
not explicitly signalled via extradata; but following a recommendation
from David Bryant, the WavPack creator, this is changed to 4.10.
Reviewed-by: David Bryant <david@wavpack.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
It might be used by the Matroska muxer. This is also the reason why the
FATE-tests for muxing WavPack into Matroska needed to be updated: They
now write the correct version 4.07 and not 4.03 as before.
Reviewed-by: David Bryant <david@wavpack.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
mkvmerge versions 6.2 to 40.0 had a bug that made it not propagate the
WavPack extradata (containing the WavPack version) during remuxing from
a Matroska file; currently our demuxer would treat every WavPack block
encountered as invalid data (unless the WavPack stream is to be
discarded (i.e. the streams discard is >= AVDISCARD_ALL)) and try to
resync to the next level 1 element.
Luckily, the WavPack version is currently not really important; so we
fix this problem by assuming a version. David Bryant, the creator of
WavPack, recommended using version 0x410 (the most recent version) for
this. And this is what this commit does.
A FATE-test for this has been added.
Reviewed-by: David Bryant <david@wavpack.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
If the buffer doesn't contain enough bytes when reading a stream,
fail rather than continuing on with unitialized data. Caught by
Chromium fuzzers (crbug.com/1054229).
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The clli atom isn't in ISO/IEC 14496-12:2015 so the flag is marked as
experimental and the clli atom is not written by default.
The clli atom is already parsed by FFmpeg in mov.c.
Signed-off-by: Michael Bradshaw <mjbshaw@google.com>
1. When set_parameters was removed from AVOutputFormat in 2fb75019, it
was forgotten to remove the comment pertaining to it. Said comment now
appeared to apply to interleave_packet(); it is of course nonsense and
has been replaced by an accurate description.
2. The description of av_write_uncoded_frame() suggested
av_interleaved_write_frame() as a replacement if the input is not
already correctly interleaved; it also referred to said function for
details. Given that said function can't write AVFrames and that the
specifics of writing uncoded frames are explained in the description
of av_interleaved_write_uncoded_frame(), both references have been fixed.
3. Removed an outdated comment about avformat_seek_file().
Reviewed-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
struct v4l2_selection contains reserved bytes which should be set to
zero before the ioctl call.
Fixes valgrind error:
Syscall param ioctl(VKI_V4L2_S_SELECTION) points to uninitialised byte(s)
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
If ff_interleave_add_packet failed, the content of the newly created
packet new_pkt would leak.
Also, it is unnecessary to zero-initialize a packet that will be put
into av_new_packet lateron as the latter already initializes the packet.
Therefore this has been removed, too.
Reviewed-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
It is unnecessary to call prepare_input_packet if there is no packet as
it doesn't do anything, except when the currently inactive code guarded
by !FF_API_COMPUTE_PKT_FIELDS2 || !FF_API_LAVF_AVCTX becomes active:
Then attempting to access pkt->stream_index will crash.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
If one calls av_opt_set() with an incorrect string to set the value of
an option of type AV_OPT_TYPE_VIDEO_RATE, the given string is used in a
log message via %s. This also happens when the string is actually a
nullpointer in which case using it for %s is forbidden.
This commit changes this by erroring out early in case of a nullpointer.
This also fixes a warning from GCC 9.2:
"‘%s’ directive argument is null [-Wformat-overflow=]"
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Currently the driver's frame period is incorrectly set to the frame
rate. This is fixed in the commit.
Signed-off-by: Ming Qian <ming.qian@nxp.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
Don't use typedef struct MXFTrack {...} MXFTimecodeComponent, in
particular given the fact that MXFTrack is a type of its own.
Reviewed-by: Tomas Härdin <tjoppen@acc.umu.se>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
end_ebml_master_crc32_preliminary() has a MatroskaMuxContext as
parameter that isn't used at all. So remove it.
Furthermore it doesn't close its dynamic buffer; it just uses the
underlying buffer and therefore it only needs a pointer to the
dynamic buffer, not a pointer to a pointer.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until now, writing level 1 elements proceeded as follows: First, the
element id was written to the ordinary output AVIOContext and a dynamic
buffer was opened for the content of the level 1 element in
start_ebml_master_crc32(). Then this buffer was actually used and after it
was closed (in end_ebml_master_crc32()), the size field corresponding to
the buffer's size was written, after which the actual data was written.
This commit changes this: Nothing is written to the main AVIOContext any
more in start_ebml_master_crc32(). end_ebml_master_crc32() now writes
both the id, the length field as well as the data. This implies that
one can start a level 1 element in memory without outputting anything.
This is done to enable to test whether enough space has been reserved
for the Cues (if space has been reserved for them) before writing them.
A large duration between outputting the header and outputting the rest
could also break certain streaming usecases like the one from #8578
(which this commit fixes).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
When the Matroska muxer writes the Cues (the index), it groups index
entries with the same timestamp into the same CuePoint to save space.
But given Matroska's variable-length length fields, it either needs
to have an upper bound of the final size of the CuePoint before writing it
or the CuePoint has to be assembled in a different buffer, so that after
having assembled the CuePoint (when the real size is known), the CuePoint's
header can be written and its data copied after it.
The first of these approaches is the currently used one. This entails
finding out the number of entries in a CuePoint before starting the
CuePoint and therefore means that the list is read at least twice.
Furthermore, a worst-case upper-bound for the length of a single entry
was used, so that sometimes bytes are wasted on length fields.
This commit switches to the second approach. This is no longer more
expensive than the current approach if one only resets the dynamic
buffer used to write the CuePoint's content instead of opening a new
buffer for every CuePoint: Writing the trailer of a file with 540.000
CuePoints improved actually from 219054414 decicycles to 2164379394
decicycles (based upon 50 iterations).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Resetting a dynamic buffer means to keep the AVIOContext and the
internal buffer used by the dynamic buffer. This is done in order to
save (re)allocations when one has a workflow where one opens and closes
dynamic buffers in sequence.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until now, the Matroska muxer would allocate a structure containing
three members: The segment offset, a pointer to an array containing Cue
(index) entries and a counter for said array. It is unnecessary to
allocate it separately and it is unnecessary to contain the segment
offset in said structure, as it duplicates another field contained in
the MatroskaMuxContext. This commit implements the corresponding
changes.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
When writing the SeekHead (a form of index) at the end of the muxing
process, mkv_write_seekhead() would first seek to the position where the
SeekHead ought to be written, then write it there and seek back to the
original position afterwards. Which means: To the end of the file.
Afterwards, a seek to the beginning of the file is performed to update
further values. This of course means that the second seek in
mkv_write_seekhead() was unnecessary.
This has been changed: A new parameter was added to mkv_write_seekhead()
containing the destination for the second seek, effectively eliminating
the seek to the end of the file after writing the SeekHead.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
mkv_write_seekhead() would up until now try to seek to the position where
the SeekHead ought to be written, write the SeekHead and seek back. The
first of these seeks was checked as was writing, yet the seek back was
unchecked. Moreover the return value of mkv_write_seekhead() was unchecked
(the ordinary return value was the position where the SeekHead was written).
This commit changes this: Everything is checked. In the unseekable case
(where the first seek may nevertheless work when it happens in the buffer)
a failure at the first seek is not considered an error. In any case,
failure to seek back is an error.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
When the Matroska muxer writes an EBML ID, it calculates the length of
said ID before; and it does this as if this were a number that needs to
be encoded as EBML number: The formula used is (av_log2(id + 1) - 1) / 7
+ 1. But the constants used already contain the VINT_MARKER (the leading
bit indicating the length of the EBML number) and therefore the algorithm
used makes no sense. Instead the position of the most significant byte
set gives the desired length.
The algorithm used until now worked because EBML numbers are subject to
restrictions: If the EBML number takes up k bytes, then the bit 1 << (7
* k) is set and av_log2(id) is 7 * k. So the current algorithm produces
the correct result unless the EBML ID is of the form 7 * k - 1 because
of the "id + 1". But contrary to encoding lengths as EBML number (where
the + 1 exists to avoid the encodings reserved for unknown length),
such EBML numbers are simply forbidden as EBML IDs and as such none of
them were ever written.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This commit updates the documentation of av_read_frame() to match its
actual behaviour in several ways:
1. On success, av_read_frame() always returns refcounted packets.
2. It can handle uninitialized packets.
3. On error, it always returns blank packets.
This will allow callers to not initialize or unref unnecessarily.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Since commit e134c203 strdups of several elements of a manifest are kept
in the DASHContext; but said commit completely forgot to free these
strings again (with xmlFree()). Given that these strings are never used
at all, this commit closes this leak by reverting said commit.
This reverts commit e134c20374.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until now, it was completely unspecified what the content of the
destination packet dst was on error. Depending upon where the error
happened calling av_packet_unref() on dst might be dangerous.
This commit changes this by making sure that dst is blank on error, so
unreferencing it again is safe (and still pointless). This behaviour is
documented.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
av_packet_ref() mostly treated the destination packet dst as uninitialized,
i.e. the destination fields were simply overwritten. But if the source
packet was not reference-counted, dst->buf was treated as if it pointed
to an already allocated buffer (if != NULL) to be reallocated to the
desired size.
The documentation did not explicitly state whether the dst will be treated
as uninitialized, but it stated that if the source packet is not refcounted,
a new buffer in dst will be allocated. This and the fact that the side-data
as well as the codepath taken in case src is refcounted always treated the
packet as uninitialized means that dst should always be treated as
uninitialized for the sake of consistency. And this behaviour has been
explicitly documented.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Validate the value of ST field in the TLM marker of JPEG2000.
Throw an error when ST takes value of 0b11.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The specifications are de-facto frozen now as they've already been used in
production for years, the author has indicated reluctance on IRC to change
it further, and the only potential changes would, from what I understand,
be forward-compatible.
Add overflow test for hevc_add_res when int16_t coeff = -32768.
The result of C is good, while ASM is not.
To verify:
make fate-checkasm-hevc_add_res
ffmpeg/tests/checkasm/checkasm --test=hevc_add_res
./checkasm --test=hevc_add_res
checkasm: using random seed 679391863
MMXEXT:
hevc_add_res_4x4_8_mmxext (hevc_add_res.c:69)
- hevc_add_res.add_residual [FAILED]
SSE2:
hevc_add_res_8x8_8_sse2 (hevc_add_res.c:69)
hevc_add_res_16x16_8_sse2 (hevc_add_res.c:69)
hevc_add_res_32x32_8_sse2 (hevc_add_res.c:69)
- hevc_add_res.add_residual [FAILED]
AVX:
hevc_add_res_8x8_8_avx (hevc_add_res.c:69)
hevc_add_res_16x16_8_avx (hevc_add_res.c:69)
hevc_add_res_32x32_8_avx (hevc_add_res.c:69)
- hevc_add_res.add_residual [FAILED]
AVX2:
hevc_add_res_32x32_8_avx2 (hevc_add_res.c:69)
- hevc_add_res.add_residual [FAILED]
checkasm: 8 of 14 tests have failed
Signed-off-by: Xu Guangxin <guangxin.xu@intel.com>
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
check_func will return NULL for functions that have already been tested. If
the func is tested and skipped (which happens several times), there is no
need to prepare data(randomize_buffers and memcpy).
Move relative code in compare_add_res(), prepare data and do check only if
the function is not tested.
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Resolution/format changes lead to re-initialization of hardware
accelerations(vaapi/dxva2/..) with new hwaccel_priv_data in
the worker-thread. But hwaccel_priv_data in user context won't
be updated until the resolution changing frame is output.
A termination with "-vframes" just after the reinit will lead to:
1. memory leak in worker-thread.
2. double free in user-thread.
Update user context in ff_frame_thread_free with the last thread
submit_packet() was called on.
To reproduce:
ffmpeg -hwaccel vaapi(dxva2) -v verbose -i
fate-suite/h264/reinit-large_420_8-to-small_420_8.h264 -pix_fmt nv12
-f rawvideo -vsync passthrough -vframes 47 -y out.yuv
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The comments for some of the markers were incorrect.
This patch fixes the comments associated with the markers.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Increasing it by 2048 entries per realloc is exessive.
Reduces memory usage, especially on long, non fragmented output.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: James Almer <jamrial@gmail.com>
Up until e7ddafd5, the Matroska muxer wrote two SeekHeads: One at the
beginning referencing the main level 1 elements (i.e. not the Clusters)
and one at the end, referencing the Clusters. This second SeekHead was
useless and has therefore been removed. Yet the SeekHead-related
functions and structures are still geared towards this usecase: They
are built around an allocated array of variable size that gets
reallocated every time an element is added to it although the maximum
number of Seek entries is a small compile-time constant, so that one should
rather include the array in the SeekHead structure itself; and said
structure should be contained in the MatroskaMuxContext instead of being
allocated separately.
The earlier code reserved space for a SeekHead with 10 entries, although
we currently write at most 6. Reducing said number implied that every
Matroska/Webm file will be 84 bytes smaller and required to adapt
several FATE tests; furthermore, the reserved amount overestimated the
amount needed for for the SeekHead's length field and how many bytes
need to be reserved to write a EBML Void element, bringing the total
reduction to 89 bytes.
This also fixes a potential segfault: If !mkv->is_live and if the
AVIOContext is initially unseekable when writing the header, the
SeekHead is already written when writing the header and this used to
free the SeekHead-related structures that have been allocated. But if
the AVIOContext happens to be seekable when writing the trailer, it will
be attempted to write the SeekHead again which will lead to segfaults
because the corresponding structures have already been freed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This fixes memleaks if an error happens after one of the allocations
in init; or if the trailer isn't written (e.g. because there was an
error when writing a packet).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
chunk_start_index (which was set via an option) was only used to
initialize chunk_index and otherwise unused. So initialize chunk_index
directly via the option and remove chunk_start_index.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Instead just reuse the filename string that is given via an option
for the child muxer's url field.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
It is no longer needed given that the function pointers of the child
muxer's AVOutputFormat are no longer called directly.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The webm_chunk muxer caches its output to a dynamic buffer and when it
outputs anything, it explicitly flushes it. So set the flags indicating
that flushing after each packet should not be done automatically
(basically avoiding avio_write_marker() to be called by flush_if_needed()
in libavformat/mux.c).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
instead of calling the write_header/packet/trailer functions directly
via the function pointers. Also, use distinct AVStreams for the child
AVFormatContext (up until now the two AVFormatContexts shared their
AVStreams because allocating their own was deemed too onerous).
Using the function pointers directly meant that the Matroska muxer's
init-function was never called, because init-functions were only
introduced a few months after webm_chunk has been added and no one
thought of/bothered to adapt webm_chunk for this (when the init-function
was added in b287d7ea, the code setting the timebase was moved to it,
so that the timebases were no longer set to ms-precision when using
the webm_chunk muxer; this has been fixed after some time in 42a635dd
by setting the timebases direcly (instead of calling the init-function)).
And when 982a98a0 added a deinit-function for the Matroska muxer, it
introduced memleaks in webm_chunk, because the child muxer's internal
structures were no longer freed when calling write_trailer directly.
(Given that the init function has never ever been called, the child
muxer has never ever been properly initialized, so that the
deinit-function was not called when freeing the child context.)
This commit stops calling the function pointers directly and instead
uses the standard API functions for muxers. This fixes the above
mentioned memleaks. (Memleaks are still possible on error. This will be
fixed in a future commit that adds a deinit-function to webm_chunk
itself.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
In particular the flags are important so that AVFMT_FLAG_BITEXACT can be
honoured by the child muxer.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The WebM DASH Manifest muxer can write manifests for live streams and
these contain an entry that depends on the time the manifest is written;
an AVOption to make the output reproducible has been added for tests.
But this is unnecessary, as there already is a method for reproducible
output: The AVFMT_FLAG_BITEXACT-flag of the AVFormatContext. Therefore
this commit removes the custom option.
Given that the description of said option contained "private option -
users should never set this" and that it was not documented in
muxers.texi, no deprecation period for this option seemed necessary.
The commands of the FATE-tests for this muxer have been changed to no
longer use this option.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This commit fixes the names and constants of the reserved NAL units
with nal_unit_type 22 resp. 23. They were "IRAP_IRAP_VLC2x", but are
actually "RSV_IRAP_VLC2x".
This also required a change to cbs_h265_syntax_template.c.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Surprisingly neither GCC nor Clang did this transformation on their own.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
If an URI indicated that the data protocol was in use, it would be
copied into a temporary buffer via strncpy(dst, src, strlen(src)),
thereby ensuring that the trailing \0 would not be copied, despite dst
being uninitialized. dst would then be av_strdup'ed, leading to
potential segfaults.
The solution to this is simple: Don't copy the URI in the temporary
buffer at all, instead av_strdup it directly.
This fixes a -Wstringop-truncation warning emitted by GCC 9.2.
Reviewed-by: Steven Liu <lq@chinaffmpeg.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
By itself, this allows 6-point, 10-point and 30-point transforms.
When the 9-point transform is added it allows for 18-point FFT,
and also for a 36-point MDCT (used by MP3).
Do it only if the packet has been successfully allocated in
av_new_packet() -- otherwise on error a completely uninitialized packet
would be unreferenced later.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
by putting the palette in the demuxer's context. This also allows to
remove this demuxer's read_close-function.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Up until now, the name of every AVBSFContext for logging purposes was
"AVBSFContext", so that the default logging callback produced output
like "[AVBSFContext @ 0x55813bae92c0] Extradata". This has been changed
to "[trace_headers @ 0x60a000000700] Extradata" by adding an item_name-
function to the AVClass for bitstream filters.
Furthermore, the correct category has been set so that the introductory
part before the actual message (everything before "Extradata" in the
above examples) are displayed in a different colour than the rest.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: James Almer <jamrial@gmail.com>
This fixes mpeg2video stream copies to mpeg muxer like this:
ffmpeg -i xdcamhd.mxf -c:v copy output.mpg
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This matches the inclusion of the other hwcontext_<hwaccel>.h headers.
The skipping of the header depending on build flags is already present.
Signed-off-by: Daniel Playfair Cal: <daniel.playfair.cal@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
PRId64 and PRIu64 already expand to the complete specifier; adding
another 'd' at the end is wrong and just leads to warnings that say
that only an option like '-frames:v 2d' will be used, although said
option won't be accepted at all ('Expected int64 for frames:v but found
2d').
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
Fixes: signed integer overflow: 1302123111085380114 - -8319005078741256972 cannot be represented in type 'long'
Fixes: 20991/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_FFWAVESYNTH_fuzzer-5148554161291264
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>
Utilizes a subpicture sample with one decodable subpicture for the
test.
Based on a failing test case in reported by Michael in
https://ffmpeg.org/pipermail/ffmpeg-devel/2019-February/240398.html
which at the time had no test case for it.
Additionally, this is the first test case for the presentation
graphics format.
Each time the sub2video structure is initialized, the sub2video
subpicture is initialized together with the first received heartbeat.
The heartbeat's PTS is utilized as the subpicture start time.
Additionally, add some documentation on the stages.
They use non-public functions, which is unacceptable for a public API
example. Rename the example back to avio_list_dir.
This effectively reverts c84d208c27 and
767d780ec0.
There was no consensus wheter or not to allow unofficial frame rates due to
possible interoperability issues, a compromise is to only allow it if -strict
mode is set to unofficial.
Signed-off-by: Marton Balint <cus@passwd.hu>
Only MXF used an actual sample array, and that is unneeded there because simple
rounding rules can be used instead.
Signed-off-by: Marton Balint <cus@passwd.hu>
If an AVFormatContext could be allocated, but white-/blacklists couldn't
be copied, the AVFormatContext would leak as it was only accessible
through a local variable that goes out of scope when one goes to fail.
Furthermore, in case writing a header of a submuxer failed, the options
used for said call could leak.
Both of these memleaks have been fixed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Don't read a 64bit number before having checked that the data is at
least 8 bytes long.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Should writing the header fail, the allocations already performed will
be freed during deinit so remove the frees in nut_write_header().
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If writing a packet didn't directly return an error, the AVIOContext's
error flag is checked for errors (if existing) by write_packet(). And if
write_packet() didn't indicate an error, its callers checked the error
flag of the AVIOContext (if existing). The latter check is redundant.
The reason for checking twice lies in the FFmpeg-Libav split: The check
in write_packet() has been added in 9ad1e0c1 in Libav. FFmpeg already
had the other checks (since aec9390a), but when 9ad1e0c1 was merged
(in 1f1c1008), no one noticed the redundant checks.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This version is able to output multiple coefficients at a time and
is able to altogether remove actual golomb code parsing.
Its also able to partially recover the last coefficient in case
the packet is incomplete.
Total decoder performance gain for 8bit 420 1080p lossless: 40%.
Total decoder performance gain for 10bit 420 1080p lossless: 40%.
clang was able to vectorize the loop much better than
my handwritten assembly, but gcc was very naive and didn't.
Lookup table is a rewritten version of vc2hqdecode.
The specifications are very vague about who has ownership, and in this case,
Vulkan takes ownership of all DMABUF FDs passed to it, causing errors
to occur if someone gave us images for mapping which were meant to be kept.
The old behavior worked with one-way VAAPI and DMABUF imports, but was broken
with clients like wlroots' dmabuf-capture.
There was a recent change in Intel's driver that triggered a driver-internal
error if the semaphore given to the command buffer wasn't initialized.
Given that the specifications require the semaphore to be initialized,
this is within spec. Unlike what's causing it in the first place, which is
that there are no ways to extract/import dma sync objects from DMABUFs,
so we must leave our semaphores bare.
The Y channel is handled by dnn, and also resized by dnn. The UV channels
are resized with swscale.
The command to use espcn.pb (see vf_sr) looks like:
./ffmpeg -i 480p.jpg -vf format=yuv420p,dnn_processing=dnn_backend=tensorflow:model=espcn.pb:input=x:output=y -y tmp.espcn.jpg
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Reviewed-by: Pedro Arthur <bygrandao@gmail.com>
Only the Y channel is handled by dnn, the UV channels are copied
without changes.
The command to use srcnn.pb (see vf_sr) looks like:
./ffmpeg -i 480p.jpg -vf format=yuv420p,scale=w=iw*2:h=ih*2,dnn_processing=dnn_backend=tensorflow:model=srcnn.pb:input=x:output=y -y srcnn.jpg
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Reviewed-by: Pedro Arthur <bygrandao@gmail.com>
This patch fixes Bug #8469
If x264 baseline profile is used with other profiles,
start_pts will be initialized to audio stream's first pts,
while the duration is calculated based on video stream's pts.
In this patch the start_pts is initialized with the correct stream's first pts.
Signed-off-by: Hongcheng Zhong <sj.hc_Zhong@sjtu.edu.cn>
Reviewed-by: Steven Liu <liuqi05@kuaishou.com>
Fixes: out of array access
Fixes: 19734/clusterfuzz-testcase-minimized-ffmpeg_BSF_TRACE_HEADERS_fuzzer-5673507031875584
Fixes: 19353/clusterfuzz-testcase-minimized-ffmpeg_BSF_TRACE_HEADERS_fuzzer-5703944462663680
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: left shift of negative value -695
Fixes: 19232/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MPEG1VIDEO_fuzzer-5702856963522560
Fixes: 19555/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MPEG1VIDEO_fuzzer-5741218147598336
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Add support for WavPack DSD files to the existing WavPack decoder using
avcodec/dsd to perform the 8:1 decimation to 32-bit float samples. We must
serialize the dsd2pcm operation (cross-boundary filtering) but would like
to use frame-level multithreading for the CPU-intensive DSD decompression,
and this is accomplished with ff_thread_report/await_progress(). Because
the dsd2pcm operation is independent across channels we use slice-based
multithreading for that part.
Also a few things were removed from the existing WavPack decoder that
weren't being used (primarily the SavedContext stuff) and the WavPack
demuxer was enhanced to correctly determine the sampling rate of DSD
files (and of course to no longer reject them).
Signed-off-by: David Bryant <david@wavpack.com>
Also remove the ancient reference to libmpcodecs while at it.
Signed-off-by: James Almer <jamrial@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Slices that end after their header (meaning slices after the header
without any data before the rbsp_stop_one_bit or possibly without any
rbsp_stop_one_bit at all) are invalid and are now dropped. This ensures
that one doesn't run into two asserts in cbs_h2645_write_slice_data().
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Fixes: 19629/clusterfuzz-testcase-minimized-ffmpeg_BSF_H264_METADATA_fuzzer-5676822528524288
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Trailing zeroes are already discarded when splitting a fragment, which
makes the code to remove them when decomposing slices dead code.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Up until now, v4l2_m2m would write via snprintf() into an intermediate
buffer and then copy from there (via strncpy()) to the end buffer. This
commit changes this by removing the intermediate buffer.
The call to strncpy() was actually of the form strncpy(dst, src,
strlen(src) + 1) which is unsafe in general, but safe in this instance
because dst and src were both of the same size and src was a proper
zero-terminated string. But this nevertheless led to a compiler warning
"‘strncpy’ specified bound depends on the length of the source argument
[-Wstringop-overflow=]" in GCC 9.2. strlen() was unnecessary anyway.
Reviewed-by: Andriy Gelman <andriy.gelman@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
If 'write_colr' movflag is set, then movflag 'prefer_icc' can
be used to first look for an AV_PKT_DATA_ICC_PROFILE entry to
encode.
If ICC profile doesn't exist, default behaviour enabled by
'write_colr' occurs.
Signed-off-by: vectronic <hello.vectronic@gmail.com>
Use desc->log2_chroma_w/h to calculate the sps->conf_win_right/bottom_offset.
Based on Table 6-1, SubWidthC and SubHeightC depend on chroma format(log2_chroma_w/h).
Based on D-28 and D-29, set the correct cropped width/height.
croppedWidth = pic_width_in_luma_samples −
SubWidthC * ( conf_win_right_offset + conf_win_left_offset );
croppedHeight = pic_height_in_luma_samples −
SubHeightC * ( conf_win_bottom_offset + conf_win_top_offset );
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Supports connecting to a RabbitMQ broker via AMQP version 0-9-1.
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
3469cfab added a check for whether the extradata coincided with the
beginning of the packet's data in order not to add extradata to packets
that already have it. But the check used was buggy for packets whose
size is smaller than the extradata's size. This commit fixes this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Mainly reindentation, but some variables were also put into a smaller
scope.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
1. Left shifts of signed values are undefined as soon as the result is
no longer representable in the target type. Therefore make nal_size
an uint32_t and drop the check for whether it is < 0.
2. The two checks for overreads (whether the length field is contained
in the packet and whether the actual unit is contained in the packet)
can be combined into one because the packet is padded, i.e. a potential
overread caused by reading the length field without checking whether
said length field is actually part of the packet's buffer is allowed
as one always stays within the padding. But one has to be aware of
a pitfall: The comparison must be performed in (at least) int64_t as
otherwise buf_end - buf might be promoted to uint32_t in which case
an already occured overread would appear as a very large number.
A comment explaining this has been added, too.
3. Units of size zero are now silently dropped; the earlier code would
instead read the first byte of the next length field (or the first byte
of padding) to infer the type of the current unit.
4. Futhermore, the earlier code returned the wrong error code. This has
been fixed, too.
Fixes#8290.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Up until now, h264_mp4toannexb would grow the output packet's buffer by
the desired amount every time another NAL unit of the input packet has
been read; this commit changes this: The input buffer is now essentially
parsed twice, once to determine the final size of the output packet and
once to write the output packet's data.
Fixes: Timeout
Fixes: 19322/clusterfuzz-testcase-minimized-ffmpeg_BSF_H264_MP4TOANNEXB_fuzzer-5688407821123584
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
h264_mp4toannexb_filter currently uses both indices/offsets as well as
direct pointers comparisons for the checks whether one has reached or
even surpassed the end. This commit removes the offsets.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If processing an input NAL unit triggers the insertion of data from
extradata in front of said NAL unit, the output packet is grown (i.e.
reallocated) once to accomodate both the new extradata as well as the
input NAL unit itself; this has been changed: In such a situation, the
packet is now grown twice. While this is bad for performance, it allows
to simplify the code and ultimately to stop reallocating the packet
altogether.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
According to the H.264 specifications, the only NAL units that need to
have four byte startcodes in H.264 Annex B format are SPS/PPS units and
units that start a new access unit. Before af7e953a, the first of these
conditions wasn't upheld as already existing in-band parameter sets
would not automatically be written with a four byte startcode, but only
when they already were at the beginning of their input packets. But it
made four byte startcodes be used too often as every unit that is written
together with a parameter set that is inserted from extradata received a
four byte startcode although a three byte start code would suffice
unless the unit itself were a parameter set.
FATE has been updated to reflect the changes. Although the patch leaves
the extradata unchanged, the size of the extradata according to the FATE
reports changes. This is due to a quirk in ff_h2645_packet_split which
is used by extract_extradata: If the input is Annex B, the first zero of
a four byte startcode is considered a part of the last unit (if any).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Up until now, h264_mp4toannexb stored the offset of the first SPS and
the first PPS in the (output) extradata in its context and used these
two numbers together with the size of the extradata and the pointer to
the extradata to determine what to insert when inserting extradata. This
led to some very long lines like "s->pps_offset != -1 ? s->pps_offset :
ctx->par_out->extradata_size - s->sps_offset". Therefore now pointers to
SPS and PPS are stored along with their respective sizes, so that e.g.
the above line can be changed to "s->sps_size".
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The format of an AVCDecoderConfigurationRecord, the out-of-band
extradata of H.264 in mp4, is as follows: First four bytes containing
version, profile and level, one byte for the length size and one byte
each for the number of SPS, followed by the SPS (each with its own size
field), followed by a byte containing the number of PPS followed by the
PPS with their size fields. While the number of SPS/PPS may be zero, the
bytes containing these numbers are mandatory. Yet the byte containing
the number of PPS has been ignored in two places:
1. In the initial check for whether the extradata can contain an
AVCDecoderConfigurationRecord. The minimum size is 7, not 6.
2. No check is made for whether the extradata ended right after the last
byte of the last SPS of the SPS array. Instead the first byte of the
padding is read as if it were part of the extradata and contained the
number of PPS (namely zero, given that the padding is zeroed). No error
or warning was ever raised.
This has been changed. Such truncated extradata is now considered
invalid; the check for 2. has been incorporated into the general size
check.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Before reading a 16bit size field during parsing of extradata, no check
is performed to make sure that said length field is actually contained
in the extradata. Given that this overread is not dangerous (the extradata
is supposed to be padded), only a comment for it has been added; the error
itself will be detected as part of the normal check for overreads.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Currently during parsing the extradata, h264_mp4toannexb checks for
overreads by adding the size of the current unit to the current position
pointer and comparing this to the end position of the extradata. But
pointer comparisons and pointer arithmetic are only defined if it does not
exceed the object it is used on (one past the last element of an array
is allowed, too). In practice, this might lead to overflows. Therefore
the check has been changed to use bytestream2_get_bytes_left() which
means that the pointers get subtracted and the result gets compared to
the available size.
Furthermore, the error code has been fixed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This is done in order to improve readability. No functional change is
intended with this commit at all; in particular, the unsafe read
functions are used throughout as h264_extradata_to_annexb already
performs its own checks. (These checks will nevertheless be improved
in further commits.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
bytestream2_get_bytes_left returns an unsigned int; as a result,
it returns big positive numbers if an overread already happened,
making it unsuitable for scenarios where one wants to allow this
in a controlled way (because the buffer is actually padded so that
no segfaults can happen). So change it to return an ordinary int.
Also, bytestream2_get_bytes_left_p has been modified in the same way.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Dashdec can able to handle MPEG-2 TS streams by default as well,
used MP4Box to create the segmented MPEG-2 TS files for
verification.
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
It's a duplicate of the properly implemented nvdec libavcodec hwaccel
Reviewed-by: Timo Rothenpieler <timo@rothenpieler.org>
Signed-off-by: James Almer <jamrial@gmail.com>
The standard does not seem to require the counter to be zero based, but some
checker tools (MyriadBits MXFInspect, Interra Baton) have validations against 0
start...
Fixes ticket #6781.
Signed-off-by: Marton Balint <cus@passwd.hu>
The second ; in a double ;; is actually a null statement. It triggers
the typical declaration-after-statement compiler-warnings if it occurs
in the middle of several declarations (like here).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The second ; in a double ;; is actually a null statement. It triggers
the typical declaration-after-statement compiler-warnings if it occurs
in the middle of several declarations (like here).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This allows for easy shortlog/log parsing, useful in determining
eligible members of the general assembly for the new FFmpeg voting
system.
Signed-off-by: Josh de Kock <josh@itanimul.li>
The capture_started variable was never set, it is simpler to call the stop
functions unconditionally if the interface is available.
Signed-off-by: Marton Balint <cus@passwd.hu>
Prevents memleaks when the trailer is never written or when shifting the
data fails when writing the trailer.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If we are given a non-render node, try to find the matching render node and
fail if that isn't possible.
libva will not accept a non-render device which is not DRM master, because
it requires legacy DRM authentication to succeed in that case:
<https://github.com/intel/libva/blob/master/va/drm/va_drm.c#L68-L75>. This
is annoying for kmsgrab because in most recording situations DRM master is
already held by something else (such as a windowing system), leading to
device derivation not working and forcing the user to create the target
VAAPI device separately.
Add function pointer field in vaapi_profile_map[], set profile_parser
for HEVC_REXT to find the exact va_profile.
Also add format map support.
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Add vaapi_parse_rext_profile and use profile constraint flags to
determine the exact va_profile for HEVC_REXT.
If profile mismatch is allowed, select Main profile by default.
Add build object in Makefile for h265_profile_level dependency.
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Parse all the constraint flags according to ITU-T Rec. H.265 (02/2018).
They have to be passed to hw decoders to determine the exact profile for Range
Extension HEVC.
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Extend ParameterBufferHEVC to ParameterBufferHEVCExtension for both
VAPicture and VASlice.
Pass Range Extension flags to support the decode for HEVC REXT.
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
VA_RT_FORMAT describes the desired sampling format for surface.
When creating surface, VA_RT_FORMAT will be used firstly to choose
the expected fourcc/media_format for the surface. And the fourcc
will be revised by the value of VASurfaceAttribPixelFormat.
Add vaapi_format_map support for new pixel_format Y210.
This is fundamental for both VA-API and QSV.
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Add swscale input support for Y210LE, output support and fate
test could be added later if there is requirement for software
CSC to this packed format.
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Otherwise the user is not able to override the SRT API connect timeout above 5
sec without also setting the timeout option.
Signed-off-by: Marton Balint <cus@passwd.hu>
Return os error code if available, check for both SRT_EASYNCRCV and
SRT_EASYNCSND when transforming them to EAGAIN and only display error if it is
not EAGAIN.
Signed-off-by: Marton Balint <cus@passwd.hu>
After f8990c5f41 we properly set non-blocking
mode which makes the connect() call return always 0 even if no connection can
be established.
Fix this by always doing a poll after calling connect(). Also there was some
leftover copy paste code which checks for various errors which are simply not
possible with SRT.
Signed-off-by: Marton Balint <cus@passwd.hu>
rw_timeout is the generic URLcontext option, not the protocol specific timeout
option, also ?rw_timeout never worked because ?timeout was parsed instead.
Signed-off-by: Marton Balint <cus@passwd.hu>
This ensures it's written at the beginning of a segment in non streaming mode
when segment duration differs from fragment duration.
Signed-off-by: James Almer <jamrial@gmail.com>
If an audio stream is present, the Sega FILM muxer checks for its
compability with the container during init, so that the very same check
needn't be repeated during writing the trailer.
Essentially the same is true for the presence of a video stream: It has
already been checked during init. Furthermore, after the check for the
presence of a video stream succeeded, a pointer is set to point to the
video stream. Yet said pointer (which was NULL before) will be
derefenced anyway regardless of the result of the check. Coverity thus
complained about this in CID 1434155 and removing this pointless check
will also fix this issue.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
No testcases, found by code review when debuging issue found by oss-fuzz
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Commit 36e156bef0 ("avformat/spdifenc: fix handling of large TrueHD
frame") added an obviously incorrect bitshift that caused incorrect
samples-per-frame calculation for TrueHD streams over 48kHz.
Fix that.
The TrueHD IEC 61937 encapsulation code uses a very naive method of
always inserting 24 TrueHD frames evenly in a MAT frame. This does not
work for larger frames as they may exceed the size of 1/24th of a MAT
frame.
To fix that, use the input_timing field in the TrueHD frame to determine
the proper position of the TrueHD frame in the MAT frame. That field is
basically a dts field, telling the time to feed this frame to the
decoder in sample count units.
This can cause a TrueHD frame to be split between two MAT frames, so a
second concatenation hd_buf is added, alternating with the first buffer.
Large frames are preceded by smaller frames that have input_timing
values that cause the frames to be sent out faster than the nominal rate
(i.e. increasing decoder latency, long decoder buffer), allowing the
larger frames to then be sent out slower than the nominal rate as the
decoder has enough data buffered to keep it busy.
This is preparation for adding a second hd_buf in a followup commit.
Also, slightly improve the comments for hd_buf_x members to clarify
which ones are actually used and kept up-to-date depending on which
codec is being muxed.
wtvfile_read_packet did not abide by the requirements of an
AVIOContext's read_packet-function: If it did not read anything,
it returned zero, which currently leads to a warning in read_packet_wrapper
in aviobuf.c. Said warning will be an av_assert2 as soon as
FF_API_OLD_AVIO_EOF_0 is zero (probably the next major version bump).
So instead forward the error code from the underlying protocol.
This error/assert is triggered in the wtv-demux FATE test.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Enables option for hevc_qsv encoder:
-extbrc
Improvements in BD-Rate could be observed with extbrc on.
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Signed-off-by: Zhong Li <zhongli_dev@126.com>
Class name is used in show_help_children(avfilter_get_class(),...)
to prompt the available filters.
$ ffmpeg -h full
Before:
qsvscale AVOptions:
After:
scale_qsv AVOptions:
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Signed-off-by: Zhong Li <zhongli_dev@126.com>
It's not exclusive for Low Latency streaming. The muxer will serve partial
segments regardless of streaming mode.
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes: signed integer overflow: 2045163756 * 2 cannot be represented in type 'int'
Fixes: Ticket5132
Found-by: tsmith
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 0 - -9223372036854775808 cannot be represented in type 'long'
Fixes: Ticket8149
Found-by: Suhwan
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -9223372036854775808 - 9223372036854775807 cannot be represented in type 'long'
Fixes: Ticket8142
Found-by: Suhwan
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 1562273630 * 17 cannot be represented in type 'int'
Fixes: Ticket8323
Found-by: Suhwan
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
- do not require a known file size for seek to work
- read the files till the actual end, do not limit data at the queried file
size
- fix a bug which causes reading 0 byte files for non-existing files
- properly check the return status of the FTP server at the end of the file
retrieval
Signed-off-by: Marton Balint <cus@passwd.hu>
v2: Use s->buffer for creating request (as the old code did) instead of
the AVBPrint internal buffer. Some minor cosmetics.
Signed-off-by: Marton Balint <cus@passwd.hu>
It is explicitly required by the HTTP RFC. Without this patch URLs like
http://example.com?query will not work.
Fixes ticket #8466.
Signed-off-by: Marton Balint <cus@passwd.hu>
RFC 3986 states that the fragment identifier is separated from the rest of the
URI prior to a dereference, and thus the identifying information within the
fragment itself is dereferenced solely by the user agent.
Signed-off-by: Marton Balint <cus@passwd.hu>
RFC 3986 states that the generic syntax uses the slash ("/"), question mark
("?"), and number sign ("#") characters to delimit components that are
significant to the generic parser's hierarchical interpretation of an
identifier.
Signed-off-by: Marton Balint <cus@passwd.hu>
A demuxer might have allocated memory while reading the header. If
reading the header was successfull and an error happens before returning
(e.g. when queueing the attached pictures), the read_close function
would have never been called, so that all those allocations would leak.
This commit changes this.
Furthermore, there would be even more memleaks if the error level was
set to AV_EF_EXPLODE in case there is both metadata and id3v2 metadata.
This has been fixed, too.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
This adds a decoder for Broderbund's sprite-based QuickTime CDToons
codec, based on the decoder I wrote for ScummVM.
Signed-off-by: Alyssa Milburn <amilburn@zall.org>
Fixes "passing argument 2 of ‘strtoul’ from incompatible pointer
type [-Wincompatible-pointer-types]" ("expected ‘char ** restrict’ but
argument is of type ‘const char **’") for GCC and "passing 'const char
**' to parameter of type 'char **' discards qualifiers in nested pointer
types [-Wincompatible-pointer-types-discards-qualifiers]" for Clang.
The cast itself is safe; it is only needed because strtoul itself is not
const-correct.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
If i is greater than 0, it is a requirement of bitstream conformance that point_y_value[ i ] is greater than point_y_value[ i - 1 ].
If i is greater than 0, it is a requirement of bitstream conformance that point_cb_value[ i ] is greater than point_cb_value[ i - 1 ].
If i is greater than 0, it is a requirement of bitstream conformance that point_cr_value[ i ] is greater than point_cr_value[ i - 1 ].
Reviewed-by: Mark Thompson <sw@jkqxz.net>
Signed-off-by: James Almer <jamrial@gmail.com>
Required minimal changes to the code so made sense to implement.
FFT and MDCT tested, the output of both was properly rounded.
Fun fact: the non-power-of-two fixed-point FFT and MDCT are the fastest ever
non-power-of-two fixed-point FFT and MDCT written.
This can replace the power of two integer MDCTs in aac and ac3 if the
MIPS optimizations are ported across.
Unfortunately the ac3 encoder uses a 16-bit fixed point forward transform,
unlike the encoder which uses a 32bit inverse transform, so some modifications
might be required there.
The 3-point FFT is somewhat less accurate than it otherwise could be,
having minor rounding errors with bigger transforms. However, this
could be improved later, and the way its currently written is the way one
would write assembly for it.
Similar rounding errors can also be found throughout the power of two FFTs
as well, though those are more difficult to correct.
Despite this, the integer transforms are more than accurate enough.
FMO is not supported and fields related to FMO are not parsed, meaning
that any fields which follow will be corrupt.
Signed-off-by: James Almer <jamrial@gmail.com>
For audio packets with dts != AV_NOPTS_VALUEs the dts was converted
twice to the muxer's timebase during streamcopy, once as a normal
packet and once specifically as an audio packet. This has been changed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Compared to ad-hoc if(printed) ... code this allows the user to disable
it by adjusting the log level
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When e2_pts == INT64_MIN and e1_pts >= 0 the calculation of
e2_pts - e1_pts will overflow an int64_t.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
In order for rate control to correctly allocate bitrate to each temporal
layer, correct temporal layer id has to be set to each frame. This
commit provides the ability to set correct temporal layer id for each
frame.
Signed-off-by: James Zern <jzern@google.com>
Previously, it was always left in the automatic mode, if the option
was set to the only special (negative) value. Now there's two separate
special values for this option, -1 for automatic (metadata based)
and -2 for explicitly disabled.
Signed-off-by: Martin Storsjö <martin@martin.st>
It was disabled by default in 2dbd35b00c
as it added delay, but now we compensate for the delay properly
by offsetting timestamps.
Signed-off-by: Martin Storsjö <martin@martin.st>
The AVPacket destined for a demuxer's output has already been
initialized before it reaches the demuxer.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
Even in this scenario, the frame still contains references to data that
won't be freed if the frame isn't unreferenced. And the AVFrame itself
will leak, too.
Fixes Coverity issue #1441422.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Since bae8844e the packet will always be unreferenced when a demuxer
returns an error, so that a lot of calls to av_packet_unref() in lots of
demuxers are now redundant and can be removed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
Tested using this command:
/ffmpeg -pix_fmt yuv420p -s 1920*1080 -i ArashRawYuv420.yuv \
-vcodec rawvideo -s 1920*1080 -pix_fmt rgb24 -f null /dev/null
The fps increase from 389 to 640 on Intel(R) Core(TM) i7-8700K CPU @ 3.70GHz
Signed-off-by: Ting Fu <ting.fu@intel.com>
Fixes: signed integer overflow: -2147479324 + -32568 cannot be represented in type 'int'
Fixes: 20103/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_GREMLIN_DPCM_fuzzer-5667667579240448
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -53716100 * 256 cannot be represented in type 'int'
Fixes: 20143/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMALOSSLESS_fuzzer-5716604000403456
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Bug #8255 points out a double free error in libwscale/utils.c file.
The double free is because the pointer to cascaded_context of an
sw_context is not set to NULL after freeing it. When the sw_context
is later freed, sws_freeContext is called on the cascaded_context,
causing a double free.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
ff_cbs_insert_unit_data() has two modes of operation: It can insert a
unit with a newly created reference to an already existing AVBuffer; or
it can take a buffer and create an AVBuffer for it. Said buffer will
then become owned by the unit lateron.
A potential memleak/double-free exists in the second case, because if
creating the AVBuffer fails, the function immediately returns, but when
it fails lateron, the supplied buffer will be freed. The caller has no
way to distinguish between these two outcomes. The only such caller
(cbs_jpeg_split_fragment() in cbs_jpeg.c) opted for a potential
double-free.
This commit changes this by explicitly stating that a non-refcounted
buffer will be freed on error. The aforementioned caller has been
brought in line with this.
Fixes CID 1452623.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
To make behavior the same as non-win32 code when the standard error is
redirected. Also restructure the code a bit.
Signed-off-by: Marton Balint <cus@passwd.hu>
Not even FFTW's output is normalized.
This should prevent at least some users from complaining that doing a forward
transform followed by an inverse transform has a mismatching output to the
original input.
apply_palette() would free an AVFrame given to it only via an AVFrame *
(and not via AVFrame **) in three of its four exists (namely in the
normal path and in two error paths). So upon error the caller has no way
to know whether the frame has already been freed or not;
load_apply_palette(), the only caller, opted to free the frame in this
scenario.
This commit changes this by making apply_palette not freeing the frame
at all, which is left to load_apply_palette().
Fixes Coverity issue #1452434.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
lsb bits may go beyond 25 bits, so to be safe use get_bits_long
Signed-off-by: Jai Luthra <me@jailuthra.in>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Affected the ffmpeg-filter_colorkey FATE-test (but only if the C version
of idct8_add is used and not e.g. the x86 SSE2 version).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Fixes: left shift of negative value -107
Fixes: 20398/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_CAVS_fuzzer-5725389278412800
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Adds support for the custom VAG container used by some Simon & Schuster
Interactive games such as Real War, and Real War: Rogue States.
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Adds support for the ADPCM variant used by some Simon & Schuster
Interactive games such as Real War, and Real War: Rogue States.
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The default is not to write SDT and PAT periodically, only in the beginning of
every segment. After this patch the user might override this if needed.
Signed-off-by: Marton Balint <cus@passwd.hu>
Sometimes it has not been checked whether opening the dynamic buffer for
writing Tags fails; this might have led to segfaults.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The original inline assembly and nasm code have the same fps when called by command.
NASM code almost has no impact on the perfromance.
Signed-off-by: Ting Fu <ting.fu@intel.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This fixes the tests filter-refcmp-ssim-yuv and filter-refcmp-ssim-rgb
on i386 after breaking in fcc0424c93.
Signed-off-by: Martin Storsjö <martin@martin.st>
The check_x86asm() checks would force enable these variables on success,
bypassing any --disable-* command line option.
This is important in the case of AVX512, where the relevant define is used
to choose between different values for memory alignment and strides in
some allocations.
Signed-off-by: James Almer <jamrial@gmail.com>
This commit adds a chromatic aberration filter for Vulkan that attempts to
emulate a lens chromatic aberration effect.
For a YUV frame it will instead shift the chroma channels, providing a
simple approximation.
This commit adds a Vulkan filtering infrastructure for libavfilter.
It attempts to abstract as much as possible of the Vulkan API from filters.
The way the hwcontext and the framework are designed permits for parallel,
non-CPU-blocking filtering throughout, with the exception of up/downloading
and mapping.
As we find ourselves wanting a way to transfer frames between
HW devices (or more realistically, between APIs on the same device),
it's desirable to have a way to describe the relationship. While
we could imagine introducing a `hwtransfer` filter, there is
almost no difference from `hwupload`. The main new feature we need
is a way to specify the target device. Having a single device
for the filter chain is obviously insufficient if we're dealing
with two devices.
So let's add a way to specify the upload target device, and if none
is specified, continue with the existing behaviour.
We must also correctly preserve the sw_format on such a transfer.
This commit adds the necessary code to initialize and use a Vulkan device
within the hwcontext libavutil framework.
Currently direct mapping to VAAPI and DRM frames is functional, and
transfers to CUDA and native frames are supported.
Lets hope the future Vulkan video decode extension fits well within this
framework.
We are beginning to consider scenarios where a given HW Context
may be able to transfer frames to another HW Context without
passing via system memory - this would usually be when two
contexts represent different APIs on the same device (eg: Vulkan
and CUDA).
This is modelled as a transfer, as we have today, but where both
the src and the dst are hardware frames with hw contexts. We need
to be careful to ensure the contexts are compatible - particularly,
we cannot do transfers where one of the frames has been mapped via
a derived frames context - we can only do transfers for frames that
were directly allocated by the specified context.
Additionally, as we have two hardware contexts, the transfer function
could be implemented by either (or indeed both). To handle this
uncertainty, we explicitly look for ENOSYS as an indicator to try
the transfer in the other direction before giving up.
Previously, the default palette would always be used.
Now, we can accept a custom palette, just like dvdsubdec does.
Signed-off-by: Michael Kuron <michael.kuron@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When no codebook is used, huff_lsbs can be more than 24 and still decode to
original values once filters are applied.
Signed-off-by: Jai Luthra <me@jailuthra.in>
* fix a possible memory leak (apply_filter returned before freeing)
* use apply_filters in process_major_frame
* revert back to checking bounds with 24 bitdepth, as huff offset takes
care of it
Signed-off-by: Jai Luthra <me@jailuthra.in>
huff offset wasn't always within the bounds before, which lead to
corrupt encoding that didn't always trigger lossless check failures
Signed-off-by: Jai Luthra <me@jailuthra.in>
A value of 24 and above can collaps the range to 0 which would not work.
Fixes: Timeout (75sec -> 21sec)
Fixes: 18707/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_LAGARITH_fuzzer-5708950892969984
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -2147407150 + -1871606 cannot be represented in type 'int'
Fixes: 18702/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-5679095417667584
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This changes the separator character from comma to colon, but since this option
was only added recently I think it should be done for consistency with other
similar options.
Signed-off-by: Marton Balint <cus@passwd.hu>
This commit reuses the configuration options for VP8 that enables
temporal scalability for VP9. It also adds a way to enable three
preset temporal structures (refer to the documentation for more
detail) that can be used in offline encoding.
Signed-off-by: James Zern <jzern@google.com>
Fixes: signed integer overflow: 2147482897 + 2048 cannot be represented in type 'int'
Fixes: 19240/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_RALF_fuzzer-5743240326414336
Fixes: 19869/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_RALF_fuzzer-5150136636538880
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: inf is outside the range of representable values of type 'int'
Fixes: signed integer overflow: -9223372036854775808 - 1 cannot be represented in type 'long'
Fixes: 19316/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMAVOICE_fuzzer-5677369365102592
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This limit is roughly based on the bitreader limit, its likely a much tighter limit
could be used
Fixes: left shift of 1965039647 by 1 places cannot be represented in type 'int'
Fixes: 19545/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMAVOICE_fuzzer-5695391899320320
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: left shift of 79 by 28 places cannot be represented in type 'int'
Fixes: 20202/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_HAP_fuzzer-5719004081815552
Fixes: 20219/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_HAP_fuzzer-5641738677125120
Fixes: 20389/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_HAP_fuzzer-5680721517871104
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
recvfrom() is not a cancellation point in pthreads-win32, see
https://sourceware.org/pthreads-win32/manual/pthread_cancel.html
In order to be able to cancel the reader thread on Win32 properly we first
shutdown the socket then call CancelIoEx to abort pending IO. Subsequent
recvfrom() calls will fail with WSAESHUTDOWN causing the thread to exit.
Fixes ticket #5717.
Signed-off-by: Marton Balint <cus@passwd.hu>
Before this commit v4l2_m2m used two different logging contexts (from
V4L2m2mPriv and AVCodecContext). For consistency always use AVCodecContext.
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
Before this commit s->avctx == NULL was used to infer that an encoder is
being initialzed. Code readability has been improved by directly using
!av_codec_is_decoder() instead.
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
There was no consensus about separating AVExprState from AVExpr so here is a
minimal patch using the existing AVExpr to fix ticket #7528.
Signed-off-by: Marton Balint <cus@passwd.hu>
this usecase will cause a division by zero trap:
1. dashenc has received one frame
2. os->max_pts and os->start_pts have same value
3. delta between max_pts and start_pts is 0
4. av_rescale_q(0, x, y) returns 0
5. this value is used as denominator in division
6. Bang! -> segfault
this fix checks that max_pts > start_pts.
the fix has been tested and works.
Signed-off-by: Alfred E. Heggestad <alfred.heggestad@gmail.com>
Reviewed-by: Jeyapal, Karthick <kjeyapal@akamai.com>
Signed-off-by: James Almer <jamrial@gmail.com>
continue is explicitly disallowed for GET_UTF8, so let's fix that as well.
Fixes crash with invalid UTF8 sequences.
Signed-off-by: Marton Balint <cus@passwd.hu>
APE in its highest compression mode is really slow so even one frame
of millions of samples takes a long time
Fixes: Timeout (too long -> 3sec)
Fixes: 19937/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-5751668818051072
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: bypassing of checks and assertion failure
Fixes: asan_1003879.mp4
Found-by: Clusterfuzz + asan
Reported-by: Thomas Guilbert <tguilbert@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: left shift of 3329 by 20 places cannot be represented in type 'int'
Fixes: signed integer overflow: -199378355 + -1948950833 cannot be represented in type 'int'
Fixes: 19837/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMALOSSLESS_fuzzer-5752565837070336
Fixes: 19839/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMALOSSLESS_fuzzer-5767483265122304
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -2143289344 - 6246400 cannot be represented in type 'int'
Fixes: 19239/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-5173755680915456
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
IFF-PBM-HAM6 can read out of array without this overallocation
Fixes: Out of array read
Fixes: 19752/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_IFF_ILBM_fuzzer-5675331403120640
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
553 -> 332 sec
Testcase: 20280/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SCREENPRESSO_fuzzer-6238663432470528
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
In the Libav commit cae448cf, the opaque of every AVIOContext opened
by ffio_fdopen() (which is used internally by avio_open() and avio_open2())
changed: It was a simple pointer to an URLContext before, but now it was
a structure (namely AVIOInternal) containing a pointer to an URLContext
as its only member. The next commits (namely 8c0ceafb and ec4c4839) added
members to AVIOInternal to allow white-/blacklisting of protocols.
But these two commits were never merged into FFmpeg (they were only
merged as no-ops in 510046c2 and 063b26d3), because FFmpeg chose
a different way to implement this (in 93629735); and so our AVIOInternal
still has exactly one member.
This of course means that it is unnecessary to use AVIOInternal as
opaque as it is just adding a level of indirection (not only pointer
dereference, but also wrapper functions). Therefore this commit
removes AVIOInternal entirely and essentially reverts cae448cf.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
fix crash when used the command like:
- ffmpeg -h protocol
- ffmpeg -h protocol=
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Besides the obvious advantages this also fixes a potential memleak:
If only one of the arrays had been successfully allocated, the other
would leak. This also fixes Coverity issues #1440386 and #1440387.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Write a few numbers directly via AV_WB32 instead of using an AVIOContext
(that is initialized only for this very purpose) to write these numbers
at known offsets into a fixed buffer.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The Matroska Projection master element has such a small maximum length
that it can always be written with a length field of length one.
So it is unnecessary to first write the element into a dynamic buffer to
get the accurate length in order not to waste bytes on the length field.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Should fix fate-lavf-fate-av1.mkv failures on builds without an AV1 decoder.
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
If no error occurs and this AVPacketList is used at all, its packet
substructure will be overwritten and its next pointer explicitly set, so
every field will still be initialized even when using av_malloc.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
In the common case that the input packet was already refcounted,
ff_interleave_add_packet would allocate a new AVPacketList, use
av_packet_ref to create a new reference to the buffer for the
AVPacketList's packet, interleave the packet and finally unreference
the original input packet.
This commit changes this: It uses av_packet_move_ref to transfer
the packet to its destination. In case the input packet is refcounted,
this saves an allocation and a free (of an AVBufferRef); if not, the
packet is made refcounted before moving it. When the input packet has
side data, one saves even more than one allocation+free.
Furthermore, when the packet is in reality an uncoded frame, a hacky
ad-hoc variant of av_packet_move_ref has been employed. Not any more.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes memleaks when the trailer is never written (e.g. if the call to
gxf_write_map_packet() at the end of gxf_write_header() fails).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes memleaks when allocating the private data of the timecode_track
fails or when the trailer is never written.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
In order to use ff_audio_rechunk_interleave() (a special interleavement
function for situations where the ordinary "interleave by dts" is not
appropriate), the AVStreams must have private data and this private data
must begin with an AudioInterleaveContext which contains a fifo that may
need to be freed and when ff_audio_interleave_close() was called, it just
assumed that everything has been properly set up, i.e. that every streams
priv_data exists. This implies that this function can not be called from
the deinit function of a muxer, because such functions might be called
if the private data has not been successfully allocated. In order to
change this, add a check for whether the private data exists before
trying to free the fifo in it.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The muxing context has already been zeroed when it was allocated, hence
it is unnecessary to do it again.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The old write_trailer only freed memory, so it is better to make a
dedicated deinit function out of it. Given that this function will also
be called when writing the header fails, one can also remove code that
frees already allocated fifos when allocating another one fails.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Certain types of OBUs are stripped away before muxing into Matroska and
ISOBMFF; there are two functions to do this: One that outputs by
directly writing in an AVIOContext and one that returns a freshly
allocated buffer with the units not stripped away copied into it.
The latter option is bad for performance, especially when the input
does already not contain any of the units intended to be stripped away
(this covers typical remuxing scenarios). Therefore this commit changes
this by avoiding allocating and copying when possible; it is possible if
the OBUs to be retained are consecutively in the input buffer (without
an OBU to be discarded between them). In this case, the caller receives
the offset as well as the length of the part of the buffer that contains
the units to be kept. This also avoids copying when e.g. the only unit
to be discarded is a temporal delimiter at the front.
For a 22.7mb/s file with average framesize 113 kB this improved the time
for the calls to ff_av1_filter_obus_buf() when writing Matroska from
313319 decicycles to 2368 decicycles; for another file with 1.5mb/s
(average framesize 7.3 kB) it improved from 34539 decicycles to 1922
decicyles. For these files the only units that needed to be stripped
away were temporal unit delimiters at the front.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
ff_av1_filter_obus_buf() and ff_avc_parse_nal_units_buf() both have a
pointer-to-pointer parameter which they use to pass a newly allocated
buffer to the caller. And both functions freed what this pointer points to
before overwriting it. But no caller of these functions used this feature,
but some had to initialize the pointer just because of this. So remove
it and update the documentation of ff_av1_filter_obus_buf() wrt this fact.
ff_hevc_annexb2mp4_buf in contrast did not free the pointer. This has been
documented, too.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Both ISOBMFF as well as Matroska require certain OBUs to be stripped
before muxing them. There are two functions for this purpose; one writes
directly into an AVIOContext, the other returns a freshly allocated
buffer with the undesired units stripped away.
The latter one actually relies on the former by means of a dynamic
buffer. This has several drawbacks: The underlying buffer might have to
be reallocated multiple times; the buffer will eventually be
overallocated; the data will not be directly copied into the final
buffer, but rather first in the write buffer (in chunks of 1024 byte)
and then written in these chunks. Moreover, the API for dynamic buffers
is defective wrt error checking and as a consequence, the earlier code
would indicate a length of -AV_INPUT_BUFFER_PADDING_SIZE on allocation
failure, but it would not return an error; there would also be no error
in case the arbitrary limit of INT_MAX/2 that is currently imposed on
dynamic buffers is hit.
This commit changes this: The buffer is now parsed twice, once to get
the precise length which will then be allocated; and once to actually
write the data.
For a 22.7mb/s file with average framesize 113 kB this improved the time
for the calls to ff_av1_filter_obus_buf() when writing Matroska from
753662 decicycles to 313319 decicycles (based upon 50 runs a 2048 frames
each); for another 1.5mb/s file (with average framesize of 7.3 kB) it
improved from 79270 decicycles to 34539 decicycles (based upon 50 runs a
4096 frames).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
If this is not done, the avio_write() calls will cause segfaults
immediately afterwards on error.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The output size is already returned via a pointer argument, so there is
no need to return it via the ordinary return value as well. The
rationale behind this is to not poison the return value on success.
It also unifies the behaviour of the *_buf-functions for AVC, AV1 and
HEVC.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
ff_hevc_annexb2mp4_buf() could indicate an error, yet leave cleaning
after itself to the caller, so that a caller could not simply return the
error, but had to free the buffer first.
(Given that all current callers have set filter_ps = 0, this error can
currently not be triggered.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Add {, } in situations like
if ()
...
else if ()
/* Comment */
...
else ...
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
This is needed especially for AV1: If a reformatting error happens (e.g.
if the length field of an OBU contained in the current packet indicates
that said OBU extends beyond the current packet), the data pointer is
still NULL, yet the size is unchanged, so that writing the data leads
to a segmentation fault.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Adds support for the ADPCM variant used by some Argonaut Games' games,
such as 'Croc! Legend of the Gobbos', and 'Croc 2'.
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
It is a common mistake that people only increase fifo_size when they experience
drops, unfortunately this does not help for higher bitrate (> 100 Mbps) streams
when the reader thread simply might not receive the packets in time (especially
under high CPU load) if the default 64 KB of kernel buffer size is used.
New default is determined so that common linux systems can set this buffer size
without tuning kernel parameters.
Signed-off-by: Marton Balint <cus@passwd.hu>
As per the PIFF standard, the timescale of 10000000
is recommended but not mandatory, so don't override
the user-set value.
A warning is shown for non-recommended values.
When clang works in MSVC mode, it does have the _byteswap_ulong
builtin, but one has to include stdlib.h before using it.
Signed-off-by: Martin Storsjö <martin@martin.st>
SetConsoleTextAttribute used to be unavailable for Windows Store apps,
but is available to them now. But GetStdHandle still is unavailable,
thus make sure to check for both functions before using code that
requires both.
Signed-off-by: Martin Storsjö <martin@martin.st>
by replacing it with a multiplication. Said multiplication can't
overflow an int32_t because lpc_coefs is limited to 16 bit precision.
Fixes the FACE-test acodec-ra144 as well as part of #8217.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This allows the fuzzer to target meaningfull codec tags instead
of hunting the 4gb space, which it seems to have problems with.
Suggested-by: James
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Also add parentheses to some lines to make the operator precedence
clearer.
This affected the FATE-tests vsynth*-jpeg2000 and vsynth*-jpeg2000-97
(where * ranges over { 1, 2, 3, _lena }) as well as ticket #7983.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Replace "(a * (1 << shift) * b + c) >> shift" by "a * b + (c >> shift)".
It is equivalent to the old code because a is in the range of uint16_t,
shift is 12 and b is effectively a signed 4-bit number, so that no
overflow/truncation of high bits happens during the multiplication
(overflow would be undefined anyway).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Replace "((a << shift) + b) >> shift" by "a + (b >> shift)". This avoids
a left shift which also happens to trigger undefined behaviour in case "a"
is negative. This affected the FATE-tests acodec-adpcm-adx and
acodec-adpcm-adx-trellis; it also fixes ticket #8008.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Affected the adpcm-afc, adpcm-ea-1, adpcm-ea-2, adpcm-ea-maxis-xa,
adpcm-thp and ea-cdata FATE-tests. Also fixes ticket #8487.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Currently when hls_enc is active and there are multiple variant stream
outputs, default key file url construction does not work, because it is
based on the FormatContext' url field. But in case of multiple variant
streams, it contains the variant m3u8 output playlist url that contains
the %v placeholder. So the result key file url will hold the %v
placeholder causing run time error message about "could not write the
key file".
This patch correct this behaviour, and use the master playlist url for
constructing the output key file url when master playlist is vailable.
Signed-off-by: Bela Bodecs <bodecsb@vivanet.hu>
When program_date_time flag is present, in m3u8 playlist file each
segment has a corresponding EXT-X-PROGRAM-DATE-TIME value. The intial
program-date-time value is the actual current time at init and
each new segment increments this value by its duration. When append_list
flags is also present, existing playlist parsing by hls_append_segment
treats existing segments as new segments regarding the program-date-time
calculation. But it should not do that, because this way all real the
new segments' EXT-X-PROGRAM-DATE-TIME values will be shifted erroneously
by the sum duration of existing segments. Instead it should have
decremented the initial program-date-time value by its duration. This
would ensure that
the first new segment's program-date-time value had the current time as
it is expected.
This patch corrects this behaviour and prevent existing segments to
increment the value of initial_prog_date_time variable but decrements
it.
Reviewed-by: Steven Liu <lq@onvideo.cn>
Signed-off-by: Bela Bodecs <bodecsb@vivanet.hu>
Also simplify it and make it always log the error.
This fixes for example the image2 muxer when used with an URL which also
contains the protocol:
ffmpeg -f lavfi -i testsrc -vframes 10 -atomic_writing 1 file:out%d.png
Signed-off-by: Marton Balint <cus@passwd.hu>
"It is a requirement of bitstream conformance that the value returned from the leb128 parsing process is less than or equal
to (1 << 32) - 1."
Fixes: assertion failure
Fixes: 19293/clusterfuzz-testcase-minimized-ffmpeg_BSF_TRACE_HEADERS_fuzzer-5749508361420800
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Both S16LE as well as DTS can have lots of 0 bytes in silent segments
Using these results in error. Thus this patch skips 0 bytes in
comparission.
Fixes Ticket6561
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
HEVC standard supports multi-layer streams (ITU-T H.265 02/2018 Annex
F). Each NAL unit belongs to a particular layer defined by nuh_layer_id
in the header.
Currently, all NAL units that do not belong to a base layer are
automatically removed in ff_h2645_packet_split(). Some data may
therefore be lost when future filters/decoders are designed to support
multi-layer streams.
A better approach is to forward nuh_layer_id > 0 packets and let blocks
down the chain decide how to process them. The condition to remove
packets has been moved to hevcdec and cbs.
Found-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
This will likely also fix CID 1452427, a false positive resulting from
Coverity thinking that av_dict_set() automatically frees its key and
value parameters (even without the AV_DICT_DONT_STRDUP_* flags).
(AV_DICT_APPEND and AV_DICT_DONT_STRDUP_VAL are compatible with each
other since a8c5b455, so we can reset this flag here. It has originally
been removed in 0dc66553 when appending was added.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
libx265.c references a member x265_picture.quantOffsets (for ROI
support) which was added in X265_BUILD 70. Increase the minimum libx265
version to fix compilation.
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
Reviewed-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When shifting the already written data in order to write the keyframe
index, the flv muxer would first store the pre-shift size, then
calculate how big the index will be eventually, then perform some seeks
to update some size fields, then seek back to the end of the file to get
the new position, followed by a seek to the position where writing will
really start. Seeking back to the (already known) end position (that is
actually used to perform this seek) to get the end position is of course
unnecessary. It has been removed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When the faststart option for the mov/mp4 muxer is used, the current
position (i.e. the size of the already written data pre-shifting) was
evaluated twice: First in an initialization and then again later,
overwriting the first value without having ever touched it. So remove
the initialization.
Also, the clone of this code in the Sega FILM muxer behaves the same and
has been treated the same.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
s337m_get_offset_and_codec does not make use of
AVFormatContext: AVClass is enough for logging.
Will facilitate further use from outside
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Allows to set an intended target latency while streaming that clients can use
to measure when using low latency mode.
Signed-off-by: James Almer <jamrial@gmail.com>
In combination with the streaming option it constrains the value of a few elements,
to prevet clients from buffering too much data before starting presentation.
Signed-off-by: James Almer <jamrial@gmail.com>
If not available, set flags to 24 (bits 4 and 5), to signal the wallclock value
is read at the time of writing the atom.
Signed-off-by: James Almer <jamrial@gmail.com>
Implemented as as a frag_duration muxer option and key=value entry in the
adaptation_sets muxer option. It has the same syntax as the seg_duration option.
A new frag_type option is also introduced to select the kind of fragmentation.
Signed-off-by: James Almer <jamrial@gmail.com>
Implemented as as a seg_duration key=value entry in the adaptation_sets muxer
option.
It has the same syntax as the global seg_duration option, and has precedence
over it if set.
Signed-off-by: James Almer <jamrial@gmail.com>
Since bae8844e35, the packet is automatically unreferenced in
ff_read_packet() when an error is returned; but the documentation of
this of AVInputFormat.read_packet has not been updated accordingly.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The current code only checks when writing the trailer whether the video
format and Codec ID are actually compatible with the container. At this
point, a lot of data will already have been written (in vain, of
course), so check during the init function instead.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Up until now, the Sega FILM muxer complained if the first stream wasn't a
video stream that there is no video stream at all which is of course
nonsense. So postpone this check.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
it's stranage to use option "level" in runtime change path but used
"quality" in option, add "quality" in runtime change path, it's more
intuitive and keep the "level" for compatibility.
Reviewe-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
Fixes: negation of -2147483648 cannot be represented in type 'int'; cast to an unsigned type to negate this value to itself
Fixes: 19235/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ADPCM_IMA_EA_EACS_fuzzer-5680878952382464
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
In order to indicate that the frames in a BlockGroup are not keyframes,
one has to add a ReferenceBlock element containing the timestamp of a
referenced Block that has already been written. The timestamp ought to be
relative to the timestamp of the Block it is attached to. Yet the
Matroska muxer used the relative timestamp of the preceding Block of the
track, i.e. the timestamp of the preceding block relative to the
timestamp of the Cluster containing said block (that need not be the
Cluster containing the current Block). This has been fixed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
libavformat/img2.h: New field export_path_metadata to
VideoDemuxData to only allow the use of the extra metadata
upon explicit user request, for security reasons.
libavformat/img2dec.c: Modify image2 demuxer to make available
two special metadata entries called lavf.image2dec.source_path
and lavf.image2dec.source_basename, which represents, respectively,
the complete path to the source image for the current frame and
the basename i.e. the file name related to the current frame.
These can then be used by filters like drawtext and others. The
metadata fields will only be available when explicitly enabled
with image2 option -export_path_metadata 1.
doc/demuxers.texi: Documented the new metadata fields available
for image2 and how to use them.
doc/filters.texi: Added an example on how to use the new metadata
fields with drawtext filter, in order to plot the input file path
to each output frame.
Usage example:
ffmpeg -f image2 -export_path_metadata 1 -pattern_type glob
-framerate 18 -i '/path/to/input/files/*.jpg'
-filter_complex drawtext="fontsize=40:fontcolor=white:
fontfile=FreeSans.ttf:borderw=2:bordercolor=black:
text='%{metadata\:lavf.image2dec.source_basename\:NA}':x=5:y=50"
output.avi
Fixes#2874.
Signed-off-by: Alexandre Heitor Schmidt <alexandre.schmidt@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
This will likely also fix CID 1452574 and 1452565, false positives
resulting from Coverity thinking that av_dict_set() automatically
frees its key and value parameters (even without the
AV_DICT_DONT_STRDUP_* flags).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The tests for concat use this option which is scheduled for removal and
does nothing any more. So remove it; otherwise, these tests would fail
at the next major version bump.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Keep all the existing data fields as they are (there's lots and
lots of nontrivial calculation and heuristics based on them in
their current form), but derive the duration as the difference
between the pts of the first packet to the maximum pts+duration
(not necessarily the last packet); use this duration in any box
where the actual presentation duration is supposed to be.
Fixes: 8420
Signed-off-by: Martin Storsjö <martin@martin.st>
If the size of the input packet is zero, av_grow_packet() used to call
av_new_packet() which would initialize the packet and (in particular)
reset the pos field. This behaviour (which was never documented and
arguably always contradicted the documented behaviour) was changed in
2fe04630. This means that it is unnecessary to save and restore the
packet's position in append_packet_chunked().
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This fixes ticket #7997 as well as the vsynth*-prores_# FATE-tests
(where * ranges over { 1, 2, 3, _lena } and # over { , _int, _444,
_444_int }).
(Given that prev_dc is in the range -0xC000..0x3FFF, no overflow can
happen upon multiplication with 2.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
After 06ec9c4746 we check for these
functions in configure (which will succeed in cygwin), but cmdutils.c
only includes windows.h if _WIN32 is defined (which it isn't in cygwin).
Retain the old intent from before 06ec9c4746,
that these functions only would be used when _WIN32 is defined, while
only using them if configure has agreed that they do exist.
Signed-off-by: Martin Storsjö <martin@martin.st>
The Dash muxer uses submuxers and when one such submuxer has been allocated,
it is initially only stored in a temporary variable. Therefore it leaks
if an error happens between the allocation and storing it permanently.
This commit changes this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: "Jeyapal, Karthick" <kjeyapal@akamai.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This commit improves returned error codes by forwarding error codes. In
some instances, the hardcoded returned error codes made no sense at all:
The normal error code for failure of av_new_packet() is AVERROR(ENOMEM),
yet there were instances where AVERROR(EIO) was returned.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Converting explicit avio_flush() calls helps us to buffer more data and avoid
flushing the IO context too often which causes reduced IO throughput for
non-streamed file output.
The user can control FLUSH_POINT flushing behaviour using the -flush_packets
option, the default typically means to flush unless a non-streamed file output
is used, so this change should have no adverse effect on streaming even if it
is assumed that after an avio_flush() the output buffer is clean so small
seekbacks within the output buffer will work even when the IO context is not
seekable.
Signed-off-by: Marton Balint <cus@passwd.hu>
These instances are simply redundant or present because avio_flush() used to be
required before doing a seekback. That is no longer the case, aviobuf code does
the flush automatically on seek.
This only affects code which is either disabled for streaming IO contexts or
does no seekbacks after the flush, so this change should have no adverse effect
on streaming.
Signed-off-by: Marton Balint <cus@passwd.hu>
Removing explicit avio_flush() calls helps us to buffer more data and avoid
flushing the IO context too often which causes reduced IO throughput for
non-streamed file output.
The user can control flushing behaviour at the end of every packet using the
-flush_packets option, the default typically means to flush unless a
non-streamed file output is used.
Therefore this change should have no adverse effect on streaming, even if it is
assumed that a new packet has a clean buffer so small seekbacks within the
output buffer work even when the IO context is not seekable.
Signed-off-by: Marton Balint <cus@passwd.hu>
To make it consistent with other muxers.
The user can still control the generic flushing behaviour after write_header
(same way as after packets) using the -flush_packets option, the default
typically means to flush unless a non-streamed file output is used.
Therefore this change should have no adverse effect on streaming, even if it is
assumed that the first packet has a clean buffer, so small seekbacks within the
output buffer work even when the IO context is not seekable.
Signed-off-by: Marton Balint <cus@passwd.hu>
The following is a python script to halve the value of the gray
image. It demos how to setup and execute dnn model with python+tensorflow.
It also generates .pb file which will be used by ffmpeg.
import tensorflow as tf
import numpy as np
from skimage import color
from skimage import io
in_img = io.imread('input.jpg')
in_img = color.rgb2gray(in_img)
io.imsave('ori_gray.jpg', np.squeeze(in_img))
in_data = np.expand_dims(in_img, axis=0)
in_data = np.expand_dims(in_data, axis=3)
filter_data = np.array([0.5]).reshape(1,1,1,1).astype(np.float32)
filter = tf.Variable(filter_data)
x = tf.placeholder(tf.float32, shape=[1, None, None, 1], name='dnn_in')
y = tf.nn.conv2d(x, filter, strides=[1, 1, 1, 1], padding='VALID', name='dnn_out')
sess=tf.Session()
sess.run(tf.global_variables_initializer())
graph_def = tf.graph_util.convert_variables_to_constants(sess, sess.graph_def, ['dnn_out'])
tf.train.write_graph(graph_def, '.', 'halve_gray_float.pb', as_text=False)
print("halve_gray_float.pb generated, please use \
path_to_ffmpeg/tools/python/convert.py to generate halve_gray_float.model\n")
output = sess.run(y, feed_dict={x: in_data})
output = output * 255.0
output = output.astype(np.uint8)
io.imsave("out.jpg", np.squeeze(output))
To do the same thing with ffmpeg:
- generate halve_gray_float.pb with the above script
- generate halve_gray_float.model with tools/python/convert.py
- try with following commands
./ffmpeg -i input.jpg -vf format=grayf32,dnn_processing=model=halve_gray_float.model:input=dnn_in:output=dnn_out:dnn_backend=native out.native.png
./ffmpeg -i input.jpg -vf format=grayf32,dnn_processing=model=halve_gray_float.pb:input=dnn_in:output=dnn_out:dnn_backend=tensorflow out.tf.png
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
do not request AVFrame's format in vf_ddn_processing with 'fmt',
but to add another filter for the format.
command examples:
./ffmpeg -i input.jpg -vf format=bgr24,dnn_processing=model=halve_first_channel.model:input=dnn_in:output=dnn_out:dnn_backend=native -y out.native.png
./ffmpeg -i input.jpg -vf format=rgb24,dnn_processing=model=halve_first_channel.model:input=dnn_in:output=dnn_out:dnn_backend=native -y out.native.png
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
The documentation of both avio_open() as well as avio_open2() states
that on failure, the pointer to an AVIOContext given to this function
(via a pointer to a pointer to an AVIOContext) will be set to NULL. Yet
it didn't happen upon failure of ffurl_open_whitelist() or when allocating
the internal buffer failed. This commit changes this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
"VAProcFilterParameterBufferHDRToneMapping" was defined in libva 2.4.1, which will lead to
build failure for the filter tonemap_vaapi for libva 2.3.0 with current check. This patch
is to fix this build error.
Signed-off-by: Xinpeng Sun <xinpeng.sun@intel.com>
Currently, the decoder checks the 128th value of the 4th quant table during
while deriving the context on each sample, in order to speed itself up. This
is due to relying on the behavior of FFmpeg's FFV1 encoder, in which if that
value is zero, the entire 4th and 5th quant tables are assumed to be entirely
zero.
This does not match the FFV1 spec, which has no such restriction, and after
some discussion, it was decided to fix FFmpeg to abide by the spec, rather
than change the spec.
We will now check whether the 4th and 5th quant tables are zero properly,
by checking the 128th valye of both tables (which means they are zero due
to the way they're coded in the bitstream).
For further context, the FFV1 issue in question is located at:
https://github.com/FFmpeg/FFV1/issues/169
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This patch rewrites the innermost loop of ff_yuv2planeX_8_neon to avoid zips and
horizontal adds by using fused multiply adds. The patch also uses ld1r to load
one element and replicate it across all lanes of the vector. The patch also
improves the clipping code by removing the shift right instructions and
performing the shift with the shift-right narrow instructions.
I see 8% difference on an m6g instance with neoverse-n1 CPUs:
$ ffmpeg -nostats -f lavfi -i testsrc2=4k:d=2 -vf bench=start,scale=1024x1024,bench=stop -f null -
before: t:0.014015 avg:0.014096 max:0.015018 min:0.013971
after: t:0.012985 avg:0.013013 max:0.013996 min:0.012818
Tested with `make check` on aarch64-linux.
Signed-off-by: Sebastian Pop <spop@amazon.com>
Reviewed-by: Clément Bœsch <u@pkh.me>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
ff_filter_frame() always frees the frame in case of error, so we don't
need to free the frame after ff_filter_frame() fails.
Fix CID 1457230.
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
In case of failure, all the successfully set entries are stored in
*pm. We need to manually free the created dictionary to avoid
memory leak.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
In case of failure, all the successfully set entries are stored in
*pm. We need to manually free the created dictionary to avoid
memory leak.
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
In case of failure, all the successfully set entries are stored in
*pm. We need to manually free the created dictionary to avoid
memory leak.
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
In case of failure, all the successfully set entries are stored in
*pm. We need to manually free the created dictionary to avoid
memory leak.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
Though this patch to fix ticket #6668, I belive it
is unnecessary to set SLICE_FLAG_ALLOW_FIELD flag to other
hwaccels(dxva, vdpau, etc). Please also refer the orginal comment
of 9cb150c9ab
Should also fix ticket #8442.
Signed-off-by: Zhong Li <zhong.li@intel.com>
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
This will likely also fix CID 1452562, a false positive resulting from
Coverity thinking that av_dict_set() automatically frees its key and
value parameters (even without the AV_DICT_DONT_STRDUP_* flags).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The Matroska muxer currently does not check the return value of
ff_isom_write_hvcc(), the function used to write mp4-style
HEVC-extradata as Matroska also uses it. This was intentionally done in
7a5356c72 to allow remuxing from mpeg-ts.
But if ff_isom_write_hvcc() fails, it has not output anything and the
file ends up without CodecPrivate and, if the input was Annex B, with
Annex B data, which is against the spec. So check the return value
again.
The underlying issue of not having extradata seems to have been fixed by
the introduction of the extract_extradata bitstream filter.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: "mypopy@gmail.com" <mypopy@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
to its actual behaviour: That it uses the least amount of bytes unless
overridden.
The current documentation leaves it undefined how many bytes will be used
when no number to use has been given explicitly. But several estimates
(used to write EBML Master elements with a small length field) require
this number to be the least amount of bytes to work. Therefore change
the documentation; and remove a comment about writing length fields
indicating "unkown length". It has been outdated since 0580a122.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When vobsub_read_packet() reads a packet, it uses a dedicated AVPacket
to get the subtitle timing and position from an FFDemuxSubtitlesQueue
(which has been filled with this data during reading the idx file in
vobsub_read_header); afterwards the actual subtitle data is read into
the packet destined for output and the timing and position are copied
to this packet. Afterwards, the local packet is unreferenced.
This can be simplified: Simply use the output packet to get the timing
and position from the FFDemuxSubtitlesQueue. The packet's size will be
zero afterwards, so that it can be directly used to read the actual
subtitle data. This makes copying the packet fields as well as
unreferencing the local packet unecessary and also removes an instance
of usage of sizeof(AVPacket) in libavformat.
The only difference is that the returned packet will already be flagged
as a keyframe. This currently only happens in compute_pkt_fields().
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 2147464192 + 21176 cannot be represented in type 'int'
Fixes: 19042/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_FFWAVESYNTH_fuzzer-5719828090585088
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: shift exponent -1 is negative
Fixes: out of array read
Fixes: 19028/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_COOK_fuzzer-5759766471376896
Fixes: 19037/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_COOK_fuzzer-5734106625474560
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 47875596 * 45 cannot be represented in type 'int'
Fixes: 19082/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMALOSSLESS_fuzzer-5687766512041984
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 1721520852 + 1721520852 cannot be represented in type 'int'
Fixes: 18346/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SONIC_fuzzer-5709623893426176
Fixes: 18753/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SONIC_fuzzer-5663299131932672
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
These are checked for early in avcodec_open2() and do not really test the decoder
but instead waste resources which could be better spend fuzzing the actual decoder
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When a Matroska Block is only stored in compressed form, the size of
the uncompressed block is not explicitly coded and therefore not known
before decompressing it. Therefore the demuxer uses a guess for the
uncompressed size: The first guess is three times the compressed size
and if this is not enough, it is repeatedly incremented by a factor of
three. But when this happens with lzo, the decompression is neither
resumed nor started again. Instead when av_lzo1x_decode indicates that x
bytes of input data could not be decoded, because the output buffer is
already full, the first (not the last) x bytes of the input buffer are
resent for decoding in the next try; they overwrite already decoded
data.
This commit fixes this by instead restarting the decompression anew,
just with a bigger buffer.
This seems to be a regression since 935ec5a1.
A FATE-test for this has been added.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
This test tests that demuxing ProRes that is muxed like it should be in
Matroska (i.e. with the first header ("icpf") atom stripped away) works;
it also tests bz2 decompression as well as the handling of
unknown-length clusters.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
This allows integrating box blur style filters in geq.
Without this computing the mean of an area in geq would have been excessivly slow
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
video is allocated before parameters like bpp are read.
Fixes: out of array access
Fixes: 19084/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_IFF_ILBM_fuzzer-5718556033679360
Fixes: 19465/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_IFF_ILBM_fuzzer-5759908398235648
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes#8079
During initialization of a v4l2m2m device, the configured pix_fmt can be
different to the pix_fmt of the encoder (i.e. avctx->pix_fmt).
For example on the Odroid XU4:
./ffmpeg -f lavfi -i yuvtestsrc -codec:v h264_v4l2m2m out.h264
will configure the v4l2 encoder to pix_fmt nv21, whereas the input
frames will be yuv444p.
This commit checks that the configured v4l2 pix_fmt on device is the
same as avctx->pix_fmt. If they are different the initialization fails
and an error is returned. Tested on RPI4 and Odroid XU4.
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
When the user decides they do not want to to send the Icy-MetaData
header, this should be respected for all requests, not just the first
one.
Fix#5578
Reviewed-by: Liu Steven <lq@chinaffmpeg.org>
Signed-off-by: James Almer <jamrial@gmail.com>
When the user decides they do not want to to send the Icy-MetaData
header, this should be respected for all requests, not just the first
one.
Fix#5578
Reviewed-by: Liu Steven <lq@chinaffmpeg.org>
Signed-off-by: James Almer <jamrial@gmail.com>
Up until now, the microdvd demuxer uses av_strdup() to allocate the
extradata from a string; its length is set to strlen() + 1, i.e.
including the \0 at the end. Upon remuxing, the muxer would simply copy
the extradata at the beginning, including the \0.
This commit changes this by not adding the \0 to the size of the
extradata; the muxer now delimits extradata by inserting a \n. This
required to change the subtitles-microdvd-remux FATE-test.
Furthermore, the extradata is now allocated with zeroed padding.
The microdvd decoder is not affected by this, as it didn't use the size
of the extradata at all, but treated it as a C-string.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
While the function adding a new element to the keyframe index checked
the allocation, the caller didn't check the return value. This has been
changed. To do so, the return value has been changed to an ordinary ret
instead of pb->error. This doesn't pose a problem, as write_packet() in
mux.c already checks for write errors (since 9ad1e0c1).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The write_trailer function doesn't write anything anyway. It only frees
memory.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Also by wrapping the SHM buffer in an AVBufferRef we eliminate yet another
possible memcpy improving performance.
Signed-off-by: Marton Balint <cus@passwd.hu>
In order to access the original opaque parameter of a buffer in the buffer
pool. (The buffer pool implementation overrides the normal opaque parameter but
also saves it so it is accessible).
v2: add assertion check before dereferencing the BufferPoolEntry.
Signed-off-by: Marton Balint <cus@passwd.hu>
!(c->pix_fmt != AV_PIX_FMT_NONE || c->got_format_from_params)
equals
(c->pix_fmt == AV_PIX_FMT_NONE) && !c->got_format_from_params
1. When (c->pix_fmt == AV_PIX_FMT_NONE) is true, got_format_from_params is
always false, the flag doesn't contribute to the result.
2. When the first part is false, the second part doesn't matter, the flag
doesn't contribute to the result.
The result only depends on c->pix_fmt.
The MPEG-TS muxer will mux streams with unsupported codec id
as a private data stream; this usually makes the stream
not recognizable by ffmpeg and likely other tools.
When a parameter like e.g. language is contained more than once in the
part of var_stream_map pertaining to a single VariantStream, the later
one just overwrites the pointer to the earlier one, leading to a
memleak. This commit changes this by handling the situation gracefully:
The earlier string is silently freed first, so that the last one wins.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Steven Liu <lq@onvideo.cn>
"If an error is detected, localtime_r() shall return a null pointer
and set errno to indicate the error." Yet in case this happened in
hls_init(), AVERROR(ENOMEM) has been returned.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Steven Liu <lq@onvideo.cn>
The hls muxer allocates an array of VariantStreams, a structure that
contains pointers to objects that need to be freed on their own. This
means that the number of allocated VariantStreams needs to be correct
when they are freed; yet the number of VariantStreams is set in
update_variant_stream_info() resp. parse_variant_stream_mapstring()
before the allocation has been checked for success, so that upon error
an attempt would be made to free the objects whose pointers are
positioned at position NULL (the location of VariantStreams) +
offsetof(VariantStream, the corresponding pointer).
Furthermore d1fe1344 added another possibility for the first function
to leave an inconsistent state behind: If an allocation of one of the
objects referenced by the VariantStream fails, the VariantStream will be
freed, but the number of allocated VariantStreams isn't reset, leading
to the same problem as above. (This was done in the mistaken belief that
the VariantStreams array would leak otherwise.)
Essentially the same also happens for the number of cc-streams. It has
been fixed, too.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Steven Liu <lq@onvideo.cn>
Under certain circumstances hls_write_packet() would add options to an
AVDictionary. Said dictionary was never explicitly freed, instead it was
presumed that these options would be consumed when opening a new
IO-context. This left several possibilities for memleaks:
a) When no new IO-context would be opened at all. This is possible when
using both the flags temp_file and single_file together with a file
output.
b) When an error happens before one actually tries to open the new
IO-context.
c) When the new IO-context does not consume all options.
All three have been fixed; furthermore, the AVDictionary has been put
into a smaller scope (namely the only part of hls_write_packet() where
it is actually used).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Steven Liu <lq@onvideo.cn>
hls_mux_init() currently leaks an AVDictionary if opening a dynamic
buffer fails or if avformat_init_output fails. This has been fixed by
moving the initialization resp. the freeing of the dictionary around:
In the former case to a place after opening the dynamic buffer, in the
latter to a place before the check for initialization failure so that it
is done unconditionally.
Furthermore, the dictionary is now only copied and freed if the options
in it are actually used (namely when in SEGMENT_TYPE_FMP4 mode).
Finally, a similar situation in hls_start() has been fixed, too.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Steven Liu <lq@onvideo.cn>
hls_init() would allocate a buffer, although it is only needed in one of
two branches that follow. This commit moves the allocation to the branch
that actually needs the buffer.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Steven Liu <lq@onvideo.cn>
Before ed897633, the hls muxer would free its child AVFormatContexts
and reset the pointer to these contexts to NULL immediately afterwards;
ed897633 moved the former to later (into a separate function), but kept
the resetting, ensuring that the child context leaks.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Steven Liu <lq@onvideo.cn>
Before this fix, ffmpeg -h full | grep map get the command dump
like:
-map [-]input_file_id[:stream_specifier][,sync_file_id[:stream_s set input stream mapping
^
|
truncated
after this fix, we can get full option dump.
Found-by: vacingfang <vacingfang@tencent.com>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
This will probably also fix CID 1452559, a false positive where Coverity
claims a double-free occurs, because it thinks that av_dict_set() frees
its key and value arguments even when the AV_DICT_DONT_STRDUP_* flags
aren't used.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Marvin Scholz <epirat07@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
CPB side_data is copied when stream-copying (see init_output_stream_streamcopy()),
but it shall not be copied when the stream is decoded.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
According to the specifications, the payloadSize includes the 16-byte size of UUID.
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
ts_target_bitrate is in kbps, not bps. This commit clarifies the unit
and modifies the example to match the description.
Signed-off-by: James Zern <jzern@google.com>
In case an AVBPrint was not complete, icecast_open() would free some
buffers that have not been allocated yet instead of freeing the data of
the AVBPrint (if they have been allocated). Because this error does not
trigger a jump to the general cleanup section any more, one can moreover
remove a (now unnecessary) initialization of a pointer.
Furthermore, finalizing an AVBPrint can fail (namely when the string
inside the AVBPrint has not been allocated yet) and so this needs to be
checked.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
They have been removed altogether without a compat implementation, and are
either no-ops or return NULL.
This fixes compiler warnings about checks always evaluating to false, and leaks
of allocated mutexes.
Signed-off-by: James Almer <jamrial@gmail.com>
It is a requirement of bitstream conformance that num_cr_points is less than or equal to 10.
It is a requirement of bitstream conformance that num_cb_points is less than or equal to 10.
Signed-off-by: James Almer <jamrial@gmail.com>
The stereo_interpolate functions add h_step to the values h
BUF_SIZE times. Within the stereo_interpolate C functions, the
values h (h0-h3, h00-h13) are declared as local float variables,
but the compiler is free to keep them in a register with extra
precision.
If the accumulation is rounded to 32 bit float precision after
each step, the less significant bits of h_step end up ignored
and the sum can deviate, affecting the end result more than
the currently set EPS.
By clearing the log2(BUF_SIZE) lower bits of h_step, we make sure
that the accumulation shouldn't differ significantly, regardless
of any extra precision in the accmulating register/variable.
This fixes the aacpsdsp checkasm test when built with clang for
mingw/x86_32.
Signed-off-by: Martin Storsjö <martin@martin.st>
In these cases, we must pass the full path of the file to ffprobe
(as the current working dir on the remote system, e.g. when invoked
with "ssh remote ffprobe ..." isn't the wanted one).
The input filename passed to ffprobe is also included in the output,
which is part of the reference test data. Add a new option to
ffprobe to allow overriding what path is printed, to keep the
original relative path in the tests.
An alternative approach could be an option to allow requesting omitting
the file name from the dumped data, and updating the test references
accordingly.
Signed-off-by: Martin Storsjö <martin@martin.st>
This patch implements ff_hscale_8_to_15_neon with NEON fused multiply accumulate
and bumps the vectorization factor from 2 to 4.
The speedup is of 25% on Graviton1 A1 instances based on A-72 cpus:
$ ffmpeg -nostats -f lavfi -i testsrc2=4k:d=2 -vf bench=start,scale=1024x1024,bench=stop -f null -
before: t:0.040303 avg:0.040287 max:0.040371 min:0.039214
after: t:0.032168 avg:0.032215 max:0.033081 min:0.032146
The speedup is of 39% on Graviton2 m6g instances based on Neoverse-N1 cpus:
$ ffmpeg -nostats -f lavfi -i testsrc2=4k:d=2 -vf bench=start,scale=1024x1024,bench=stop -f null -
before: t:0.019446 avg:0.019423 max:0.019493 min:0.019181
after: t:0.014015 avg:0.014096 max:0.015018 min:0.013971
Tested with `make check` on aarch64-linux.
Signed-off-by: Sebastian Pop <spop@amazon.com>
Reviewed-by: Jean-Baptiste Kempf <jb@videolan.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
5 cabac states for cbf_cb and cbf_cr are supported according to
Table 9-4.
Add a test for 64x64 4:4:4 8bit HEVC clips with TUDepth = 4, cbf_cr > 0.
Signed-off-by: Xu Guangxin <guangxin.xu@intel.com>
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The max transform depth is 5(from 0 to 4), so we need 5 cabac states for
cbf_cb and cbf_cr.
See Table 9-4 for details.
Signed-off-by: Xu Guangxin <guangxin.xu@intel.com>
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Wavelet types with large amounts of overreading/writing like 9_7 would
write into the padding at high wavelet depths, which would remain and be
read by the next frame's transform and quickly cause artifacts to appear
for subsequent frames.
This fix affects all frames encoded with a non-power-of-two width, with
the artifacts varying between non-observable to very noticeable,
depending on encoder settings, so reencoding is advisable.
When testing on a memory limited system, these tests consume a
significant amount of memory and can often fail if testing by running
multiple processes in parallel.
Signed-off-by: Martin Storsjö <martin@martin.st>
Currently, assigning new buffer for pkt when multiple buffers were returned
from vaMapBuffer will overwrite the previous encoded pkt data and lead
to encode issues.
Iterate through the buf_list first to find out the total buffer size
needed for the pkt, allocate the whole pkt to avoid repeated reallocation
and memcpy, then copy data from each buf to pkt.
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
It performs HDR(High Dynamic Range) to SDR(Standard Dynamic Range) conversion
with tone-mapping. It only supports HDR10 as input temporarily.
An example command to use this filter with vaapi codecs:
FFMPEG -hwaccel vaapi -vaapi_device /dev/dri/renderD128 -hwaccel_output_format vaapi \
-i INPUT -vf 'tonemap_vaapi=format=p010' -c:v hevc_vaapi -profile 2 OUTPUT
Signed-off-by: Xinpeng Sun <xinpeng.sun@intel.com>
Signed-off-by: Zachary Zhou <zachary.zhou@intel.com>
Signed-off-by: Ruiling Song <ruiling.song@intel.com>
payload_count is used to track the number of SEI payloads. It is also
used to free the SEIs in cbs_h264_free_sei()/cbs_h265_free_sei().
Currently, payload_count is set after for loop is completed. Hence if
there is an error and the function exits, the payload remains zero
causing a memleak.
This commit keeps track of payload_count inside the for loop to fix the
issue. Note that that the contents of current are initialized with
av_mallocz() so there is no need to zero initialize payload_count.
Found-by: libFuzzer
Reviewed-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
There can be at most 31 SPS and 255 PPS in the mp4/Matroska extradata.
Given that each has a size of at most 2^16-1, the length of the output
derived from these parameter sets can never overflow an ordinary 32 bit
integer. So use a simple uint32_t instead of uint64_t and replace the
unnecessary check with an av_assert1.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -2147483648 - 13 cannot be represented in type 'int'
Fixes: 18893/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ADPCM_IMA_APC_fuzzer-5630760442920960
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 2038337026 + 109343477 cannot be represented in type 'int'
Fixes: 18886/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMALOSSLESS_fuzzer-5673660505653248
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 1677721600 * 32 cannot be represented in type 'int'
Fixes: 18885/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_FFWAVESYNTH_fuzzer-5741242185154560
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The documentation still mentions numerical constants in addition to textual
ones. It is also wrong to use distinct modes as flags and it disallows us to
actually use the flags field for real flags in the future.
Signed-off-by: Marton Balint <cus@passwd.hu>
"It is a requirement of bitstream conformance that num_y_points is less than or equal to 14."
Fixes: index 24 out of bounds for type 'uint8_t [24]'
Fixes: 19282/clusterfuzz-testcase-minimized-ffmpeg_BSF_AV1_FRAME_MERGE_fuzzer-5747424845103104
Note, also needs a23dd33606
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: jamrial
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The IVF muxer autoinserts the av1_metadata filter unconditionally, which is
not desirable for these tests.
Signed-off-by: James Almer <jamrial@gmail.com>
These functions aren't available when building for the restricted
UWP/WinRT/WinStore API subsets.
Normally when building in this mode, one is probably only building
the libraries, but being able to build ffmpeg.exe still is useful
(and a ffmpeg.exe targeting these API subsets still can be run
e.g. in wine, for testing).
Signed-off-by: Martin Storsjö <martin@martin.st>
As the values generated by av_bmg_get can be arbitrarily large
(only the stddev is specified), we can't use a fixed tolerance.
Calculate a dynamic tolerance (like in float_dsp from 38f966b222),
based on the individual steps of the calculation.
This fixes running this test with certain seeds, when built with
clang for mingw/x86_32.
Signed-off-by: Martin Storsjö <martin@martin.st>
These functions already free it themselves before they allocate the new
extradata.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
It is not uncommon to find code where the caller thinks to know better
what the return value should be than the callee. E.g. something like
"if (av_new_packet(pkt, size) < 0) return AVERROR(ENOMEM);". This commit
changes several instances of this to instead forward the actual error.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The extradata will be freed automatically when the corresponding stream
gets freed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
"HEVC HDR UHDTV Bitstreams using HLG10 shall also contain the
alternative_transfer_characteristics SEI message. The
alternative_transfer_characteristics SEI message shall be inserted on
the HEVC DVB_RAP, and preferred_transfer_characteristics shall be set
equal to "18", indicating Recommendation ITU-R BT. 2100 [45] HLG
system."
Signed-off-by: Zhong Li <zhongli_dev@126.com>
Add -tile_rows and -tile_cols option to specify the number of tile rows
and columns for ICL+ (gen 11) platform.
A tile must wholly contain all the slices within it. Slices cannot cross
tile boundaries. So the slice number would be implicitly resized to the
max(nSlice, nTile).
Example:
ffmpeg -v verbose -hwaccel qsv -init_hw_device qsv=hw
-filter_hw_device hw -f rawvideo -s:v 1920x1080 -i ./input.nv12 -vf
format=nv12,hwupload=extra_hw_frames=64 -c:v hevc_qsv -tile_rows 2
-tile_cols 2 -slices 4 -y output.h265
Also dump the actual quantity of encoded tiled rows and columns in run
time.
Fix the enhancement #8400.
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Implemented according to the specification at https://www.iso.org/standard/69561.html
The 'mhm1' sample entry is registered with MP4RA, which is defined as MHAS encapsulated single stream MPEG-H 3D Audio.
'MHAS' stands for MPEG-H audio stream, which contains encoded audio data and corresponds metadata for decoding.
This patch enables extracting the MHAS bitstream from MP4 and remuxing into MP4.
Signed-off-by: James Almer <jamrial@gmail.com>
Introduce AV_CODEC_PROP_INTRA_ONLY flag to audio codec as well as video codec to support non intra-only audio codec.
Signed-off-by: Yuki Tsuchiya <Yuki.Tsuchiya@sony.com>
Signed-off-by: James Almer <jamrial@gmail.com>
This particular function is only required to return nonzero on
errors, but use the common AVERROR() pattern for consistency.
Signed-off-by: Martin Storsjö <martin@martin.st>
These dependencies are evaluted by make and must be expressed with
the paths as in the local filesystem.
Signed-off-by: Martin Storsjö <martin@martin.st>
Taking into account the code
fb(2, ar_coeff_lag);
num_pos_luma = 2 * current->ar_coeff_lag * (current->ar_coeff_lag + 1);
if (current->num_y_points)
num_pos_chroma = num_pos_luma + 1;
else
num_pos_chroma = num_pos_luma;
Max value for ar_coeff_lag is 3 (two bits), for num_pos_luma 24, and for
num_pos_chroma 25.
Both ar_coeffs_cb_plus_128 and ar_coeffs_cr_plus_128 may have up to
num_pos_chroma values.
Reviewed-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The tremolo filter uses floating point internally, and uses
multiplication factors derived from sin(fmod()), neither of
which is bitexact for use with framecrc.
This fixes running this test when built with for mingw/x86_32
with clang.
In this case, a 1 ulp difference in the output from fmod() would
end up in an output from the filter that differs by 1 ulp, but
which makes the lrint() in swresample/audioconvert.c round in a
different direction.
Signed-off-by: Martin Storsjö <martin@martin.st>
As the values generated by av_bmg_get can be arbitrarily large
(only the stddev is specified), we can't use a fixed tolerance.
This matches what was done for test_vector_dmul_scalar in
38f966b222.
This fixes the float_dsp checkasm test for some seeds, when built
with clang for mingw/x86_32.
Signed-off-by: Martin Storsjö <martin@martin.st>
rename enable_intraintra_compound to enable_interintra_compound,
which keep same as AV1 sepc(v1.0.0-errata1).
Signed-off-by: Fei Wang <fei.w.wang@intel.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Besides improved readability it also zeroes the padding which has been
forgotten here.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The packet a demuxer receives is freshly initialized, hence it is
unnecessary to reset any flags on them (as none are set), yet apc did
this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
liuqideMacBook-Pro:build liuqi$ ffmpeg --help full | grep cache
cannot find cache protocol options.
after patch:
bogon:dash liuqi$ ./ffmpeg --help full | grep cache
cache AVOptions:
can find the cache AVOptions after patch.
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
fix when pkg-config fail and openssl > 1.1.0 --enable-openssl fail,
the root cause is check_lib can't found the SSL_library_init().
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: macweng <macweng@tencent.com>
add linger parameter to libsrt, it's setting the number of seconds
that the socket waits for unsent data when closing.
Reviewed-by: Andriy Gelman <andriy.gelman@gmail.com>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
Fixes: signed integer overflow: -9223372036854775808 - 17 cannot be represented in type 'long'
Fixes: 18768/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5674385247830016
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 2119056926 - -134217728 cannot be represented in type 'int'
Fixes: 18728/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-5747539563511808
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 1778647621 + 574372924 cannot be represented in type 'int'
Fixes: 18692/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TRUEMOTION2_fuzzer-6248679635943424
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 2147483188 + 2048 cannot be represented in type 'int'
Fixes: 18741/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TRUESPEECH_fuzzer-5748950460268544
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
ff_id3v2_parse_priv_dict() uses av_dict_set() with the flags
AV_DICT_DONT_STRDUP_KEY and AV_DICT_DONT_STRDUP_VAL. In this case both
key and value are freed on error (and owned by the destination
dictionary on success), so that freeing them again on error is a
double-free and therefore forbidden. But it nevertheless happened.
Fixes CID 1452489 and 1452421.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
contained in Vorbis comments in the CodecPrivate of flac tracks.
Moreover, it also tests header removal compression.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Adjustment of evaluated values shifted to ff_adjust_scale_dimensions
Shifted code for force_original_aspect_ratio and force_divisble_by from
vf_scale so it is now available for scale_cuda, scale_npp and
scale_vaapi as well.
This test contains a track with zlib compressed CodecPrivate in addition
to compressed frames; the former was unchecked before.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
ProRes in Matroska is supposed to not contain the first atom header
(containing a size field and the tag "icpf") and therefore the Matroska
demuxer has to recreate it; this involves an allocation and copy, of
course. Whether the old buffer (containing the data without the atom
header) needs to be freed or not depends upon whether it is what was
directly read (in which case it is owned by an AVBuffer) or whether it
has been allocated when reversing the track's content compression (e.g.
zlib compression) that Matroska supports.
So there are three pointers involved: The one pointing to the directly
read data (owned by the AVBuffer), the one pointing to the currently
valid data (which coincides with the former if no content compression
needed to be reverted) and the one pointing to the new data with the
first atom header. The check for whether to free the second of these is
simply whether the first two are different.
This works mostly, but there is a complication: Some muxers don't strip
the first atom header away and in this case, it is also not reinserted
and no new buffer is allocated; instead, the second and the third
pointers agree. In this case, one must never free the second buffer.
Yet it is currently done if the track is e.g. zlib compressed.
This commit fixes this.
This is a regression since b8e75a2a.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
In the worst case the startcode prefix has 4 bytes.
This fixes a trigerred assertion:
Assertion dp <= max_size failed at libavcodec/cbs_h2645.c:1451
Found-by:libFuzzer
Reviewed-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
Width and height expressions can refer to each other. Width is
evaluated twice to allow for reference to output height. So we
should not error out upon failure of first evaluation of width.
Fixes: fate-fitsdec-bitpix-64
Possibly Fixes: -nan is outside the range of representable values of type 'unsigned short'
Possibly Fixes: 17769/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_FITS_fuzzer-5678314672357376
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If an error happens in vobsub_read_header() after allocating the
AVFormatContext intended to read the sub-file, both the AVFormatContext
as well as the data in the subtitles queues leaks. This has been fixed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
vobsub_read_header() uses an AVBPrint to write a string and up until
now, it collected the string stored in the AVBPrint via
av_bprint_finalize(), which might involve an allocation and copy of the
string. But this is unnecessary, as the lifetime of the returned string
does not exceed the lifetime of the AVBPrint. So use the string in the
AVBPrint directly.
This also makes it possible to easily fix a memleak: In certain error
situations, the string stored in the AVBPrint would not be freed (if it
was dynamically allocated). This has been fixed, too.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When the VobSub demuxer was added, the fields it required were simply
added to the MpegDemuxContext (if the VobSub demuxer was selected at
all). The mpeg demuxer of course doesn't use these fields even if they
are there; and the VobSub demuxer doesn't use the old ones: It opens an
mpeg subdemuxer of its own and uses this where a mpeg demuxer is
required. Hence the two contexts can be split, saving memory.
Furthermore several headers can now be moved to the section that is
guarded by #if CONFIG_VOBSUB_DEMUXER (this even includes avassert.h
which was unguarded and has been added in 9cde9f70 despite not being
used in that patch).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When parsing EBML lacing, for every number read, a new AVIOContext has
been initialized (via ffio_init_context()) just for this number. This
has been changed: The context is kept now.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
When parsing the sizes of the frames in a lace fails, sometimes no
error message was raised (e.g. when using xiph or fixed-size lacing).
Only EBML lacing generated error messages (which were wrongly declared
as AV_LOG_INFO), but even here not all errors resulted in an error
message. So add a generic error message to catch them all.
Moreover, if parsing one of the EBML numbers fails, ebml_read_num already
emits its own error messages, so that all that is needed is a generic error
message to indicate that this happened during parsing the sizes of the
frames in a block; in other words, the error messages specific to
parsing EBML lace numbers can be and have been removed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
870e7552 introduced validating the lace sizes when they are parsed and
removed the old check; yet when merging this libav commit in 6902c3ac,
the old check for whether the frame extends beyond the frame has been kept.
It is unnecessary and has been removed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Up until now, when an error happened in one of the inner loops in
matroska_parse_laces, a variable designated for the return value has
been set to an error value and break has been used to exit the
current loop/case. This was done so that the end of matroska_parse_laces
is reached, because said function allocated memory which is later used
and freed in the calling function and passed at the end of
matroska_parse_laces.
But given that there is no allocation any more, one can now return
immediately. And this commit does this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The maximal number of frames in a lace can be 256; hence one has a not
excessive upper bound on the size of an array that can hold the sizes of
all the frames in a lace. Yet up until now, said array has been
dynamically allocated. This has been changed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Up until c4e0e314, the seek table has been included in the tta
extradata, so that the size of said extradata was 22 (the size of a TTA1
header) + 4 * number of frames. The decoder rejected anything below a
size of 30 and so the Matroska demuxer exported 30 byte long extradata,
of which only 18 were set (it ignores a CRC-32 and simply leaves it at
0). But this is unnecessary since said commit, so reduce the size to 22.
Furthermore, replace 30 by 22 in a comment about the extradata size in
libavcodec/tta.c.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
That way one doesn't have to free later. In this case (concerning TTA
extradata), this also fixes a memleak when the output samplerate is
invalid.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
This sets the range of the first automatically assigned PMT PID or elementary
stream PID parameters to [0x20, 0x1ffa]. You can still assign manually a PID
for a stream using AVStream->id in the wider [0x10, 0x1ffe] range as specified
by ISO13818-1. But since DVB and ATSC both reserves some PIDs, let's not allow
them to be automatically assigned.
Also make sure that assigned PID numbers are valid and fix the error message
for the previous PID collision checks.
Signed-off-by: Marton Balint <cus@passwd.hu>
The unsharp filter uses an array of arrays of uint32_t, each of which is
separately allocated. These arrays also need to freed separately; but
before doing so, one needs to check whether the array of arrays has
actually been allocated, otherwise one would dereference a NULL pointer.
This fixes#8408.
Furthermore, the array of arrays needs to be zero-initialized so that
no uninitialized pointer will be freed in case an allocation of one of
the individual arrays fails.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This optimizes the code slightly (116 -> 80sec)
Testcase: 18668/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ALS_fuzzer-5710836719157248
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
buf_size is not updated as buf is advanced so it is wrong after the first
iteration
Fixes: Timeout (160sec -> 27sec)
Fixes: 18658/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_G729_fuzzer-5729784269373440
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The 0 case was added with the support for multiple packets. It
appears unintended and causes extra complexity and out of array
accesses (though within padding)
No testcase
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This combination would assume different block sizes throughout the code so its
better to error out.
Fixes: signed integer overflow: -1082385168 * 2 cannot be represented in type 'int'
Fixes: 19110/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ACELP_KELVIN_fuzzer-5643993950191616
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -2147483648 * -1 cannot be represented in type 'int'
Fixes: 18643/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ALAC_fuzzer-5672182449700864
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 50176 * 262144 cannot be represented in type 'int'
Fixes: 18629/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_VC1IMAGE_fuzzer-5182370286403584
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 4 * 538976288 cannot be represented in type 'int'
Fixes: 18622/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_VMDAUDIO_fuzzer-5092166174507008
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
ff_flac_parse_picture() parses a buffer containing a flac metadata
picture block by wrapping it in an AVIOContext and using the AVIOContext
API. Consequently, when not enough data could be read AVERROR(EIO) was
returned although reading didn't really fail: A block that contains a
subfield whose size field indicates that it is so big as to extend
beyond the buffer is just invalid.
This commit changes this by using the bytestream2 API instead;
furthermore, the checks for whether there is enough data left are
performed before allocating a buffer for said data.
Finally, if the length of the picture description is bigger than
INT_MAX, it will now raise an error.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
During parsing a flac picture metadata block, the mimetype is read as
follows: Its 32b size field is read and checked for being in the range
1..63; afterwards, the actual mimetype-string is read into a buffer of
size 64, where the length to read is the minimum of the length field and
the size of the destination buffer -1. Then an assert guards that length
is indeed < the size of the destination buffer before the string in the
buffer is zero-terminated.
The FFMIN as well as the assert are actually redundant, as it has
been checked that the string (even after terminating) fits into the
buffer. In order to make this clear, reword the check "len >= 64" to
"len >= sizeof(mimetype)" and drop the FFMIN as well as the assert.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Put an AVIOContext whose lifetime doesn't extend beyond the function where
it is allocated on the stack instead of allocating and freeing it. This
also avoids the need to free it, which in this case fixes possible
memleaks on error.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
By using avio_get_dyn_buf() + ffio_free_dyn_buf() instead of
avio_close_dyn_buf() + av_free() one can avoid an allocation + copy for
small dynamic buffers.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
By using avio_get_dyn_buf() + ffio_free_dyn_buf() instead of
avio_close_dyn_buf() + av_free() one can avoid an allocation + copy for
small dynamic buffers (i.e. small master elements).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
By using avio_get_dyn_buf() + ffio_free_dyn_buf() instead of
avio_close_dyn_buf() + av_free() one can avoid an allocation + copy for
small tags. Furthermore, it simplifies freeing.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
By using avio_get_dyn_buf() + ffio_free_dyn_buf() instead of
avio_close_dyn_buf() + av_free() one can avoid an allocation + copy for
small extradata. Furthermore, it simplifies freeing.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
By using avio_get_dyn_buf() + ffio_free_dyn_buf() instead of
avio_close_dyn_buf() + av_free() one can avoid an allocation + copy for
small headers. Furthermore, it simplifies freeing.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
By using avio_get_dyn_buf() + ffio_free_dyn_buf() instead of
avio_close_dyn_buf() + av_free() one can avoid an allocation + copy for
small tags. Furthermore, it simplifies freeing.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until now, using a dynamic buffer entailed at least three
allocations: One for the AVIOContext, one for the AVIOContext's opaque
(which, among other things, contains the small write buffer), and one
for the big buffer that is independently allocated that is returned when
calling avio_close_dyn_buf().
It is possible to avoid the third allocation if one doesn't use a
packetized dynamic buffer, if all the data written so far fit into the
write buffer and if one does not require the actual (big) buffer to have
an indefinite lifetime. This is done by making avio_get_dyn_buf() return
a pointer to the data in the write buffer if nothing has been written to
the main buffer yet. The dynamic buffer will then be freed using
ffio_free_dynamic_buffer (which needed to be modified not to call
avio_close_dyn_buf() internally).
So a typical use-case like:
size = avio_close_dyn_buf(dyn_pb, &buf);
do something with buf
av_free(buf);
can be converted to:
size = avio_get_dyn_buf(dyn_pb, &buf);
do something with buf
ffio_free_dynamic_buffer(&dyn_pb);
In more complex scenarios this can simplify freeing as well, because it
is now clear that freeing always has to be performed via
ffio_free_dynamic_buffer().
Of course, in case this saves an allocation it also saves a memcpy.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Disable by default to output all the layers, to match libaomdec wrapper.
Add option to select the operating point for the spatial layers.
Update the documentation with the new options.
Signed-off-by: James Almer <jamrial@gmail.com>
The decoder hardcodes that audio is stream_id = 1 so it does not
currently work with more or less than 1 video stream at st=0
Fixes: assertion failure
Fixes: 18602/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-6259277199310848
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 6175076100092079360 - -5034989061050195840 cannot be represented in type 'long'
Fixes: 18614/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_FFWAVESYNTH_fuzzer-5704508847423488
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 1145975808 - -1146173210 cannot be represented in type 'int'
Fixes: 18616/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_RALF_fuzzer-5121296757424128
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The G729 reference decoder clips after each individual operation and keeps track if overflow
occurred (in the fixed point implementation), this here is
simpler and faster but not 1:1 the same what the reference does.
Non fuzzed samples which trigger any such overflow are welcome, so
the need and impact of different clipping solutions can be evaluated.
Fixes: signed integer overflow: 1271483721 + 1073676289 cannot be represented in type 'int'
Fixes: 18617/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ACELP_KELVIN_fuzzer-5137705679978496
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 2147483647 + 1 cannot be represented in type 'int'
Fixes: 18333/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_COMFORTNOISE_fuzzer-5668481831272448
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
As is the decoder will never stop, it will cause an infinite loop. The RFC seems only
to speak of non empty packets so endlessly generating noise from the last empty flush
packets seems wrong.
Fixes: infinite loop
Fixes: 18333/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_COMFORTNOISE_fuzzer-5668481831272448
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The threshold is chosen so that the worse frames would together not take
excessive time.
A better solution is welcome!
Fixes: Timeout (308sec ->102ms)
Fixes: 18314/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_VP9_fuzzer-5701689176227840
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This comment does not account for the fact that the limits on cluster
size and duration are configurable by the user since 98308bd4.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If mkv_write_trailer() is not called, the cached audio packet might
leak; so unref it in mkv_deinit().
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
c0c7946196 unintentianally changed the initialization flow of the
decoder: It caused the capture buffers to be initialized on
v4l2_m2m.c:180 in v4l2_configure_contexts(). This breaks h264 decoding
on the Odroid XU4 (RPI4 was not affected).
This commit postpones capture buffer initialization
as before c0c7946196 to fix the issue.
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
Signed-off-by: Aman Gupta <aman@tmm1.net>
Fixes: index 9 out of bounds for type 'const uint64_t [8][256]'
Fixes: 18409/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_IFF_ILBM_fuzzer-5767030560522240
Fixes: 18720/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_IFF_ILBM_fuzzer-5651995784642560
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The largest documented frame size = block align is 1024 bytes
(https://wiki.multimedia.cx/index.php/ATRAC3)
Without a limit this can allocate arbitrary memory and trigger OOM
Fixes: OOM
Fixes: 18337/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ATRAC3_fuzzer-5763861478637568
Fixes: 18556/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ATRAC3AL_fuzzer-5646183334936576
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The intermediates are required to fit in 12bit (8.1.3.9 Coefficient Scaling)
See SMPTE 421M-2006 and Amendment 1-2007
Fixes: signed integer overflow: -20691 * 262144 cannot be represented in type 'int'
Fixes: 18479/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_VC1_fuzzer-5128912371187712
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 7400 + 2147482786 cannot be represented in type 'int'
Fixes: 18405/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-5708834760294400
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 5 * -1094995529 cannot be represented in type 'int'
Fixes: 18346/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SONIC_fuzzer-5709623893426176
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Clipping is done as it was preferred in review
See: [FFmpeg-devel] [PATCH 1/5] avcodec/atrac9dec: Check precision_fine/coarse
Fixes: out of array access
Fixes: 18330/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ATRAC9_fuzzer-5641113058148352
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This ensures they will be reference counted, as required by the AVCodec.receive_packet()
API.
Should fix ticket #8386.
Signed-off-by: James Almer <jamrial@gmail.com>
In scearios where a Temporal Unit is written right after reading it using the same
CBS context (av1_metadata, av1_frame_merge, etc), the reference frame state used
by the writer must not be the state that's the result of the reader having already
parsed the current frame in question.
This fixes writing Switch frames, and frames using short ref signaling.
Signed-off-by: James Almer <jamrial@gmail.com>
The number of bits in a PutBitContext must fit into an int, yet nothing
guaranteed the size argument cbs_write_unit_data() uses in init_put_bits()
to be in the range 0..INT_MAX / 8. This has been changed.
Furthermore, the check 8 * data_size > data_bit_start that there is
data beyond the initial padding when writing mpeg2 or H.264/5 slices
could also overflow, so divide it by 8 to get an equivalent check
without this problem.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
All cbs-functions to write units share a common pattern:
1. They check whether they have a write buffer (that is used to store
the unit's data until the needed size becomes known after writing the
unit when a dedicated buffer will be allocated).
2. They use this buffer for a PutBitContext.
3. The (codec-specific) writing takes place through the PutBitContext.
4. The return value is checked. AVERROR(ENOSPC) here always indicates
that the buffer was too small and leads to a reallocation of said
buffer.
5. The final buffer will be allocated and the data copied.
This commit factors this common code out in a single function in cbs.c.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
flac_read_timestamp() applied av_init_packet() to a packet (which
initializes all fields of the packet except for data and size) and then
went on to use only the data and size fields. In other words: Said
packet can be removed and replaced by an uint8_t * and an int.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
1)Some filters allow cross-referenced expressions e.g. x=y+10. In
such cases, filters evaluate expressions multiple times for
successful evaluation of all expressions. If the expression for one or
more variables contains a RNG, the result may vary across evaluation
leading to inconsistent values across the cross-referenced expressions.
2)A related case is circular expressions e.g. x=y+10 and y=x+10 which
cannot be succesfully resolved.
3)Certain filter variables may only be applicable in specific eval modes
and lead to a failure of evaluation in other modes e.g. pts is only
relevant for frame eval mode.
At present, there is no reliable means to identify these occurrences and
thus the error messages provided are broad or inaccurate. The helper
function introduced - av_expr_count_vars - allows developers to identify
the use and count of variables in expressions and thus tailor the error
message, allow for a graceful fallback and/or decide evaluation order.
The earlier requirement was for the new buffer to be bigger than the old
one. This has been relaxed to only demand that the new buffer can hold
all the data written so far. This is in preparation for further commits.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Encoders must return reference counted packets.
This was checked only for encoders using the AVCodec->encode2() API, while
blindly accepting whatever encoders using the AVCodec->receive_packet() API
were returning.
Signed-off-by: James Almer <jamrial@gmail.com>
The current approach has two different calls to av_bsf_send_packet():
A normal one, sending a packet; and an extraordinary one just for
flushing. These can be unified into one by making use of the newly
documented fact that av_bsf_send_packet() allows to signal flushing via
empty packets (i.e. packets without data and side-data).
This also fixes CID 1455685 which resulted from the fact that the call
for flushing was not checked given that it couldn't fail.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Explicitly allowing empty packets to signal flushing helps getting rid
of special cases. It does not hinder the ability to send i.e.
timing-only packets, because one can send packets with zero size and
pkt->data set.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
This BSF takes Temporal Units split across different AVPackets and merges them
by looking for Temporal Delimiter OBUs.
Signed-off-by: James Almer <jamrial@gmail.com>
This will be needed by the next commit, where packets with a single Temporal
Delimiter OBU with no size will need to be parsed
Signed-off-by: James Almer <jamrial@gmail.com>
remove_side_data is supposed to remove a single instance by design.
Since new_side_data() doesn't forbid add multiple instances of the
same type, remove_side_data should deal with that.
Signed-off-by: Marton Balint <cus@passwd.hu>
mp3 header bitstream syntax:
header()
{
syncword 12bits bslsf
id 1bit bslsf
layer 2bit bslsf
protection_bit 1bit bslsf
bitrate_index 4bits bslsf
sampling_frequency 2bits bslsf
padding_bit 1bit bslsf
private_bit 1bit bslsf
mode 2bits bslsf
mode_extension 2bits bslsf
copyright 1bit bslsf
original/home 1bit bslsf
emphasis 2bits bslsf
}
if the header is masking with MP3_MASK(0xFFFE0CCF), below fields will be cleared:
protection_bit, bitrate_index, sampling_freqency, mode
with SAME_HEADER_MASK(0xFFFE0C00), extra below fields will be cleared which didn't make
sense:
mode_extension, copyright, original/home, emphasis
As the MP3_MASK is good for same mp3 header masking and is defined in the
header, so it's preferable to remove SAME_HEADER_MASK to keep the masking same.
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
004ebd4b added a function with a parameter that was declared as restrict
and not av_restrict. This is not supported by MSVC as several FATE-boxes
that now fail to build show. So use av_restrict.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Tested-by: Matthias Troffaes <matthias.troffaes@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -2147450880 - 65535 cannot be represented in type 'int'
Fixes: 18393/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_INTERPLAY_ACM_fuzzer-5667520110919680
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 1494495519 + 1494495519 cannot be represented in type 'int'
Fixes: 18347/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SBC_fuzzer-5711714661695488
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 538976288 * 8 cannot be represented in type 'int'
Fixes: 18348/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_METASOUND_fuzzer-6681325716635648
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 2117181180 + 60483298 cannot be represented in type 'int'
Fixes: 18344/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-5685327791915008
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
ISO/IEC 14496-3:2005(E): "Each entry of P_one[ ][ ] is in the range of 1 to
128, corresponding to a probability of 1/256 to 128/256 of the next error bit (bit E, See Figure 10.5)..."
Fixes: Timeout (42sec ->1sec)
Fixes: 18181/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DST_fuzzer-5736646250594304
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This avoids unneeded operations and makes the code faster.
Fixes: Timeout
Fixes: 15724/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_PAF_VIDEO_fuzzer-5750842205929472 (12sec -> 9sec)
Fixes: 17625/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_PAF_VIDEO_fuzzer-5640515311108096 (16sec -> 4sec)
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
ff_format_io_close will check the AVIOContext pointer pb, so drop
the unnecessary check before ff_format_io_close.
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
Enable probesize/max_analyze_duration setting when open the sub-demuxer,
it's will be used to minimizing the initial delay.
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
Add probesize/max_analyze_duration support when open the sub-demuxer,
it's will be used to minimizing the initial delay.
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
This filter accepts all the dnn networks which do image processing.
Currently, frame with formats rgb24 and bgr24 are supported. Other
formats such as gray and YUV will be supported next. The dnn network
can accept data in float32 or uint8 format. And the dnn network can
change frame size.
The following is a python script to halve the value of the first
channel of the pixel. It demos how to setup and execute dnn model
with python+tensorflow. It also generates .pb file which will be
used by ffmpeg.
import tensorflow as tf
import numpy as np
import imageio
in_img = imageio.imread('in.bmp')
in_img = in_img.astype(np.float32)/255.0
in_data = in_img[np.newaxis, :]
filter_data = np.array([0.5, 0, 0, 0, 1., 0, 0, 0, 1.]).reshape(1,1,3,3).astype(np.float32)
filter = tf.Variable(filter_data)
x = tf.placeholder(tf.float32, shape=[1, None, None, 3], name='dnn_in')
y = tf.nn.conv2d(x, filter, strides=[1, 1, 1, 1], padding='VALID', name='dnn_out')
sess=tf.Session()
sess.run(tf.global_variables_initializer())
output = sess.run(y, feed_dict={x: in_data})
graph_def = tf.graph_util.convert_variables_to_constants(sess, sess.graph_def, ['dnn_out'])
tf.train.write_graph(graph_def, '.', 'halve_first_channel.pb', as_text=False)
output = output * 255.0
output = output.astype(np.uint8)
imageio.imsave("out.bmp", np.squeeze(output))
To do the same thing with ffmpeg:
- generate halve_first_channel.pb with the above script
- generate halve_first_channel.model with tools/python/convert.py
- try with following commands
./ffmpeg -i input.jpg -vf dnn_processing=model=halve_first_channel.model:input=dnn_in:output=dnn_out:fmt=rgb24:dnn_backend=native -y out.native.png
./ffmpeg -i input.jpg -vf dnn_processing=model=halve_first_channel.pb:input=dnn_in:output=dnn_out:fmt=rgb24:dnn_backend=tensorflow -y out.tf.png
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
Before this commit an av_assert0 would fail if a v4l2 device did not
support a target format.
For example,
./ffmpeg -f v4l2 -codec:v h264 -i /dev/video0 -f mpegts -
would signal an abort if /dev/video0 did not support h264.
The new behaviour is to return an AVERROR(EINVAL) error code. An
av_assert0 has been added to verify this return.
Fixes#6629
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If the alpha decoder init failed we presented the error message from the normal
decoder. This change should prevent such mistakes.
Signed-off-by: Marton Balint <cus@passwd.hu>
Signed-off-by: James Zern <jzern@google.com>
This compared to the other suggestions is cleaner and easier to understand
keeping the condition in the if() simple.
This affects alot of fate tests.
See: [FFmpeg-devel] [PATCH 05/11] avformat/nutenc: Don't pass NULL to memcmp
See: [FFmpeg-devel] [PATCH]lavf/nutenc: Do not call memcmp() with NULL argument
Fixes: Ticket 7980
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This introduces two new AVOption options for the FTP protocol,
one named ftp-user to supply the username to be used for auth,
one named ftp-password to supply the password to be used for auth.
These are useful for when an API user does not wish to deal with
URL manipulation and percent encoding.
Setting them while also having credentials in the URL will use the
credentials from the URL. The rationale for this is that credentials
embedded in the URL are probably more specific to what the user is
trying to do than anything set by some API user.
Signed-off-by: Nicolas Frattaroli <ffmpeg@fratti.ch>
Signed-off-by: Marton Balint <cus@passwd.hu>
Allows user to set maximum number of buffered packets when
probing a codec. It was a hard-coded parameter before this commit.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
1. must enable low_power mode since just VDENC can be supported by iHD
driver right now
2. Coding option1 and extra_data are not supported by MSDK
3. IVF header will be inserted in MSDK by default, but it is not needed
for FFmpeg, so disable it.
Signed-off-by: Zhong Li <zhongli_dev@126.com>
MFXVideoENCODE_Query calls CheckVideoParamQueryLike in MSDK and
will determine whether to set param.mfx.TargetKbps to the allowed
minTargetKbps according to the bitrate_limit in extco2 buffer.
Thus q->param.ExtParam must be set before MFXVideoENCODE_Query in case
minTargetKbps is written to TargetKbps by default.
1080P AVC encoding with option "-bitrate_limit 0 -b:v 100k":
Before patch:
902 kbps
After patch:
156 kbps
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
This part of the code counts the number of planes returned by the v4l2
device for each queried capture/output buffer.
When testing the GPU h264 encoder on Nvidia's Jetson Nano, this caused an
infinite loop because avbuf->buf.length included some empty buffers (i.e.
where avbuf->buf.m.planes[i].length = 0), meaning that the counter was
never incremented and break was never reached.
This is fixed in the commit by using a well defined iteration range.
Signed-off-by: Aman Gupta <aman@tmm1.net>
G2M allows large images from small input and also reallocates
multiple buffers on any resolution change.
Fixes: Timeout (22sec -> 5sec)
Fixes: 18022/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_G2M_fuzzer-5089192530411520
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 2 * 1881153568 cannot be represented in type 'int'
Fixes: 17996/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AAC_FIXED_fuzzer-5687126468853760
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 1077952576 + 1355863565 cannot be represented in type 'int'
Fixes: 16196/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TRUEMOTION2_fuzzer-5679842317565952
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 538976288 - -2080374792 cannot be represented in type 'int'
Fixes: 16196/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TRUEMOTION2_fuzzer-5144044274974720
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: 18432/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMAV2_fuzzer-5675574936207360
Fixes: 18326/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMAV2_fuzzer-5071752362721280
Fixes: 18384/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMAV1_fuzzer-5769439500304384
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
A too big picture will case the muxer to write a truncated block size (uint24)
causing the output file to be corrupt.
How to reproduce:
Write a file with truncated block size:
ffmpeg -y -f lavfi -i sine -f lavfi -i color=red:size=2400x2400 -map 0:a:0 -map 1:v:0 -c:v:0 bmp -disposition:1 attached_pic -t 1 test.flac
Try to decode:
ffmpeg -i test.flac test.wav
Signed-off-by: Mattias Wadman <mattias.wadman@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
As of libtwolame 0.4.0, 384 kbps is not accepted as a valid bitrate
for encoding mono audio and the maximum bitrate is now halved to 192
kbps to comply with the MP2 standard. Example error:
twolame_init_params(): 384kbps is an invalid bitrate for mono encoding.
Adjust the default bitrate calculation to take this into account.
Signed-off-by: James Cowgill <jcowgill@debian.org>
Prevents memleaks when the trailer is never written (e.g. when there was
a write error when writing the header).
Fixes ticket #8347.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Performance of WMV3 decoding has speed up from 3.66x to 5.23x tested on 3A4000.
Reviewed-by: Shiyou Yin <yinshiyou-hf@loongson.cn>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If filter changes frame w/h, AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC
cannot be supported.
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
to support dnn networks more general, we need to know the input info
of the dnn model.
background:
The data type of dnn model's input could be float32, uint8 or fp16, etc.
And the w/h of input image could be fixed or variable.
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
so, we can make a filter more general to accept different network
models, by adding a data type convertion after getting data from network.
After we add dt field into struct DNNData, it becomes the same as
DNNInputData, so merge them with one struct: DNNData.
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
Unlike other tf.*.conv2d layers, tf.nn.conv2d does not create many
nodes (within a scope) in the graph, it just acts like other layers.
tf.nn.conv2d only creates one node in the graph, and no internal
nodes such as 'kernel' are created.
The format of native model file is also changed, a flag named
has_bias is added, so change the version number.
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
There have been many reports over the years about problems when
taking an HLS stream as input to `ffmpeg` where there are timestamp
discontinuities present. This is explicitly supported in the
HLS spec (EXT-X-DISCONTINUITY) and often used for ad injection.
Various fixes and work-arounds have been proposed over the years,
but one step that seems obvious, even if it's not a complete fix,
is to mark the HLS input format as supporting discontinuities. This
will prevent timestamp fixup logic in ffmpeg.c kicking in that ends
up mangling the timestamps unnecessarily.
I've tested this out with an example provided by Joe Koberg early
last year, and it is sufficient to allow `ffmpeg` to download and
mux the stream correctly. Joe had briefly suggested that other
situations can still be handled incorrectly, but this seems like
a strict improvement.
Joe's example:
https://s3.amazonaws.com/playon-test-videos/discont_test_new/discont_test.m3u8
Reviewed-by: Steven Liu <lq@onvideo.cn>
Reviewed-by: Dennis Mungai <dmngaie@gmail.com>
VP4 applies a loop filter during motion compensation, causing the block offset
will often by unaligned. This produces a bus error on some platforms, namely
ARMv7 NEON.
This patch adds a unaligned version of the loop filter function pointer
to VP3DSPContext.
Reported-by: Mike Melanson <mike@multimedia.cx>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Or it'll cause invalid color and s->filter is NULL.
Please reproduce it with below command on big endian system:
$ ./ffmpeg -f lavfi -i "anoisesrc=d=60:c=1:r=48000" -f s16le -c:a pcm_s16le -f
null -
Segmentation fault (core dumped)
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Both attributes.h and bswap.h have been included from the very beginning
of this muxer without there being any reason to do so.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Support for VDPAU accelerated VP9 decoding was added with libvdpau-1.3.
Support for the same in ffmpeg is added with this patch. Profiles
related to VDPAU VP9 can be found in latest vdpau.h present in
libvdpau-1.3. DRC clips are not supported yet due to
http://trac.ffmpeg.org/ticket/8068
Add VP9 VDPAU to list of hwaccels and supported formats
Added file vdpau_vp9.c and Modified configure to add VDPAU VP9 support.
Mapped VP9 profiles to VDPAU VP9 profiles. Populated the codec specific
params that need to be passed to VDPAU.
Signed-off-by: Philip Langdale <philipl@overt.org>
04d2540c added intreadwrite.h to avienc.c, although there was (and is)
no need to do so. The inclusion seems to be a mistake as this commit
added a AV_WL32 to avidec.c.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
bswap.h was included since 7b114c09, yet since 3788a3c0 no explicit use
of anything from bswap.h has been made, so remove this header.
(Only AV_RL32 is used and while this might imply swapping on
big-endian systems, it is contained in libavutil/intreadwrite.h.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
libavutil/parseutils.h has been included in 22bbd6e8 for av_parse_time()
and the header has not been removed when said function was replaced by
ff_parse_creation_time_metadata() in ea1bf08a.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fix#7620
In the case tee muxer with both "bsf" and "use_fifo" parameters
will trigger this bug. Tee muxer will first steal parameters (like "f",
"select"...) and then "use_fifo" will try reading out remaining options
and pass them to fifo as option "format_options".
Current code miss the part of stealing "bsf" options.
Fixes: signed integer overflow: -1801695444 + -830224908 cannot be represented in type 'int'
Fixes: 17995/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TRUESPEECH_fuzzer-5648084880588800
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -760459023 + -1520918047 cannot be represented in type 'int'
Fixes: 17994/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_INTERPLAY_ACM_fuzzer-5647123042795520
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
ffwavesynth can produce large amounts of data relatively slowly on very small input
Fixes: Timeout (60sec -> 9sec)
Fixes: 17970/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_FFWAVESYNTH_fuzzer-5689121279836160
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
In order to fix a potential memleak upon failure, 0b8956b2 made sure that
a buffer given by a pointer was freed upon error. But this pointer was
only initialized upon use and in several cases (Clang gives no fewer
than 13 -Wsometimes-uninitialized warnings) this meant that an
uninitialized pointer was used to free a buffer. So initialize the
pointer.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
This option allows more control over the use of intra macroblocks in
predictive frames.
By using '-intra_penalty max', intra macroblocks are never used in
predictive frames.
It is useful for glitch artists to generate input material. This option
allows them to split and merge two video files while maintaining fluid
motion from the second video without having intra macroblocks restoring
chunks of the first video.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Prevents memleaks in situations where the trailer isn't written, e.g.
because of errors during writing the header.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Extradata is supposed to be padded with AV_INPUT_BUFFER_PADDING_SIZE bytes,
yet the VobSub demuxer used av_strdup for the allocation of extradata.
This has been changed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Contains renaming of variables (e.g. mkv_write_cues() contained
variables called tracknum that actually contain the index of a track in
s->streams and not the track number (which can differ in case an
explicit dash track number is set)).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
mkv_start_new_cluster() actually didn't start a new cluster, but ended
the old one instead and emitted a debug message that it had started a
new cluster. This has been changed: The debug message has been moved to
the place that really starts a new cluster and the function has been
renamed to mkv_end_cluster().
Furthermore, without this debug message the function can be used for
flushing.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The Matroska muxer groups index entries with the same pts together in
order to save a few bytes. Because of Matroska's variable-length length
fields, mkv_write_cues() does this by first finding out how many index
entries will be grouped together before actually writing them.
Currently, it is asserted at both of these stages that the stream index
of the list of designated index entries is valid. But the second assert
is redundant, because the very same index entries have already been
checked.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The Matroska muxer up until now leaked memory in two scenarios:
1. If an error happened during writing the trailer, as
mkv_write_trailer() returned early without cleaning up.
2. If mkv_write_header() indicated success despite an error in the
underlying AVIOContext. In this case avformat_write_header() returned
the IO error and according to the API the caller is not allowed to call
av_write_trailer(), so that no cleanup happened for the allocations made
in mkv_write_header().
This has been fixed by using a dedicated deinit function.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
If the trailer is never writen, there could be buffered pages that would leak.
Reviewed-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Despite the doxy stating that it's called when the muxer is destroyed,
this was not true in practice. It's only called by av_write_trailer()
and on init() failure.
An AVFormatContext may be closed without writing the trailer if errors
ocurred while muxing packets, so in order to prevent memory leaks, it
should effectively be called when freeing the muxer.
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes: signed integer overflow: -3 - 9223372036854775807 cannot be represented in type 'long'
Fixes: 17828/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5645915116797952
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 2147483628 + 128 cannot be represented in type 'int'
Fixes: 17783/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ALAC_fuzzer-5146470595952640
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -2147481503 + -32732 cannot be represented in type 'int'
Fixes: 17782/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SMACKAUD_fuzzer-5769672225456128
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -1795675744 + -1926578528 cannot be represented in type 'int'
Fixes: 17741/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_FFWAVESYNTH_fuzzer-5131336402075648
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 62220 * 262144 cannot be represented in type 'int'
Fixes: 17145/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_VC1IMAGE_fuzzer-5667394743173120
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
avformat_free_context() expects AVFormatContext->internal to not be NULL.
Reviewed-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
zmbv has only one function for decoding intra frames, namely
decode_intra; and yet up until now it has been called via a function
pointer. This has been changed.
This also removes spec-incompliant conversions between function pointers
and pointers of type void * and thereby fixes the warning "ISO C forbids
assignment between function pointer and ‘void *’" that GCC emits with
the -pedantic option.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
1. Since bd90a2ec, mpeg4_unpack_bframes caches whole packets instead of
just the pointer to the buffer and the buffer's size in order to be able
to make use of refcounting to avoid copying of data; this unfortunately
introduced copies of packet structures and side data (if existing),
although the only fields that are needed are the buffer-related ones
(data, size and buf). This can be changed without compromising the
advantages of refcounting by storing a reference to the buffer.
2. This change also makes it easy to use only one packet throughout
so that an allocation and free of an AVPacket structure per filtered
packet can be saved by switching to ff_bsf_get_packet_ref.
3. Furthermore, this commit also fixes a memleak introduced in bd90a2ec:
If a stored b_frame with side data was used for a later frame, the side
data would leak when the input frame's properties were copied into the
output frame.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If there is an error in mpeg_mux_init() (the write_header function of
the various MPEG-PS muxers), two things might happen:
1. Several fifos might leak. Instead of freeing them, the goto fail part
of the functions freed the private data of the AVStreams instead,
although this will be freed later in free_stream() anyway.
2. And if the function is exited via goto fail, it automatically
returned AVERROR(ENOMEM), although this is also used when the error is
not a memory allocation failure.
Both of these issues happened in ticket #8284 and have been fixed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Up until now, the vp9_superframe bsf used distinct packets for input and
output. But at no point in the bsf are the input and output packets used
at the same time (except during a call to av_packet_move_ref()), so that
one can avoid using two packets if one switches to ff_bsf_get_packet_ref
at the same time. This also saves one malloc+free of an AVPacket
structure per filtered packet.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
remove_doubling() returned a const float, but returning qualified types
is pointless anyway. This is all the more important given that GCC 4.2.1
(and maybe other compilers, too) seems to treat this as an error as several
FATE units that don't compile any more show.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Fixes: signed integer overflow: 518484152 + 1868182638 cannot be represented in type 'int'
Fixes: 17732/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SBC_fuzzer-5663738132168704
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -2147483360 - 631 cannot be represented in type 'int'
Fixes: 17701/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ADPCM_IMA_EA_EACS_fuzzer-5711517319692288
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The pointer fp after the call to chromaprint_get_raw_fingerpoint() points to an array
of uint32_t whereas the current code assumed just a char stream. Thus when writing the
raw fingerprint, the output would be truncated by a factor of 4.
For reference the declaration of the function from chromaprint.h is:
int chromaprint_get_raw_fingerprint(ChromaprintContext *ctx, uint32_t **fingerprint, int *size);
and also change CONV to DLT_CONV2D for better description
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
Removed (new_size - pkt->nals_allocated) because this value is always 1
during the call.
Based on commit 78b86c30d3.
Signed-off-by: James Almer <jamrial@gmail.com>
Without reseting nal_buffer_size, av_fast_realloc will fail if
ff_h2645_packet_split is called with the unitialized pkt as argument.
Signed-off-by: James Almer <jamrial@gmail.com>
The "type" entry was hardcoded with an trailing comma, even if it was
the only entry in the section.
Fixes ticket #8228.
Signed-off-by: James Almer <jamrial@gmail.com>
With floats we cannot represent all 32bit integer dimensions
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Change the stride argument to ptrdiff_t in the following functions:
ff_put_no_rnd_vc1_chroma_mc8_mmi, ff_put_no_rnd_vc1_chroma_mc4_mmi,
ff_avg_no_rnd_vc1_chroma_mc8_mmi, ff_avg_no_rnd_vc1_chroma_mc4_mmi.
Reviewed-by: Shiyou Yin <yinshiyou-hf@loongson.cn>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Allows the creation of the sdtp atom while remuxing MP4 to MP4. This
atom is required by Apple devices (iPhone, Apple TV) in order to accept
2160p medias.
zmbvenc allocates a buffer for a picture with padding on all four sides:
The stride is choosen so large that it already contains padding on the
right; the height also includes padding rows. The padding on the right
of each row is also reused as padding for the left of the next row. So
one still needs to add padding on the left for the first row. This is done
by offsetting the actual pointer used to access the picture from the
pointer returned by av_mallocz and the formula for this offset was
wrong, because it ignored that a pixel can take more than one byte when
calculating the offset resulting from the left padding of the first row.
This fixes accesses outside of the allocated buffer that were reported
in tickets #7980 and #7994. No writes were ever attempted outside of
the buffer.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Tomas Härdin <tjoppen@acc.umu.se>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
do_streamcopy() has a packet that gets zero-initialized first, then gets
initialized via av_init_packet() after which some of its fields are
oerwritten again with the actually desired values (unless it's EOF): The
side data is copied into the packet with av_copy_packet_side_data() and
if the source packet is refcounted, the packet will get a new reference
to the source packet's data. Furthermore, the flags are copied and the
timestamp related fields are overwritten with new values.
This commit replaces this by using av_packet_ref() to both initialize
the packet as well as populate its fields with the right values (unless
it's EOF again in which case the packet will still be initialized). The
differences to the current approach are as follows:
a) There is no call to a deprecated function (av_copy_packet_side_data())
any more.
b) Several fields that weren't copied before are now copied from the source
packet to the new packet (e.g. pos). Some of them (the timestamp related
fields) may be immediately overwritten again and some don't seem to be
used at all (e.g. pos), but in return using av_packet_ref() allows to forgo
the initializations.
c) There was no check for whether copying side data fails or not. This
has been changed: Now the program is exited in this case.
Using av_packet_ref() does not lead to unnecessary copying of data,
because the source packets are already always refcounted (they originate
from av_read_frame()).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Not only the first, but each latest chunk must be cached to allow
seekback after finding the mime boundary.
Fixes trac #5023 and #5921.
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
According to RFC1341, the multipart boundary indicated by the
Content-Type header must be prepended by CRLF + "--", and followed
by CRLF. In the case of strict MIME header boundary handling, the
"--" was forgotten to add.
Fixes trac #7921.
A side effect is that this coincidentally breaks enforcement of
strict MIME headers against servers running motion < 3.4.1, where
the boundary announcement in the HTTP headers incorrectly used the
prefix "--", which exactly matched this bug's behavior.
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The string matching function's return value was evaluated incorrectly.
Fixes trac #7920.
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: shift exponent -2 is negative
Fixes: 17736/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_PCM_F16LE_fuzzer-5742815929171968
Fixes: 17998/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_PCM_F24LE_fuzzer-5716980383875072
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
when the area outside of the frame, then use expr should
give user warning message and auto set to the area inside of the frame.
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Enabled one thread per plane, used the test command for 1080P video
(YUV420P format) as follow:
ffmpeg -i 1080p.mp4 -an -vf hqdn3d -f null /dev/nul
This optimization improved the performance about 30% in 1080P YUV420P
case (from 110fps to 143fps), also pass the framemd5 check and FATE.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Reviewed-by: Moritz Barsnick <barsnick@gmx.net>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
GPU copy enables or disables GPU accelerated copying between video
and system memory. This may lead to a notable performance improvement.
Memory must be sequent and aligned with 128x64.
CMD:
ffmpeg -init_hw_device qsv=hw -filter_hw_device hw -c:v h264_qsv
-gpu_copy on -i input.h264 -f null -
or:
ffmpeg -c:v h264_qsv -gpu_copy on -i input.h264 -f null -
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Signed-off-by: ChaoX A Liu <chaox.a.liu@intel.com>
Signed-off-by: Zhong Li <zhong.li@intel.com>
There is no change in the encoded bitstream, but this
ensures that the written field length is consistent
with the reference implementation.
Unused bytes are zeroed out for backwards compatibility.
Signed-off-by: Raphaël Zumer <rzumer@tebako.net>
the data_start is after reading 12 bytes and if its subtracted
at the very end the intermediate might overflow
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This seeks to the position the previous call to dxv_decompress_opcodes()
positioned us in case of success
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
made with persistent connections to prevent incorrect reset of
offset when demuxing HLS+FMP4
Signed-off-by: Steven Liu <lq@onvideo.cn>
Signed-off-by: vectronic <hello.vectronic@gmail.com>
add ff_http_do_new_request2() which supports options to be applied to
HTTPContext after initialisation with the new uri
Signed-off-by: Steven Liu <lq@onvideo.cn>
Signed-off-by: vectronic <hello.vectronic@gmail.com>
Linux and OSX systems support basename and dirname via <libgen.h>, I plan to
make the wrapper interface conform to the standard interface first.
If it is feasible, I will continue to modify it to call the system interface
if there is already a system call interface.
You can get more description about the system interface by below command:
"man 3 basename"
Reviewed-by: Marton Balint <cus@passwd.hu>
Reviewed-by: Tomas Härdin <tjoppen@acc.umu.se>
Reviewed-by: Steven Liu <lq@chinaffmpeg.org>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Current default is 200kbps, which produces inconsistent
results (too high for low-res, too low for hi-res). Use
CRF instead, which will adapt. Affects VP9. Also have
VP8 use a default bitrate of 256kbps.
Signed-off-by: James Zern <jzern@google.com>
When flushing, MAX_FRAME_HEADER_SIZE bytes (always zero) are supposed to
be written to the fifo buffer in order to be able to check the rest of
the buffer for frame headers. It was intended to write these by writing
a small buffer of size MAX_FRAME_HEADER_SIZE to the buffer. But the way
it was actually done ensured that this did not happen:
First, it would be checked whether the size of the input buffer was zero,
in which case it buf_size would be set to MAX_FRAME_HEADER_SIZE and
read_end would be set to indicate that MAX_FRAME_HEADER_SIZE bytes need
to be written. Then it would be made sure that there is enough space in
the fifo for the data to be written. Afterwards the data is written. The
check used here is for whether buf_size is zero or not. But if it was
zero initially, it is MAX_FRAME_HEADER_SIZE now, so that not the
designated buffer for writing MAX_FRAME_HEADER_SIZE is written; instead
the padded buffer (from the stack of av_parser_parse2()) is used. This
works because AV_INPUT_BUFFER_PADDING_SIZE >= MAX_FRAME_HEADER_SIZE.
Lateron, buf_size is set to zero again.
Given that since 7edbd536, the actual amount of data read is no longer
automatically equal to buf_size, it is completely unnecessary to modify
buf_size at all. Moreover, modifying it is dangerous: Some allocations
can fail and because buf_size is never reset to zero in this codepath,
the parser might return a value > 0 on flushing.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
For a parser, the input buffer is always != NULL: In case of flushing,
the indicated size of the input buffer will be zero and the input buffer
will point to a zeroed buffer of size 0 + AV_INPUT_BUFFER_PADDING.
Therefore one does not need to check for whether said buffer is NULL or
not.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The flac parser uses a fifo to buffer its data. Consequently, when
searching for sync codes of flac packets, one needs to take care of
the possibility of wraparound. This is done by using an optimized start
code search that works on each of the continuous buffers separately and
by explicitly checking whether the last pre-wrap byte and the first
post-wrap byte constitute a valid sync code.
Moreover, the last MAX_FRAME_HEADER_SIZE - 1 bytes ought not to be searched
for (the start of) a sync code because a header that might be found in this
region might not be completely available. These bytes ought to be searched
lateron when more data is available or when flushing.
Unfortunately there was an off-by-one error in the calculation of the
length to search of the post-wrap buffer: It was too large, because the
calculation was based on the amount of bytes available in the fifo from
the last pre-wrap byte onwards. This meant that a header might be
parsed twice (once prematurely and once regularly when more data is
available); it could also mean that an invalid header will be treated as
valid (namely if the length of said invalid header is
MAX_FRAME_HEADER_SIZE and the invalid byte that will be treated as the
last byte of this potential header happens to be the right CRC-8).
Should a header be parsed twice, the second instance will be the best child
of the first instance; the first instance's score will be
FLAC_HEADER_BASE_SCORE - FLAC_HEADER_CHANGED_PENALTY ( = 3) higher than
the second instance's score. So the frame belonging to the first
instance will be output and it will be done as a zero length frame (the
difference of the header's offset and the child's offset). This has
serious consequences when flushing, as returning a zero length buffer
signals to the caller that no more data will be output; consequently the
last frames not yet output will be dropped.
Furthermore, a "sample/frame number mismatch in adjacent frames" warning
got output when returning the zero-length frame belonging to the first
header, because the child's sample/frame number of course didn't match
the expected sample frame/number given its parent.
filter/hdcd-mix.flac from the FATE-suite was affected by this (the last
frame was omitted) which is the reason why several FATE-tests needed to
be updated.
Fixes ticket #5937.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The FLACHeaderMarker structure contained a pointer to an array of int;
said array was always allocated and freed at the same time as its
referencing FLACHeaderMarker; the pointer was never modified to point to
a different array and each FLACHeaderMarker had its own unique array.
Furthermore, all these arrays had a constant size. Therefore include
this array in the FLACHeaderMarker struct.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
FLAC sync codes contain a byte equal to 0xFF and so the function that
searches for sync codes first searched for this byte. It did this by
checking four bytes at once; these bytes have been read via AV_RB32, but
the test works just as well with native endianness.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Put an AVIOContext whose lifetime doesn't extend beyond the function
where it is allocated on the stack instead of allocating and freeing it.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
They are not allowed outside of functions. Fixes the warning
"ISO C does not allow extra ‘;’ outside of a function [-Wpedantic]"
when compiling with GCC and -pedantic.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Fixes: left shift of 1 by 31 places cannot be represented in type 'int'.
Affected the FATE-tests vsynth1-cinepak, vsynth2-cinepak and
vsynth_lena-cinepak. Also fixes ticket #8220.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Put an AVIOContext whose lifetime doesn't extend beyond the function where
it is allocated on the stack instead of allocating and freeing it.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When used ROUNDED_DIV(a,b), if a is unsigned integer zero, it's
will lead to an underflow issue(it called unsigned integer
wrapping).
Fixes#8062
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Mengye Lv <mengyelv@tencent.com>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
The argument to vec_splat_u16 must be a literal. By making the
function always inline and marking the arguments const, gcc can
turn those into literals, and avoid build errors like:
swscale_vsx.c:165:53: error: argument 1 must be a 5-bit signed literal
Fixes#7861.
Signed-off-by: Daniel Kolesa <daniel@octaforge.org>
Signed-off-by: Lauri Kasanen <cand@gmx.com>
While this technically compiles in current ffmpeg, this is only
because ffmpeg is compiled in strict ISO C mode, which disables
the builtin 'vector' keyword for AltiVec/VSX. Instead this gets
replaced with a macro inside altivec.h, which defines vector to
be actually __vector, which accepts random types.
Normally, the vector keyword should be used only with plain
scalar non-typedef types, such as unsigned int. But we have the
vec_(s|u)(8|16|32) macros, which can be used in a portable manner,
in util_altivec.h in libavutil.
This is also consistent with other AltiVec/VSX code elsewhere in
the tree.
Fixes#7861.
Signed-off-by: Daniel Kolesa <daniel@octaforge.org>
Signed-off-by: Lauri Kasanen <cand@gmx.com>
The structure of a ProRes frame in mov/mp4 is that of a typical atom:
First a 32 bit BE size field, then a tag detailling the content. Said
size field includes the eight bytes of the atom header.
This header is actually redundant, as the size of the atom is already
known from the containing atom. It is therefore stripped away when muxed
into Matroska and so the Matroska demuxer has to recreate upon demuxing.
But it did not account for the fact that the size field includes the
size of the header and this can lead to problems when a decoder uses the
in-band size field.
Fixes ticket #8210.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes: null pointer dereference
Fixes: signed integer overflow: 512 * 2147483647 cannot be represented in type 'int'
Fixes: 17809/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_XMA1_fuzzer-5634409947987968
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The check "if (!pb->seekable & AVIO_SEEKABLE_NORMAL)" is wrong, because
! has higher precendence than &. But it is also redundant, because this
part of the code is only ever reached when the AVIO_SEEKABLE_NORMAL flag
is set for pb. So simply remove the check.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Matthieu Bouron <matthieu.bouron@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Up until now, aiffenc didn't rely on the standard functions for adding
an element to a linked list and freeing the list, but instead
reimplemented them. This has been changed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Matthieu Bouron <matthieu.bouron@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This basically checks if a pixel that was coded with prediction
and residual could have been stored using a previous case.
This avoids basically a string of 0 symbols stored in less than
50 bytes to hit a O(n²) codepath.
Fixes: Timeout (too slow to wait -> immediately)
Fixes: 8668/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_G2M_fuzzer-4895946310680576
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 2 * -1306460384 cannot be represented in type 'int'
Fixes: 17685/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AAC_fuzzer-5747390337777664
Fixes: 17688/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_INTERPLAY_ACM_fuzzer-5739287210885120
Fixes: 17699/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_INTERPLAY_ACM_fuzzer-5678394531905536
Fixes: 17738/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TAK_fuzzer-5763415733174272
Fixes: 17746/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_BINKAUDIO_RDFT_fuzzer-5703008159006720
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
add new function duration_estimate_name to dump duration estimate
method, it's will help to debug some duration issue.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
change the log level to warning if can't get duration, it's will help
to debug some duration issue
Signed-off-by: vacingfang <vacingfang@tencent.com>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
When performing a comparison of a signed int and an unsigned int, the
signed int is first converted to an unsigned int, so that negative
values are being treated as big, positive values. This can become a
problem in an overread check, namely when an overread already happened.
So change the type of the variable containing the amount of bits that
need to be left to signed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The maximum allowed index for an array access is FF_ARRAY_ELEMS - 1; yet
the current code allowed FF_ARRAY_ELEMS. This wasn't dangerous in practice,
as parameter sets with invalid ids were already filtered out during
reading.
Found via PVS-Studio (see ticket #8156).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
header.data_max and header.data_min are not necessarely set on all decoding scenarios.
Fixes a Valgrind reported regression since cfa1937791.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Up until now, read_frame_internal always initialized the packet it
received. But since the recent changes to ff_read_packet, this is no
longer needed: If the parsing queue is initially empty upon entering
read_frame_internal, the packet will now either contain content upon
success or be blank upon failure of ff_read_packet. If the parsing
queue is initially not empty, the packet will be overwritten with the
oldest one from the parsing queue.
Similarly, it is unnecessary to initialize ret in read_frame_internal.
In parse_packet, it is easily possible to only initialize the packet
used as temporary storage for the output if said packet is used at all;
furthermore, this packet doesn't need to be zero-initialized, because
av_init_packet will initialize every field except size and data and
those fields will be set by av_parser_parse2.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until now, parse_packet() used a stack packet in case the stream is
flushed. But using such a packet is unnecessary as there is an AVPacket
readily available, it just needs to be used. Whether flushing is intended
or not will now be signalled by an explicit parameter rather than by
whether the packet parameter is NULL. This removes a few checks in
parse_packet(), gets rid of the initialization of the stack packet and
also reduces usage of sizeof(AVPacket) in libavformat.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until now, read_frame_internal in avformat/utils.c uses a spare
packet on the stack that serves no real purpose: At no point in this
function is there a need for another packet besides the packet destined
for output:
1. If the packet doesn't need a parser, but is output as is, the content
of the spare packet (that at this point contains a freshly read packet)
is simply copied into the output packet (via simple assignment, not
av_packet_move_ref, thereby confusing ownership).
2. If the packet needs parsing, the spare packet will be reset after
parsing and any packets resulting from the packet read will be put into
a packet list; the output packet is not used here at all.
3. If the stream should be discarded, the spare packet will be
unreferenced; the output packet is not used here at all either.
Therefore the spare packet and the copies can be removed in principle.
In practice, one more thing needs to be taken care of: If ff_read_packet
failed, the output packet was not affected, now it is. But given that
ff_read_packet returns a blank (as if reset via av_packet_unref) packet
on failure, there is no problem from this side either.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
When AVFMT_FLAG_GENPTS is set, av_read_frame would put a reference to a
packet in the packet list (via av_packet_ref) and then immediately
thereafter unreference the original packet. This has been changed to
move the reference instead.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until now, avformat_find_stream_info had a potential for memleaks:
When everything was fine, it read packets and (depending upon whether
AVFMT_FLAG_NOBUFFER was set) put them in a packet list or unreferenced
them when they were no longer needed. But upon failure, said packets
would leak if they were not already on the packet list. This patch fixes
this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Since the recent changes to ff_packet_list_put, the source packet will
be automatically reset when the reference is moved to the packet list,
so that it is unnecessary to reinitialize the packet in the loops in
parse_packet and ff_read_packet; initializing once at the beginning is
enough.
This also fixes a potential, but currently unexisting problem: If the
raw packet buffer was initially not empty and probe_codec() failed,
then the packet returned would not be initialized. But given that
probe_codec() currently can't fail (always returns 0) this was not an
acute danger.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Fixes: signed integer overflow: -1094995519 * 64 cannot be represented in type 'int'
Fixes: 17030/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_RALF_fuzzer-5640695838146560
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: overflow in aspect ratio calculation
Fixes: signed integer overflow: 393215 * 14594 cannot be represented in type 'int'
Fixes: 15728/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMV3IMAGE_fuzzer-5661588893204480
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
"9.1.1.43 P Reference Distance (REFDIST)"
"The value of REFDIST shall be less than, or equal to, 16."
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
We need to reset samples in this case to avoid being stuck with incorrect
samples value.
Fixes: OOM
Fixes: Timeout
Fixes: 16627/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-5638059583864832
Fixes: 17089/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-5672188463546368
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: negation of -2147483648 cannot be represented in type 'int'; cast to an unsigned type to negate this value to itself
Fixes: signed integer overflow: -14527961 - 2147483425 cannot be represented in type 'int'
Fixes: 16380/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-5645957131141120
Fixes: 16968/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-5716169901735936
Fixes: 17074/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-5198710497083392
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Values larger would fail subsequent tests.
Fixes: signed integer overflow: 5 + 2147483646 cannot be represented in type 'int'
Fixes: 16966/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_HEVC_fuzzer-5695709549953024
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Affected every usage of this filter; in particular, it affected the
FATE-tests filter-2xbr, filter-3xbr and filter-4xbr.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Affected the vsynth*-jpeg2000 and the vsynth*-jpeg2000-97 FATE tests
(where * ranges over { 1, 2, 3, _lena }) as well as ticket #7983.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Affected the FATE-tests vsynth_lena-dv-411, vsynth1-dv-411,
vsynth2-dv-411 and hevc-paramchange-yuv420p.yuv420p10.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
libavcodec/ffv1enc.c accessed an array of uint8_t [32] via array[0][j]
in order to loop over all the uint8_t in this array of arrays. Of course
this implied an out-of-bounds access for array[0] and UBSan complained
about this. So replace this with nested loops; furthermore, factor this
out into a function of its own to easily break out of the nested loops.
This affected the FATE-tests vsynth1-ffv1, vsynth1-ffv1-v3-yuv420p,
vsynth1-ffv1-v3-yuv422p10, vsynth1-ffv1-v3-yuv444p16,
vsynth1-ffv1-v3-bgr0, vsynth1-ffv1-ffv1-v3-rgb48 as well as the
corresponding vsynth2-*, vsynth3-* and the vsynth_lena-* tests.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This affected many FATE-tests: The number of failing tests went down
from 663 to 344. (Both numbers exclude tests that failed because of
unaligned accesses in code that is inside #if HAVE_FAST_UNALIGNED.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Identical to avpriv_mpeg4audio_get_config() except taking a size argument in
bytes, and featuring a new logging context paremeter.
Schedule avpriv_mpeg4audio_get_config() for removal as soon as major is bumped
as well.
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes: Infinite loop
Fixes: signed integer overflow: 2147483644 + 16 cannot be represented in type 'int'
Fixes: 16169/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_FOURXM_fuzzer-5662570416963584
Fixes: 16782/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_FOURXM_fuzzer-5743163859271680
Fixes: 17641/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_FOURXM_fuzzer-5711603562971136
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Affected the FATE-tests exr-rgb-scanline-pxr24-half-uint32-13x9 and
exr-rgb-scanline-pxr24-uint32.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Since reading 4 bits always returns a value in the range [0, 15], the
check for vps_id >= HEVC_MAX_VPS_COUNT, where HEVC_MAX_VPS_COUNT = 16, is redundant.
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The MPEG-1/2 decoder uses avpriv_find_start_code to search for start
codes and worked with the resulting start code before checking that it
is really a start code of a slice. In particular, if the picture is so
big that a slice_vertical_position_extension is present, it added the
slice_vertical_position_extension as if it had a slice. Then a left
shift is performed, without making sure that the value to be shifted is
nonnegative.
Afterwards the end result is checked, but even if a start code of a
non-slice has been found, it might pass these checks: If
slice_vertical_position_extension is present a start code <
SLICE_MIN_START_CODE can lead to a macroblock-row index that appears
valid. Furthermore, the left shift might make an invalid start code
appear valid by discarding the highest bit.
This has been fixed by checking directly after avpriv_find_start_code
has returned.
Fixes ticket #8162 (which is about the undefined left shifts).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
MSDK does not create internal acceleration device on Linux,
So MFXVideoCORE_SetHandle() is necessary.
It has been added for ff_qsv_init_session_device().
But missed for ff_qsv_init_internal_session() due to commit
1f26a23 overwrited commit db89f45Fix#7030
Signed-off-by: Zhong Li <zhong.li@intel.com>
This is done so that its data is really owned by the packet.
This was already true for the current callers.
Signed-off-by: James Almer <jamrial@gmail.com>
Up until now, ff_packet_list_put had a flaw: When it moved a packet to
the list (meaning, when it ought to move the reference to the packet
list instead of creating a new one via av_packet_ref), it did not reset
the original packet, confusing the ownership of the data in the packet.
This has been done because some callers of this function were not
compatible with resetting the packet.
This commit changes these callers and fixes this flaw. In order to
indicate that the ownership of the packet has moved to the packet list,
pointers to constant AVPackets are used whenever the target of the
pointer might already be owned by the packet list.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
ff_read_packet had potential memleaks:
1. If av_packet_make_refcounted fails, it means that the packet is not
refcounted, but it could nevertheless carry side data and therefore
needs to be unreferenced.
2. If putting a packet on a packet list fails, it wasn't unreferenced.
Furthermore, read_frame_internal leaked a packet's (side) data if a
context update was required and failed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
There is currently an ordinary check for this (which would lead to a
memleak), but given that no demuxer should ever return a packet with an
invalid stream_index it is more appropriate for this to be an assert.
FATE passes with this change.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
4650975 -> 4493240 dezicycles
This optimizes lines 2 and later. Line 1 still uses av_memcpy_backptr()
This change originally fixed ossfuzz 10790 but this is now fixed by other
optimizations already
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 238 * 16843009 cannot be represented in type 'int'
Fixes: 16958/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SMACKER_fuzzer-5193905355620352
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -2147483648 * -1 cannot be represented in type 'int'
Fixes: 16786/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ALAC_fuzzer-5632818851348480
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Several subtitle demuxers set negative durations
Fixes: signed integer overflow: 9223372036854775807 - -1 cannot be represented in type 'long'
Fixes: 16925/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5766519790764032
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 9223371075321077760 * 2 cannot be represented in type 'long'
Fixes: 16447/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_FFWAVESYNTH_fuzzer-5698937431785472
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If adding an SEI message to an access unit fails, said SEI message was
not touched, so that the caller had to free any data associated with it
that might need to be freed. But given that ff_cbs_h264_add_sei_message
can simply call cbs_h264_free_sei_payload, one can easily free
the content of the SEI payload.
This fixes a memleak when inserting a user data unregistered string for
h264_metadata fails.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This can happen when av_bsf_free() is called on av_bsf_alloc() failure.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
index into segmentlists_tab was specified as 4 instead of 3 causing invalid access
further fix to: 8135
Reviewed-by: Steven Liu <lq@onvideo.cn>
Signed-off-by: vectronic <hello.vectronic@gmail.com>
prevent attempt to call xmlFree if val was not allocated
fixes: 8135
Reviewed-by: Steven Liu <lq@onvideo.cn>
Signed-off-by: vectronic <hello.vectronic@gmail.com>
this conflicts with the current contributing guidance:
http://ffmpeg.org/developer.html#Contributing
FFmpeg is programmed in the ISO C90 language with a few additional
features from ISO C99, namely:
...
for loops with variable definition (‘for (int i = 0; i < 8; i++)’);
Signed-off-by: James Zern <jzern@google.com>
Implemented as a variant of the hash muxer, reusing most functions,
and making use of the previously introduced array of hashes.
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Only the first element of the array is used currently, the other
elements are in preparation for a new muxer calculating multiple
hashes.
Also move alloc/init code from the write_header() functions to
dedicated init() functions, and the cleanup code from the
write_trailer() functions to dedicated deinit() functions.
hash_free() and framehash_free() turn out to be identical here,
but will differ in the subsequent commit, so they are not consolidated.
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Only the frame* muxers support the format_version option.
Use macros to ease the proliferation of identical options to
coming muxers as well.
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The pointer arguments to memcpy (and several other functions of the
C standard library) are not allowed to be NULL, not even when the number
of bytes to copy is zero. An AVEncryptionInitInfo's data pointer is
explicitly allowed to be NULL and yet av_encryption_init_info_add_side_data
unconditionally used it as a source pointer to copy from. This commit changes
this so that copying is only done if the number of bytes to copy is > 0.
Fixes ticket #8141 as well as a part of ticket #8150.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Tomas Härdin <tjoppen@acc.umu.se>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
It's often not obvious how option constants relate to numerical values.
Defaults are sometimes printed as numbers and from/to are always printed as numbers.
Printing the numeric values of options constants avoids this confusion.
It also allows to see which constants are equivalent.
Before this patch:
-segment_list_type <int> E........ set the segment list type (from -1 to 4) (default -1)
flat E........ flat format
csv E........ csv format
ext E........ extended format
ffconcat E........ ffconcat format
m3u8 E........ M3U8 format
hls E........ Apple HTTP Live Streaming compatible
Afterwards:
-segment_list_type <int> E........ set the segment list type (from -1 to 4) (default -1)
flat 0 E........ flat format
csv 1 E........ csv format
ext 3 E........ extended format
ffconcat 4 E........ ffconcat format
m3u8 2 E........ M3U8 format
hls 2 E........ Apple HTTP Live Streaming compatible
Signed-off-by: softworkz <softworkz@hotmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Integer values should not be printed using format specifier '%g' which leads to inexact display in case of higher values.
Before this patch:
-trans_color <int> .D.V..... color value [...] (default 1.67772e+07)
Afterwards:
-trans_color <int> .D.V..... color value [...] (default 16777215)
Signed-off-by: softworkz <softworkz@hotmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes the FATE-tests unknown_layout-ac3, ac3-fixed-encode, ac3-encode
and eac3-encode. It furthermore fixes the ac3-encoder bugs mentioned in
tickets #7994, #8144 and #8159.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The reason to add this layer is that it is used by srcnn in vf_sr.
This layer is currently ignored in native mode. After this patch,
we can add multiple outputs support for native mode.
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
the logic is that one layer in one separated source file to make
the source files simple for maintaining.
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
the logic is that one layer in one separated source file to make
the source files simple for maintaining.
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
av_mod_uintp2_c uses a bitwise AND with (1 << p) - 1 to clear the high
bits of an unsigned int. But this is undefined if p == 31, because 1 is
an int and 2^31 is not representable in an int. So make 1 unsigned.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
av_mallocz + av_init_packet leads to the same result as av_mallocz +
av_packet_unref, but faster.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
1. Instead of relying on ff_packet_list_get to get the oldest element in
an AVPacketList, ff_read_packet used its own ad-hoc code. Said code
forgot to set the end of the list to NULL if the last element of the
list has been removed, thereby leaving the list in an inconsistent state.
2. Furthermore, if the list was not empty, the oldest element of the
list would always be copied into another packet structure before it was
known whether the oldest entry of the list would be removed. This makes
the ownership confusing and potentially copies unnecessarily.
Both of these issues have been fixed. ff_packet_list_get is used now.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The documentation of ff_packet_list_get currently didn't match the
actual usage:
1. It said that the destination packet is supposed to be initialized.
But this makes no sense given that it will be overwritten completely and
flacenc, mp3enc and ttaenc ignored this.
2. ff_packet_list_get returns an int, although it can't fail in case the
packet list is not empty (for which there is an assert). Again, several
callers didn't check for any return value.
In both cases, the documentation has been adapted to match actual usage.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When the mov/mp4 demuxer encounters an error during decrypting a packet,
it returns the error, yet doesn't free the packet, so that the packet
leaks. This has been fixed in this commit.
Fixes the memleaks from ticket #8150.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
ttaenc contained (1 << unary) - 1 as an argument for a function
expecting an unsigned int. unary can be as big as 31 in this case.
The type of the shift and the whole expression is int, because 1 fits
into an integer, so that the behaviour is undefined if unary == 31
as the result of the shift can't be represented in an int §. Subtraction
by 1 (which makes the result of the whole expression representable in
an int) doesn't change that this is undefined (it usually leads to
signed integer overflow which is undefined, too).
The solution is simple: Make 1 unsigned to change the type of the
whole expression to unsigned int (as the function expects anyway).
Fixes ticket #8153.
§: This of course presupposes the common int range of -2^31..2^31-1
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
On iOS 11, encoding a frame may return error with log
"Error encoding frame 0", which means vtenc_output_callback
is called with status=0 and sample_buffer=NULL. Then the
encoding session will be crashed on next callback wether or not
closing the codec context.
Let us look through the link below introducing VTCompressionOutputCallback,
https://developer.apple.com/documentation/videotoolbox/vtcompressionoutputcallback?language=objc
"status=0" (noErr) means compression was successful.
"sampleBuffer=NULL" means the frame was dropped when compression
was successful (status=0) or compression was not successful (status!=0).
So we should not set AVERROR_EXTERNAL on "status=0" and "sample_buffer=NULL"
as it is not a error.
The fix is that we only set AVERROR_EXTERNAL with status value non zero.
When sample_buffer is NULL and status value is zero, we simply return
with no other operation.
This crash often occurs on iOS 11 for example encoding 720p@25fps.
Signed-off-by: sharpbai <sharpbai@gmail.com>
Signed-off-by: Rick Kern <kernrj@gmail.com>
Fixes: signed integer overflow: 57342 * 120830 cannot be represented in type 'int'
Fixes: 16426/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_CFHD_fuzzer-5758744817827840
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
snow allows creating large output from tiny input, like other
wavelet codecs
Fixes: Timeout (82sec -> 1.5sec)
Fixes: 9520/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SNOW_fuzzer-6286612576075776
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Changing details as following:
1. The previous order of parameters are irregular and difficult to
understand. Adjust the order of the parameters according to the
rule: (RTYPE, input registers, input mask/input index/..., output registers).
Most of the existing msa macros follow the rule.
2. Remove the redundant macro SLDI_Bn_0 and use SLDI_Bn instead.
Reviewed-by: Shiyou Yin <yinshiyou-hf@loongson.cn>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This was partially fixed by 233cd89, but it made changes to AVFrame fields
from within end_frame, which doesn't work consistently when multithreading
is enabled. This is what the post_process function is for.
Signed-off-by: Aman Gupta <aman@tmm1.net>
ffio_init_context with NULL seek callback will setting seekable with 0, so
remove the redundancy seekable setting.
Reviewed-by: Steven Liu <lq@onvideo.cn>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
ff_format_io_close will check the AVIOContext pointer pb, so drop
the unnecessary check before ff_format_io_close.
Reviewed-by: Steven Liu <lq@onvideo.cn>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
This commit fixes a memleak in the hls muxer when one uses a single file
as output. It has been forgotten to free the temporary buffers used to write
the packets so that the size of the leaks basically amounts to the size
of the output file. This commit adds the necessary free.
Reviewed-by: Steven Liu <lq@onvideo.cn>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The specification does not seem to list what the maximum valid
value is
Fixes: shift exponent 32 is too large for 32-bit type 'unsigned int'
Fixes: 16268/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ALS_fuzzer-5638164544225280
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -2424832 - 2145653689 cannot be represented in type 'int'
Fixes: 16138/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TAK_fuzzer-5643451346976768
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -822667928 + -1399761199 cannot be represented in type 'int'
Fixes: 15756/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AAC_FIXED_fuzzer-5645182051024896
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
A threshold of 1 is sufficient for simple_dump_cut.webm, 10 is used
just to be sure the next truncated file doesnt cause the same issue
Obvious alternative fixes are to simply accept that the file is broken or to
write some advanced error concealment or to
simply accept that the decoder wont stop at the end of input.
Fixes: Ticket 8069 (artifacts not the differing md5 which was there before 1afd246960)
Fixes: simple_dump_cut.webm
Fixes: regression of 1afd246960
fate-vp5 changes because the last frame is truncated and now handled
differently.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* FFmpeg retrieves a packet from the bitstream
* It attempts to get an input buffer (from its own list or by dequeuing one from the driver)
* If no input buffer is found, the bitstream packet is dropped instead of scheduled for trying again later
It's an issue that showed especially at high speeds (like using `-f null -` as output parameters).
Signed-off-by: Aman Gupta <aman@tmm1.net>
Without this ffmpeg will attempt to copy the dts from the
most recently enqueued packet into the most recently dequeued
frame, which does not account for the buffering inside v4l2
and is not accurate.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Frames are generally dequeued into capture buffers, so using
the output properties here was incorrect. It happened to work
fine for decoding, since the output/capture buffers have the same
dimensions.
For the v4l2 scaler, the dimensions can be different between output
and capture. Using the buffer's associated context makes this code
work correctly regardless of where the frame is coming from.
Signed-off-by: Aman Gupta <aman@tmm1.net>
free work&pi_wraps in error handling path to avoid memory leak.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
free the pad.name in error handling path to avoid memory leak.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
The C and asm versions behaved differently _outside_ of the codec.
The C version returned pre-multiplied 'state' for the next execution
to use right away, while the assembly version outputted non-multiplied
'state' for the next execution to multiply to save instructions.
Since the initial state when initialized or seeking is always 0,
and since C and asm versions were never mixed, there was no issue.
However, comparing outputs directly in checkasm doesn't work without
dividing the initial state by CELT_EMPH_COEFF and multiplying the
returned state by CELT_EMPH_COEFF for the assembly function.
Since its actually faster to do this in C as well, copy the behavior the
asm versions use. As a reminder, the initial state 0 is divided by
CELT_EMPH_COEFF on seek and init (just in case in the future this is
changed, its technically more correct to init with CELT_EMPH_COEFF than 0,
however when seeking this will result in more audiable pops, unlike with 0
where the output gets in sync over a few samples).
This just makes the M2MDEC_CLASS similar to how it is done in rkmpp. It looks
clean and has proper indentation
Signed-off-by: Aman Gupta <aman@tmm1.net>
On the RPI, three different /dev/video devices exist (decoder, scaler, encoder).
When probing the devices in order, the originally requested pix fmt
would be mutated causing the wrong one to be chosen when a matching
device was finally found.
Signed-off-by: Aman Gupta <aman@tmm1.net>
In rare circumstances, if the codec is not configured with the
proper parameters the input buffers can be allocated with a size
that's too small to hold an individual packet. Since MediaCodec
expects exactly one incoming buffer with a given PTS, it is not
valid to split data for a given PTS across two input buffers.
See https://developer.android.com/reference/android/media/MediaCodec#data-processing:
> Do not submit multiple input buffers with the same timestamp
Signed-off-by: Aman Gupta <aman@tmm1.net>
When duplicating frames we need to schedule for activation again, otherwise frames can build up in the inlink fifo.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
some bitstream have trailing junk, despite being valid webp data.
In case of apparent error, abort the loop and let *got_frame
decide whether this is an error or not.
fixes trac #8107 (/#7612)
Another possibility would be turning the loop into:
while (!*got_frame) {...}
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The indentation of code dose not reflect the if block structure in
'apply_ltp_mips', and this will generate a warnning when build with
'-Wall' or '-Wmisleading-indentation'.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The loop may mutate the input buffer, so re-fetch it to ensure
the current one is always used.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Matthieu Bouron <matthieu.bouron@gmail.com>
Because the lavf_container is sometimes called with only 2 arguments,
fate tests produce bash errors like this:
tests/fate-run.sh: 299: test: =: unexpected operator
This commit fixes this.
Reviewed-by: Limin Wang <lance.lmwang@gmail.com>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Due to the recent addition of Vulkan support to AMF, we require more
recent headers that include the new structures, which have been
available since AMF 1.4.9 released in September 2018.
Fixes Ticket #8125
Right now, the concat filter does not set the frame_rate value on any of
the out links. As a result, the default ffmpeg behaviour kicks in - to
copy the framerate from the first input to the outputs.
If a later input is higher framerate, this results in dropped frames; if
a later input is lower framerate it might cause judder.
This patch checks if all of the video inputs have the same framerate, and
if not it sets the out link to use '1/0' as the frame rate, the value
meaning "unknown/vfr".
A test is added to verify the VFR behaviour. The existing test for CFR
behaviour passes unchanged.
this function is used to get the previous shutdown status
when reusing the old connection in block mode.
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Use the command ./configure with/without --disable-v4l2-m2m test.
Reviewed-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
Some decoders may not need a writable buffer in some specific cases, but only
a reference to the existing buffer with updated frame properties instead, for
the purpose of returning duplicate frames. For this, the
FF_REGET_BUFFER_FLAG_READONLY flag is added, which will prevent potential
allocations and buffer copies when they are not needed.
Signed-off-by: James Almer <jamrial@gmail.com>
On some DVB stream SCTE-35 data packet are available before the end of
MpegTSContext initialization. We have to check if it is the case to
avoid a SEGFAULT.
Signed-off-by: Marton Balint <cus@passwd.hu>
When compiling natively on an RPI where libomxil-bellagio-dev
was also installed, `check_headers OMX_Core.h` succeeded and
the -isystem compiler flag was never added to the build.
For non-native builds, the error message now mentions the
raspberrypi/firmware repository where the RPI specific
headers are available.
Signed-off-by: Aman Gupta <aman@tmm1.net>
ENOMEM indicates an allocation failure, and there are no allocations
happening here. The buffers are pre-allocated and there are simply
none available at this time.
Signed-off-by: Aman Gupta <aman@tmm1.net>
When ffmpeg was streaming, multiple clients were only supported by using a
multicast destination address. An alternative was to stream to a server which
re-distributes the content. This commit adds ZeroMQ as a protocol, which allows
multiple clients to connect to a single ffmpeg instance.
Signed-off-by: Marton Balint <cus@passwd.hu>
Integer overflow in the Q16 framerate calculation was sending
invalid values to the OMX encoder.
On the RPI4, this manifested as bitrate controls being ignored
on video streams with 60000/1001 framerates. Video streams with
30000/1001 framerates were not affected.
Signed-off-by: Aman Gupta <aman@tmm1.net>
7d097a0fc5 had the same purpose as
3700f655c5 but the former is much simpler, so
let's remove the latter.
Unfortunately both checks were wrong, because in order to make sure DTS > PCR
we have to give us some headroom, so instead of using a dts_difference <
max_delay check let's use a dts_difference < max_delay/2 check.
Fixes DTS < PCR errors with this command line:
./ffmpeg -loglevel verbose -y -f lavfi -i \
"testsrc=s=64x64:d=20,split=2[out0][tmp1];[tmp1]vflip[out1];sine=d=20,asetnsamples=1000[out2]" \
-flags +bitexact -fflags +bitexact -sws_flags +accurate_rnd+bitexact \
-codec:v libx264 -codec:a mp2 -b:a 32k -pix_fmt yuv420p \
-map '0:v:0' \
-map '0:v:1' \
-map '0:a:0' \
-muxrate 800000 \
-program st=0:st=2 -program st=1:st=2 -program st=2 -program st=0 -f mpegts out1.ts
Signed-off-by: Marton Balint <cus@passwd.hu>
add inner loop specialisations for 2 bpp and 4 bpp
These are all cases for which i found testsamples.
30M cycles -> 5M cycles
Testcase: fate-rgbapng-4816
Testcase: 16097/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APNG_fuzzer-5664690889293824
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 2021654528 + 2032575680 cannot be represented in type 'int'
Fixes: 16270/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-5732438816325632
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
ff_v4l2_m2m_create_context initialized V4L2m2mContext.fd to 0
which is a valid file descriptor value. Next ff_v4l2_m2m_codec_init
failed and v4l2_m2m_destroy_context closed file descriptor 0 even
though it didn't belong to V4L2m2mContext.
Signed-off-by: Pavel Koshevoy <pkoshevoy@gmail.com>
The current code in libavfilter/af_sofalizer.c requires
mysofa_neighborhood_init_withstepdefine function, which only appeared
in libmysofa 0.7. Use this function in configure script to bail out
early if a too old libmysofa is found in the system instead of failing
at compile time.
It is not clear what was real intention of previous commit to this filter.
It was not working correctly, hopefully this is fixed now.
It never checked that new x/y/w/h are actually valid, hopeffully this is fixed now.
It uses named variables in expressions that are never set, still not fixed.
It does not set named variables that uses actual frame widht/height, making actual
expressions less usable for our users, still now fixed.
"LinearRaw" is a value that the PhotometricInterpretation tag can be set
to on DNG images that contain color information for all channels instead
of being bayer-encoded ("CFA" value).
The DNG decoder is complete enough that we can enable this now.
Sample:
- http://www.rawsamples.ch/raws/nikon/SCANNER_NIKON_LS5000.DNG
Signed-off-by: Nick Renieris <velocityra@gmail.com>
Initialized to `(2 ^ BitsPerSample) - 1` as per the DNG Specification.
Also make sure that `BlackLevel < WhiteLevel`.
This fixes decoding for "X7 CinemaDNG" samples here:
- https://www.dji.com/gr/zenmuse-x7/info#downloads
Signed-off-by: Nick Renieris <velocityra@gmail.com>
Also, ensure no false positives when determining DNG bayer images, by
setting them in tiff.c instead of relying on a heuristic. There's no
way to determine this just from the JPEG data, so we have to pass this
information from outside the MJPEG decoder.
Signed-off-by: Nick Renieris <velocityra@gmail.com>
When the height is equal to the tile length (full-height tile)
after `height % tile_length` is applied with the current code,
it results in the operating tile_length to be 0. This commit
makes this leftover logic only applies if it's necessary.
Signed-off-by: Nick Renieris <velocityra@gmail.com>
Some JPEGs [1] have incorrect DHT entries that map 2 codes to
the same value.
The second (last) mapping does not ever actually appear in the
code stream, therefore ignoring any mappings after the first one
fixes this.
Without this, an "mjpeg_decode_dc: bad vlc: 0:0" error is thrown.
In all known files, the 2 codes are mapped to symbol 0 so only
that case is checked.
[1]: Embedded JPEGs in "X7 RAW" and "X7 CinemaDNG" samples here:
https://www.dji.com/gr/zenmuse-x7/info#downloads
Signed-off-by: Nick Renieris <velocityra@gmail.com>
Used a technique similar to lavc/tdsc.c for invoking the MJPEG decoder.
This commit adds support for:
- DNG tiles
- DNG tile huffman lossless JPEG decoding
- DNG 8-bpp ("packed" as dcraw calls it) decoding
- DNG color scaling [1]
- LinearizationTable tag
- BlackLevel tag
[1]: As specified in the DNG Specification - Chapter 5
Signed-off-by: Nick Renieris <velocityra@gmail.com>
Main image data in DNGs is usually comprised of tiles, each of which is a Huffman-encoded lossless JPEG.
Tested for ljpeg regressions with:
`ffmpeg -f lavfi -i testsrc=d=1 -vcodec ljpeg test.avi`
`ffmpeg test.avi out.avi`
The modified code in ljpeg_decode_rgb_scan runs without issues.
Signed-off-by: Nick Renieris <velocityra@gmail.com>
This is an alias for JEDEC P22.
The name associated with the value is also changed
from jedec-p22 to ebu3213 to match ITU-T H.273.
Signed-off-by: Raphaël Zumer <rzumer@tebako.net>
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes: signed integer overflow: 1795032576 + 598344192 cannot be represented in type 'int'
Fixes: 16196/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TRUEMOTION2_fuzzer-5636723419119616
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: left shift of negative value -1
Fixes: 16424/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMV3_fuzzer-5656579055026176
Fixes: 16358/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_VC1IMAGE_fuzzer-5714436358144000
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
non res_sprite leads to decoder delay which leads to assertion failure
Fixes: Assertion failure
Fixes: 16402/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMV3IMAGE_fuzzer-5704510034411520
Fixes: left shift of 1073741824 by 1 places cannot be represented in type 'int'
Fixes: 16425/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WMV3IMAGE_fuzzer-5692858838810624
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
avio_feof() returns true both in case of actual EOF and in case of IO errors.
Some demuxers (matroska) have special handling to be able to return the proper
error for this exact reason, e.g.:
if (avio_feof(pb)) {
if (pb->error) {
return pb->error;
} else {
return AVERROR_EOF;
}
}
However, most of the demuxers do not, and they simply return AVERROR_EOF if
avio_feof() is true, so there is a real chance that IO errors are mistaken for
EOF.
We might just say that the API user should always check the IO context error
attribute on EOF to make sure no IO errors happened, but not even ffmpeg.c does
this. It should be more intuitive to the API user if we simply return the IO
error as the return value of av_read_frame() instead of AVERROR_EOF.
Signed-off-by: Marton Balint <cus@passwd.hu>
Some security cams generate this, as well as some versions of VirtualDub and
VLC so support for _reading_ such files is justified.
Fixes ticket #7110.
See also this discussion: https://patchwork.ffmpeg.org/patch/8744/
Signed-off-by: Marton Balint <cus@passwd.hu>
After a1b4f120c0 the teletext PTS values were set
to AV_NOPTS_VALUE if the stream of the PCR pid was discarded.
What actually matters is that if we parse the PCR of the PCR PID or not, so
let's use the cached discard value of the actual PCR PID instead of the stream
discard value, which may be different.
Also fixes ticket #7567, which was caused by the fact that teletext PTS values
were not touched if the PCR pid was discarded even before
a1b4f120c0.
Signed-off-by: Marton Balint <cus@passwd.hu>
currently, the layer number is at the beginning of the .model file,
so we have to scan twice in python script, the first scan to get the
layer number. Only one scan needed after put the layer number at the
end of .model file.
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
the info can be saved in dnn operand object without regenerating again and again,
and it is also needed for layer split/merge, and for memory reuse.
to make things step by step, this patch just focuses on c code,
the change within python script will be added later.
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
Many ffmpeg + rpi compilation guides on the internet recommend
using `./configure --enable-omx --enable-omx-rpi`. This fails
to find the IL OMX headers on device because the omx require_headers
check happens first before the add_cflags in omx_rpi.
A workaround is to use `./configure --enable-omx-rpi` only, since
omx_rpi already implies omx. But because many users expect to use
existing scripts and commands, we swap the order here so omx_rpi
special cases are applied first.
In the past this wasn't an issue because users noticed the OMX_Core.h
missing error and installed libomxil-bellagio-dev. But since
76c82843cc, the rpi specific headers from /opt/vc/include/IL
are required.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Fixes: signed integer overflow: -395281576 + -1827578048 cannot be represented in type 'int'
Fixes: 16038/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TTA_fuzzer-5646109705240576
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This makes the code bitexact between platforms.
Intermediate timestamps between frames are preserved.
The timebase is simplified.
Rounding differs from doubles in cases where timestamps/durations
are "funny"
Suggested-by: jb
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Added linux support for amf encoder through vulkan.
To use h.264(AMD VCE) encoder on linux amdgru-pro version 19.20+ and
amf-amdgpu-pro package(amdgru-pro contains, but does not install
automatically) are required.
This driver can be installed using amdgpu-pro-install script in
official amd driver archive.
Initialization of amf encoder occurs in this order:
1) trying to initialize through dx11(only windows)
2) trying to initialize through dx9(only windows)
3) trying to initialize through vulkan
Only Vulkan initialization available on linux.
ff_reget_buffer() will attempt to create a writable copy of the frame,
which is not needed when the decoder intends to return a reference to
the same buffer as the previous frame.
Should reduce data copy, hopefully achieving a similar speed up as
a9dacdeea6 without dropping frames.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
This reverts commit a9dacdeea6.
This patch effectively made the decoder output vfr content out of samples
where cfr is expected.
Addresses ticket #7880.
Signed-off-by: James Almer <jamrial@gmail.com>
wavelets allow significant size expansion and they are also not
very fast.
Fixes: Timeout
Fixes: 16480/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DIRAC_fuzzer-5659892689403904 (108sec -> 17ms)
Fixes: 16480/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DIRAC_fuzzer-5749422717140992 (big ->15sec)
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Tomas Härdin <tjoppen@acc.umu.se>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Add the support of dehaze filter in existing derain filter source
code. As the processing procedure in FFmpeg is the same for current
derain and dehaze, we reuse the derain filter source code. The
model training and generation scripts are in repo
https://github.com/XueweiMeng/derain_filter.git
Reviewed-by: Steven Liu <lq@onvideo.cn>
Signed-off-by: Xuewei Meng <xwmeng96@gmail.com>
vtctx->cached_hw_frames_ctx is unref'd in videotoolbox_uninit,
but videotoolbox_hevc used ff_videotoolbox_uninit which
doesn't unref cache_hw_frames_ctx.
Signed-off-by: Pavel Koshevoy <pkoshevoy@gmail.com>
See https://trac.ffmpeg.org/ticket/7687
If an encoded frame is returned split over two or more
IL buffers due to the size, then there is a race between
whether get_buffer will fail, return NULL, and a truncated
frame is passed on, or IL will return the remaining part
of the encoded frame.
If get_buffer returns NULL, part of the frame is left behind
in the codec, and will be collected on the next call. That
then leaves a frame stuck in the codec. Repeat enough times
and the codec FIFO is full, and the pipeline stalls.
A performance improvement in the Raspberry Pi firmware means
that the timing has changed, and now frequently drops into the
case where get_buffer returns NULL.
Add code such that should a buffer be received without
OMX_BUFFERFLAG_ENDOFFRAME that get_buffer is called with wait
set, so we wait for the remainder of the frame.
This code has been made conditional on the Pi build in case
other IL implementations don't handle ENDOFFRAME correctly.
Signed-off-by: Dave Stevenson <dave.stevenson@raspberrypi.org>
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Martin Storsjö <martin@martin.st>
The packet counting based approach caused excessive sdt/pat/pmt for VBR, so
let's use a timestamp based approach instead similar to how we emit PCRs.
SDT/PAT/PMT period should be consistent for both VBR and CBR from now on.
Also change the type of sdt_period and pat_period to AV_OPT_TYPE_DURATION so no
floating point math is necessary.
Fixes ticket #3714.
Signed-off-by: Marton Balint <cus@passwd.hu>
Fixes: signed integer overflow: 2088795537 + 2147254401 cannot be represented in type 'int'
Fixes: signed integer overflow: -1500363496 + -1295351808 cannot be represented in type 'int'
Fixes: signed integer overflow: -79560 * 32640 cannot be represented in type 'int'
Fixes: signed integer overflow: 2088910005 + 2088796058 cannot be represented in type 'int'
Fixes: signed integer overflow: -117258064 - 2088725225 cannot be represented in type 'int'
Fixes: signed integer overflow: 2088725225 - -117258064 cannot be represented in type 'int'
Fixes: 15739/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ALAC_fuzzer-5630664122040320
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If all tables are skipped it would be impossible to encode any
"non black" video.
Fixes: Timeout (78sec -> 1ms)
Fixes: 15821/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SMACKER_fuzzer-5652598838788096
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This is significantly faster
Fixes: Timeout (1sec after this and the previous commit)
Fixes: 15558/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_PPM_fuzzer-5705273643106304
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
None of the keys we support is that long and other keys
lead to decoder failure. None of the values is expected
to be longer, they are all numbers or short keywords.
This simplifies the code
Fixes: Timeout (9sec->43ms)
Fixes: 15177/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_PAM_fuzzer-5080556716425216
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Clipping was added in 2009 to avoid crashes.
The clipped case would produce a 2nd slice with 0 input
thus also producing 0 output.
Subsequent checks will cause decoder failure unless both
slices have the same output length. thus the only way this
would not already fail is if the output from both slices
was 0 bytes.
Fixes: Timeout (134sec -> 241ms)
Fixes: 15599/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MSZH_fuzzer-5658127116009472
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The dimensions are always 320x200 they are hardcoded in the demuxer.
Hardcode them instead in the decoder.
Fixes: Timeout (16sec -> 400ms)
Fixes: 15574/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_RL2_fuzzer-5158614072819712
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Current default is 256kbps, which produces inconsistent
results (too high for low-res, too low for hi-res).
Use CRF instead, which will adapt.
Signed-off-by: James Zern <jzern@google.com>
Fix#6591
The content has no rbsp_stop_one_bit for ending the SPS, that
causes the decoding SPS failure, results decoding frame failure as well.
The patch is just adding a retry with complete NALU, copied from the retry in decode_nal_unit()
Signed-off-by: James Almer <jamrial@gmail.com>
As of LLVM r368102, Clang will set a pointer tag in bits 56-63 of the
address of a global when compiling with -fsanitize=hwaddress. This requires
an adjustment to assembly code that takes the address of such globals: the
code cannot use the regular R_AARCH64_ADR_PREL_PG_HI21 relocation to refer
to the global, since the tag would take the address out of range. Instead,
the code must use the non-checking (_NC) variant of the relocation (the
link-time check is substituted by a runtime check).
This change makes the necessary adjustment in the movrel macro, where it is
needed when compiling with -fsanitize=hwaddress.
Signed-off-by: Peter Collingbourne <pcc@google.com>
Reviewed-by: Martin Storsjö
Reviewed-by: Janne Grunau
There seems to be no limit in the specification and upto 64k could be stored
512 is choosen as limit as thats the maximum in a conformance sample
An alternative to this patch would be a max_channels variable
Fixes: OOM
Fixes: 16200/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ALS_fuzzer-5764788793114624
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Suggested-by: Thilo Borgmann <thilo.borgmann@mail.de>
Reviewed-by: Thilo Borgmann <thilo.borgmann@mail.de>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Using MSDK parser can improve qsv decoder pass rate in some cases (E.g:
sps declares a wrong level_idc, smaller than it should be).
And it is necessary for adding new qsv decoders such as MJPEG and VP9
since current parser can't provide enough information.
Actually using MFXVideoDECODE_DecodeHeader() was disscussed at
https://ffmpeg.org/pipermail/ffmpeg-devel/2015-July/175734.html and merged as commit 1acb19d,
but was overwritten when merged libav patches (commit: 1f26a23) without any explain.
Split decode header from decode_init, and call it for everyframe to
detect format/resoultion change. It can fix some regression issues such
as hevc 10bits decoding.
Signed-off-by: Zhong Li <zhong.li@intel.com>
Signed-off-by: Dmitry Rogozhkin <dmitry.v.rogozhkin@intel.com>
ff_merge_samplerates will be deallocate a or b in some case, so
add a check before free the format.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
ff_filter_frame fail will free the frame, so we just returen after this
function fail.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
free the old_filname to avoid memory leak in error handle
path.
Reviewed-by: Steven Liu <lq@onvideo.cn>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
background:
DNN (deep neural network) is a sub module of libavfilter, and FATE/dnn
is unit test for the DNN module, one unit test for one dnn layer.
The unit tests are not based on the APIs exported by libavfilter,
they just directly call into the functions within DNN submodule.
There is an issue when run the following command:
build$ ../ffmpeg/configure --disable-static --enable-shared
make
make fate-dnn-layer-pad
And part of error message:
tests/dnn/dnn-layer-pad-test.o: In function `test_with_mode_symmetric':
/work/media/ffmpeg/build/src/tests/dnn/dnn-layer-pad-test.c:73: undefined reference to `dnn_execute_layer_pad'
The root cause is that function dnn_execute_layer_pad is a LOCAL symbol
in libavfilter.so, and so the linker could not find it when build dnn-layer-pad-test.
To check it, just run: readelf -s libavfilter/libavfilter.so | grep dnn
So, add dependency in fate/dnn Makefile with ffmpeg static libraries.
This is the same method used in fate/checkasm
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
if it is initialized randomly, the tensorflow lib will report
error message such as:
Attempt to add output -7920 of depth_to_space4 not in range [0, 1) to node with type Identity
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
write mpegts or fmp4 context into buffer, and flush the buffer into
output file when split fragment. merge two format split workflow into
one workflow
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
show real information instead of the unknown side data type message for HDR10 stream
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
These functions can be used to print a variable number of strings consecutively
to the IO context. Unlike av_bprintf, no temporary buffer is necessary.
Signed-off-by: Marton Balint <cus@passwd.hu>
This fixes make fate issue for frame thread scale in my local testing
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
At the moment scene change detection score uses all planes to detect scene
changes. In this regard this is similar how the frozen frames detection works.
However, in classic encoding scene change detection typically only uses the Y
plane.
We might get more resonable scores for scene change if we also use only
the Y plane for calculating the score if the pixel format is YUV. Although
this will require additional work once packed YUV formats are added,
because for the moment the generic scene sad score calculation has no way
to ignore some components in a packed format.
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
matroska_reset_status (a function that is used during seeking (among
other things)) used an int for the return value of avio_seek which
returns an int64_t. Checking the return value then indicated an error
even though the seek was successfull for targets in the range of
2GB-4GB, 6GB-8GB, ... This error implied that the status hasn't been
reset and in particular, the old level was still considered to be in
force, so that ebml_parse returned errors because the newly parsed
elements were of course not contained in the previously active and still
wrongly considered active master element any more.
Addresses ticket #8084.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
conv2d with dilation > 1 generates tens of nodes in graph, it is not
easy to parse each node one by one, so we do special tricks to parse
the conv2d layer.
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
fix error like "offset 0x1f85: partial file", the root cause is when
read the mp4 file from http, and the moov in the end of the mp4 file,
reconfig the buffer will drop some data.
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
Signed-off-by: Zhongxing Jiang <tomajsjiang@tencent.com>
ff_write_chained essentially duplicated the functionality of
av_packet_rescale_ts. This has been changed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The option tables of the various fftools (in particular ffprobe) are
arrays of OptionDef; said type contains a union of a pointer to void and
a function pointer of type int (*)(void *, const char *, const char *)
as well as a size_t. Some entries (namely the common entry for writing a
report as well as several more of ffprobe's entries) used the pointer to
void to store a pointer to functions of type int (*)(const char *) or
type int (*)(const char *, const char *); nevertheless, when the functions
are actually called in write_option (in cmdutils.c), it is done via a
pointer of the first type.
There are two things wrong here:
1. Pointer to void can be converted to any pointer to incomplete or
object type and back; but they are nevertheless not completely generic
pointers: There is no provision in the C standard that guarantees their
convertibility with function pointers. C90 lacks a generic function
pointer, C99 made every function pointer a generic function pointer and
still disallows the convertibility with void *.
2. The signature of the called function differs from the signature
of the pointed-to type. This is undefined behaviour in C99 (given that
C90 lacks a way to convert function pointers at all, it doesn't say
anything about such a situation). It only works because none of the
functions this patch is about make any use of their parameters at all.
Therefore this commit changes the type of the relevant functions
to match the type used for the call and uses the union's function
pointer to store it. This is legal even in C90.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
PCR generation was based on counting packets for both CBR and VBR streams.
Couting packets might have worked for CBR streams (when muxrate was specified)
but it only took into account the packets of a service (or the packets of the
PCR stream lately), so even that was problematic for multi program streams.
The new code works on actual timestamps for both CBR and VBR streams. For VBR
streams the behaviour of the old code is simulated by selecting a PCR interval
which is the highest multiple of the frame duration but still less than 100 ms.
It should be trivial to add support for setting the PCR interval for VBR
streams as well in a later patch.
The accuracy of PCR packets for CBR streams was greatly improved by preemtively
sending them at PCR intervals even if sending the payload of another stream
is in progress.
This may fix these tickets:
- #5750
- #7524
Signed-off-by: Marton Balint <cus@passwd.hu>
The MPEG-TS muxer had a serious bug related to the use of multiple programs:
in that case, the PCR pid selection was incomplete for all services except one.
This patch solves this problem and selects a stream to become PCR for each
service, preferably the video stream.
This patch also moves pcr calculation attributes to MpegTSWriteStream from
MpegTSService. PCR is a per-stream and not per-service thing, so it was
misleading to refer to it as something that is per-service.
Also remove *service from MpegTSWriteStream because a stream can belong to
multiple services so it was misleading to select one for each stream.
You can check the result with this example command:
./ffmpeg -loglevel verbose -y -f lavfi -i \
"testsrc=s=64x64:d=10,split=2[out0][tmp1];[tmp1]vflip[out1];sine=d=10,asetnsamples=1152[out2]" \
-flags +bitexact -fflags +bitexact -sws_flags +accurate_rnd+bitexact \
-codec:v libx264 -codec:a mp2 -pix_fmt yuv420p \
-map '0:v:0' \
-map '0:v:1' \
-map '0:a:0' \
-program st=0:st=2 -program st=1:st=2 -program st=2 -program st=0 -f mpegts out.ts
You should now see this:
[mpegts @ 0x37505c0] service 1 using PCR in pid=256
[mpegts @ 0x37505c0] service 2 using PCR in pid=257
[mpegts @ 0x37505c0] service 3 using PCR in pid=258
[mpegts @ 0x37505c0] service 4 using PCR in pid=256
Fixes ticket #8039.
v2: a video is stream is preferred if there are no programs, just like before
the patch.
Signed-off-by: Marton Balint <cus@passwd.hu>
98e419cb added SIMD for the convolution filter for x64 systems. As
usual, it used a check of the form
if (ARCH_X86_64)
ff_convolution_init_x86(s);
and thereby relied on the compiler eliminating this pseudo-runtime check
at compiletime for non x64 systems (for which ff_convolution_init_x86
isn't defined) to compile. But vf_convolution.c contains more than one
filter and if the convolution filter is disabled, but one of the other
filters (prewitt, sobel, roberts) is enabled, the build will fail on x64,
because ff_convolution_init_x86 isn't defined in this case.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This patch adds a new option to the scale filter which ensures that the
output resolution is divisible by the given integer when used together
with `force_original_aspect_ratio`. This works similar to using `-n` in
the `w` and `h` options.
This option respects the value set for `force_original_aspect_ratio`,
increasing or decreasing the resolution accordingly.
The use case for this is to set a fixed target resolution using `w` and
`h`, to use the `force_original_aspect_ratio` option to make sure that
the video always fits in the defined bounding box regardless of aspect
ratio, but to also make sure that the calculated output resolution is
divisible by n so in can be encoded with certain encoders/options if
that is required.
Signed-off-by: Lars Kiesow <lkiesow@uos.de>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Changing details as following:
1. Remove the local variable 'out_m' in 'CLIP_SH' and store the result in
source vector.
2. Refine the implementation of macro 'CLIP_SH_0_255' and 'CLIP_SW_0_255'.
Performance of VP8 decoding has speed up about 1.1%(from 7.03x to 7.11x).
Performance of H264 decoding has speed up about 0.5%(from 4.35x to 4.37x).
Performance of Theora decoding has speed up about 0.7%(from 5.79x to 5.83x).
3. Remove redundant macro 'CLIP_SH/Wn_0_255_MAX_SATU' and use 'CLIP_SH/Wn_0_255'
instead, because there are no difference in the effect of this two macros.
Reviewed-by: Shiyou Yin <yinshiyou-hf@loongson.cn>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Up until now, when performing a SEEK_END seek, the subfile protocol
ignored the desired position (relative to EOF) and used the current
absolute offset in the input file instead.
And when performing a SEEK_CUR seek, the current position has been
ignored.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
gcc 6.x and 7.x generate wrong code for little endian machines
for the vec_lvsl/vec_perm instruction combos in some cases.
The bug was fixed in version 8.x
If these instructions are replaced with vec_xl, the problem goes
away for all versions of the compilers.
Fixes ticket #7124.
This should reduces the number of uninteresting timeouts encountered
A single threshold for all codecs did not work
Fixes: 13979/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_QTRLE_fuzzer-5629872380051456 (14sec -> 4sec)
Fixes: 14709/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_GDV_fuzzer-5704215281795072 (179sec -> 7sec)
Fixes: 16296/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_HNM4_VIDEO_fuzzer-5756304521428992 (108sec -> 9sec)
Fixes: 15620/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_GIF_fuzzer-5657214435459072 (26sec -> 26ms)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes the following ubsan errors with the sample from ticket #7980:
libavcodec/zmbvenc.c:243:29: runtime error: left shift of negative value -4
libavcodec/zmbvenc.c:244:28: runtime error: left shift of negative value -2
Otherwise the user might get a silence padded frame in the beginning or in the
middle of the encoding.
Some other bug uncovered this:
./ffmpeg -loglevel verbose -y -f data -i /dev/zero \
-filter_complex "nullsrc=s=60x60:d=10[v0];sine=d=10[a]" \
-map '[v0]' -c:v:0 rawvideo \
-map '[a]' -c:a:0 mp2 \
-f mpegts out.ts
Signed-off-by: Marton Balint <cus@passwd.hu>
This makes the changed code-path faster.
Change not tested except with the fuzzer testcase as I found no other testcase.
Improves: Timeout (136sec -> 74sec)
Improves: 16040/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SCPR_fuzzer-5705876062601216
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
This may break some valid tiff files, it appears the specification does not require
the offsets to be increasing. They increase in the 2 test files i have though except
the last offset which is 0 (an end marker) and for which a special case is added to
avoid asking for a sample for that end marker.
See: [FFmpeg-devel] [PATCH 2/2] avcodec/tiff: Detect infinite retry loop
for an alternative implementation
Fixes: Timeout (Infinite -> Finite)
Fixes: 15706/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TIFF_fuzzer-5114674904825856
This variant was requested by paul on IRC
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This also makes the code consistent with the existing similar MUL64()
in decode_var_block_data()
Fixes: signed integer overflow: -7277630735906765035 + -3272193951413647896 cannot be represented in type 'long'
Fixes: 16015/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ALS_fuzzer-5666552818434048
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 2147483424 - -1772303236 cannot be represented in type 'int'
Fixes: 15708/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ALS_fuzzer-5067890362941440
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
assuming each block contains an empty chunk there has to be at least 8 bytes extra.
Fixes: 15327/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_LSCR_fuzzer-5676669303521280
Fixes: Timeout (11->5sec)
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: left shift of negative value -9
Fixes: 15299/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_MSS2_fuzzer-5660922678345728
Fixes: 15557/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_VC1IMAGE_fuzzer-5673351911047168
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 2147483645 + 4 cannot be represented in type 'int'
Fixes: 15418/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AAC_FIXED_fuzzer-5685269069561856
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The current function will report one error message, but the caller func
haven't check it, so change the default to process as AVCOL_TRC_UNSPECIFIED.
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Signed-off-by: Rick Kern <kernrj@gmail.com>
Fixes the following ubsan error:
libavcodec/r210enc.c:69:28: runtime error: left shift of 522 by 22 places cannot be represented in type 'int'
Fixes ticket #7982.
=== PROBLEM ===
I was trying to record h264 + aac streams from an RTSP server to mp4 file. using this command line:
ffmpeg -v verbose -y -i "rtsp://<ip>/my_resources" -codec copy -bsf:a aac_adtstoasc test.mp4
FFmpeg then fail to record audio and output this logs:
[rtsp @ 0xcda1f0] The profile-level-id field size is invalid (40)
[rtsp @ 0xcda1f0] Error parsing AU headers
...
[rtsp @ 0xcda1f0] Could not find codec parameters for stream 1 (Audio: aac, 48000 Hz, 1 channels): unspecified sample format
In SDP provided by my RTSP server I had this fmtp line:
a=fmtp:98 streamType=5; profile-level-id=40; mode=AAC-hbr; config=1188; sizeLength=13; indexLength=3; indexDeltaLength=3;
In FFmpeg code, I found a check introduced by commit 24130234cd. It disallows values greater than 32 for fmtp line parameters.
RFC-4566 (SDP: Session Description Protocol) do not give any limit of size on interger parameters given in an fmtp line.
However, In RFC-6416 (RTP Payload Format for MPEG-4 Audio/Visual Streams) give examples of "profile-level-id" values for AAC, up to 55.
=== FIX ===
As each parameter may have its own min and max values
I propose to introduce a range for each parameter.
For this patch I used RFC-3640 and ISO/IEC 14496-1 as reference for validity ranges.
This patch fix my problem and I now can record my RTSP AAC stream to mp4.
It has passed the full fate tests suite sucessfully.
Signed-off-by: Olivier Maignial <olivier.maignial@smile.fr>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This patch avoids a read past the end of the input buffer in memcpy since the size
of the received zmq message is recv_buf_size - 1.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Only add sequence end code for mpeg1/mpeg2 video, or else use the encoder
libx264 or libx265 in this sample, decoding the output file will get
unknow NALU type error.
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
Use error level when get invalid sidedata, and remove a unnecessary
newline in error message.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
Add http_seekable option for HTTP partial requests, when The
EXT-X-BYTERANGE tag indicates that a Media Segment is a sub-range
of the resource identified by its URI, we can use HTTP partial
requests to get the Media Segment.
Reviewed-by: Steven Liu <lq@chinaffmpeg.org>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
MSYS2 converts paths to MinGW-based applications from unix to
pseudo-windows paths on execution time.
Since there was no space between '-include' and the path, MSYS2 doesn't
detect the path properly.
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
The previous calculation code did not account for the fact that the
copy_offset for the start of the frame array is at index 0, yet the
scan line number from the rfc4175 RTP header starts at 1.
This caused 2 issues to appear:
- The first scan line was being copied into the array where the second
scan line should be. This caused the resulting video to have a green
line at the top of it.
- Since the packet containing the last scan line would fail the
calculation, the packet with the RTP marker would not be processed
which caused a log message saying "Missed previous RTP marker" to be
outputted for each frame.
Signed-off-by: Jacob Siddall <kobe@live.com.au>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The initialization of the uint32_t plane32_lut matrix uses left shifts
of the form 1 << plane; plane can be as big as 31 which means that this
is undefined behaviour as 1 will be simply an int. So make it unsigned
to avoid this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 1603085316 + 1238786562 cannot be represented in type 'int'
Fixes: 16203/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_RALF_fuzzer-5086088934195200
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
It appears no valid file uses this, so this is not testable with
a valid file.
Fixes: assertion failure
Fixes: 16187/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_VORBIS_fuzzer-5638880618872832
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 write
Fixes: 16105/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_CFHD_fuzzer-5690817309573120
Fixes: 16119/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_CFHD_fuzzer-5099050675732480
Fixes: 16135/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_CFHD_fuzzer-5705501601431552
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: index -1 out of bounds for type 'AV1ReferenceFrameState [8]'
Fixes: 16079/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5758807440883712
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: James Almer <jamrial@gmail.com>
See: [FFmpeg-devel] [PATCH 05/13] avcodec/cbs_av1_syntax_template: Check ref_frame_idx before use
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: left shift of 1073741824 by 1 places cannot be represented in type 'int'
Fixes: signed integer overflow: 2049431315 + 262759074 cannot be represented in type 'int'
Fixes: 16012/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-5719016003338240
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Alternatively the array could be made bigger but the extra values
would not be read without other changes.
Fixes: Out of array access
Fixes: 15658/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ATRAC9_fuzzer-5738260074070016
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Lynne <dev@lynne.ee>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
cuda_runtime.h as well as dynlink_loader.h used nonstandard inclusion
guards with an AV_ prefix, although these files are not in an libav*/
path. So change the inclusion guards and adapt the ref file of the
source fate test accordingly.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
This avoids using the CUDA SDK at all; instead, we provide a minimal
reimplementation of the basic functionality that lavfi actually uses.
It generates very similar code to what NVCC produces.
The header contains no implementation code derived from the SDK.
The function and type declarations are derived from the SDK only to the
extent required to build a compatible implementation. This is generally
accepted to qualify as fair use.
Because this option does not require the proprietary SDK, it does not require
the "--enable-nonfree" flag in configure.
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
Recently we encountered an issue when decoding a h.263 file:
FFmpeg will freeze when decoding h.263 video with NVDEC. Turns out this is not directly related to NVDEC but is a problem that shows with several other HW decoders like VDPAU, though the exact kind of error is different (either error messages or freezing[1]). The root cause is that ff_thread_finish_setup() is called twice per frame from ff_h263_decode_frame(). This is not supported by ff_thread_finish_setup() and specifically checked for and warned against in the functions code. The issue is also specific to hw accelerated decoding only as the second call to ff_thread_finish_setup() is only issued when hw acceleration is on. The fix is simple: add a check that the first call is only send when hw acceleration is off, and the second call only when hw acceleration is on (see attached patch). This works fine as far as I was able to test with vdpau and nvdec/nvcuvid hw decoding. The patch also adds NVDEC to the hw config list if available.
I also noticed a secondary issue when browsing through the code which is that, according to documentation, ff_thread_finish_setup() should only be called if the codec implements update_thread_context(), which h263dec does not. The patch does not address this and I'm not sure any action needs to be taken here at all.
[1] This is depending on whether or not the hw decoder sets the HWACCEL_CAPS_ASYNC_SAFE flag
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
Fixes: signed integer overflow: -1241665686 + -1340629419 cannot be represented in type 'int'
Fixes: 15922/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5692826442006528
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
codecpar->extradata is not going to change between packets. New extradata
is instead propagated using packet side data.
Use ff_alloc_extradata() as well.
Signed-off-by: James Almer <jamrial@gmail.com>
Theora is forced to be non zero if it is zero and a sample
is asked for, as suggested by reimar
Fixes: Timeout (2min -> 600ms)
Fixes: 15366/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_THEORA_fuzzer-5737849938247680
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Simply moves and templates the actual transforms to support an
additional data type.
Unlike the float version, which is equal or better than libfftw3f,
double precision output is bit identical with libfftw3.
Fixes: signed integer overflow: 32796 * 65536 cannot be represented in type 'int'
Fixes: 15430/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_VC1IMAGE_fuzzer-5735424087031808
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: 15524/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TRUEMOTION2_fuzzer-5173148372172800
Fixes: signed integer overflow: 13701388 - -2134868270 cannot be represented in type 'int'
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Changing the type to an unsigned one to avoid many casts was suggested
This may be inadequate for fixing the UB on ILP64
Fixes: signed integer overflow: -1418162611 * 383 cannot be represented in type 'int'
Fixes: 15547/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-5691384901664768
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 755176387 + 1515360583 cannot be represented in type 'int'
Fixes: 15506/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-5706859232624640
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: left shift of negative value -4
Fixes: signed integer overflow: -15091694 * 167 cannot be represented in type 'int'
Fixes: signed integer overflow: 1898547155 + 453967445 cannot be represented in type 'int'
Fixes: 15258/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-5759095564402688
Fixes: signed integer overflow: 962196438 * 31 cannot be represented in type 'int'
Fixes: 15364/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_APE_fuzzer-5718799845687296
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Behave like we do for V4L2_BUF_FLAG_ERROR, implemented in commit 28f20d2ff4 .
For some devices (probably also related to the V4L driver implementation)
it happens that when invoking the ioctl DQBUF, the returned buffer is not
of the expected size. Here are two examples for such occurrences:
[video4linux2,v4l2 @ 0x258b440] Dequeued v4l2 buffer contains 609596 bytes, but 614400 were expected. Flags: 0x00000001.
/dev/video1: Invalid data found when processing input
[video4linux2,v4l2 @ 0x225f440] Dequeued v4l2 buffer contains 609508 bytes, but 614400 were expected. Flags: 0x00000001.
/dev/video1: Invalid data found when processing input
For the ffmpeg CLI tool this means it will stop capturing and exit.
The described behaviour was observed at least with one OmniVision USB
web cam and with some stk1160 devices.
If you search the web for the error message, you will find quite a few
instances of this problem. Some of them experienced on other devices.
Probably fixes ticket #4795
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
There is one way to find out if avpriv_find_start_code has found a start
code or not: One has to check whether the state variable contains a
start code, i.e. whether the three most significant bytes are 0x00 00 01.
Checking for whether the return value is the end of the designated
buffer is not enough: If the last four bytes constitute a start code,
the return value is also the end of the buffer. This happens with
sequence_end_codes which have been ignored for exactly this reason,
although e.g. all three files used for fate tests of cbs_mpeg2 contain
sequence_end_codes.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
1. Currently, cbs_mpeg2_split_fragment uses essentially three variables
to hold the start code values found by avpriv_find_start_code. By
rearranging the code, one of them can be omitted.
2. The return value of avpriv_find_start_code points to the byte after
the byte containing the start code identifier (or to the byte after the
last byte of the fragment's data if no start code was found), but
cbs_mpeg2_split_fragment needs to work with the pointer to the byte
containing the start code identifier; it already did this, but in a
clumsy way. This has been changed.
3. Also use the correct type for the variable holding the
CodedBitstreamUnitType.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Sequence End units (or actually, sequence_end_codes) have up until now
not been decomposed; in fact due to a bug in cbs_mpeg2_split_fragment they
have mostly been treated as part of the preceding unit. So implement
decomposing them as preparation for fixing said bug.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until now, if an SEI messages was to be added to a fragment, it was
tried to add said SEI message to the first SEI NAL unit of the fragment
and if this SEI NAL unit already contained H264_NAL_SEI SEI messages (an
arbitrary limit imposed by cbs_h264), adding failed; if there was no SEI
NAL unit, a new one has been added.
With this commit, the fragment is searched for further NAL units to add
the SEI messages to. If all of them are full, a new SEI NAL unit is added.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
cbs is currently inconsistent regarding the opaque field that can be
used as a special argument to av_buffer_create in order to be used
during freeing the buffer: ff_cbs_alloc_unit_content and all the free
functions used name this parameter as if it should contain a pointer to
the unit whose content is about to be created; but both
ff_cbs_alloc_unit_content as well as ff_cbs_h264_add_sei_message
actually use a pointer to the CodedBitstreamContext as opaque. It should
actually be neither, because it is unneeded (as is evidenced by the fact
that none of the free functions use this pointer at all) and because it
ties the unit's content to the lifetime of other objects, although a
refcounted buffer is supposed to have its own lifetime that only ends
when its reference count reaches zero. This problem manifests itself in
the pointer becoming dangling.
The pointer to the unit can become dangling if another unit is added to
the fragment later as happens in the bitstream filters; in this case,
the pointer can point to the wrong unit (if the fragment's unit array
needn't be relocated) or it can point to where the array was earlier.
It can also become dangling if the unit's content is meant to survive
the resetting of the fragment it was originally read with. This applies
to the extradata of H.264 and HEVC.
The pointer to the context can become dangling if the context is closed
before the content is freed. Although this doesn't seem to happen right
now, it could happen, in particular if one uses different
CodedBitstreamContexts for in- and output.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The minimum header is 8 bytes, the smallest bitstream that is passed to
the MB decode code is 4 bytes
Fixes: Timeout (35sec -> 18sec)
Fixes: 15800/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_EATQI_fuzzer-5684154517159936
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The minimal size which unpack() will not fail on is 5 bytes
Fixes: Timeout (14sec -> 77ms) (testcase 15508)
Fixes: 15508/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_EATGV_fuzzer-5700053513011200
Fixes: 15996/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_EATGV_fuzzer-5751353223151616
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
since tf.pad is enabled, the conv2d(valid) changes back to its original behavior.
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
the reason to add this layer first is that vf_sr uses it in its
tensorflow model, and the next plan is to update the python script
to convert tf.pad into native model.
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
In function ff_h264_add_pixels4_8_mmi, there is no need to reset '%[ftmp0]'
to 0, because it's value has never changed since the start of the asm block.
This patch remove the redundant 'xor' and set src to zero once it was loaded.
In function ff_h264_idct_add_8_mmi, 'block' is seted to zero twice.
This patch removed the first setting zero operation and move the second one
after the load operation of block.
In function ff_h264_idct8_add_8_mmi, 'block' is seted to zero twice too.
This patch just removed the second setting zero operation.
This patch mainly simplifies the implementation of functions above,
the effect on the performance of whole h264 decoding process is not obvious.
According to the perf data, proportion of ff_h264_idct_add_8_mmi decreased from
0.29% to 0.26% and ff_h264_idct8_add_8_mmi decreased from 0.62% to 0.59% when decoding
H264 format on loongson 3A3000(For reference only , not very stable.).
Reviewed-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The earlier version had three deficits:
1. It allowed to set the stream to RGB although this is not allowed when
the profile is 0 or 2.
2. If it set the stream to RGB, then it did not automatically set the
range to full range; the result was that one got a warning every time a
frame with color_config element was processed if the frame originally
had TV range and the user didn't explicitly choose PC range. Now one
gets only one warning in such a situation.
3. Intra-only frames in profile 0 are automatically BT.601, but if the
user wished another color space, he was not informed about his wishes
being unfulfillable.
The commit also improves the documentation about this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
If the relevant elements (the color description elements for AV1 and the
VUI elements in general for H.264 (since 1156b507)) are absent, then their
correct values (usually meaning unknown) have already been inferred by
the reading process, so that it is unnecessary to initialize them again
in the av1/h264_metadata filters even when they were initially absent.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
1. The extra information in slice headers was parsed incorrectly:
In the first reading pass to derive the length of the extra information,
one should look at bits n, n + 9, n + 18, ... and check whether they
equal one (further extra information) or zero (end of extra information),
but instead bits n, n + 8, n + 16, ... were inspected. The second pass
of reading (where the length is already known and the bytes between the
length-determining bits are copied into a buffer) did not record what
was in bits n, n + 9, n + 18, ..., presuming they equal one. And during
writing, the bytes in the buffer are interleaved with set bits and
written. This means that if the detected length of the extra information
was greater than the real length, the output was corrupted. Fortunately
no sample is known that made use of this mechanism: The extra information
in slices is still marked as reserved in the specifications. cbs_mpeg2
is now ready in case this changes.
2. Furthermore, the buffer is now padded and slightly different, but
very similar code for reading resp. writing has been replaced by code
used for both. This was made possible by a new macro, the equivalent
to cbs_h2645's fixed().
3. These changes also made it possible to remove the extra_bit_slice
element from the MPEG2RawSliceHeader structure. Said element was always
zero except when the detected length of the extra information was less
than the real length.
4. The extra information in picture headers (which uses essentially the
same syntax as the extra information in slice headers) has simply been
forgotten. This meant that if this extra information was present, it was
discarded during reading; and unfortunately writing created invalid
bitstreams in this case (an extra_bit_picture - the last set bit of the
whole unit - indicated that there would be a further byte of data,
although the output didn't contain said data).
This has been fixed; both types of extra information are now parsed via
the same code and essentially passed through.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until now, a temporary variable was used and initialized every time a
value was read in CBS; if reading turned out to be successfull, this
value was overwritten (without having ever been looked at) with the
value read if reading was successfull; on failure the variable wasn't
touched either. Therefore these initializations can be and have been
removed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
If a sequence display extension is read with colour_description equal to
zero, but a user wants to add one or more of the colour_description
elements, then the colour_description elements the user did not explicitly
request to be set are set to zero and not to the value equal to
unknown/unspecified (namely 2). A value of zero is not only inappropriate,
but explicitly forbidden. This is fixed by inferring the right default
values during the reading process if the elements are absent; moreover,
changing any of the colour_description elements to zero is now no longer
possible.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Fixes: null pointer passed as argument 2, which is declared to never be null
Fixes: 16008/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SSA_fuzzer-5650582821404672 (this is a separate issue found in this testcase)
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
An incomplete image is not supported prior to this and will
not produce any output. This commit moves the failure before
time consuming operations.
Fixes: Timeout (81sec -> 76ms)
Fixes: 15723/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_BRENDER_PIX_fuzzer-5147265653538816
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
it is expected that there will be more files to support native mode,
so put all the dnn codes under libavfilter/dnn
The main change of this patch is to move the file location, see below:
modified: libavfilter/Makefile
new file: libavfilter/dnn/Makefile
renamed: libavfilter/dnn_backend_native.c -> libavfilter/dnn/dnn_backend_native.c
renamed: libavfilter/dnn_backend_native.h -> libavfilter/dnn/dnn_backend_native.h
renamed: libavfilter/dnn_backend_tf.c -> libavfilter/dnn/dnn_backend_tf.c
renamed: libavfilter/dnn_backend_tf.h -> libavfilter/dnn/dnn_backend_tf.h
renamed: libavfilter/dnn_interface.c -> libavfilter/dnn/dnn_interface.c
Signed-off-by: Guo, Yejun <yejun.guo@intel.com>
Signed-off-by: Pedro Arthur <bygrandao@gmail.com>
avio_read can successfully return even when less than the requested
amount of input was read. wavdec's bext parsing mistakenly assumed a
successful avio_read always read the full amount that was requested.
The result could be dictionary tags populated with partially
uninitialized values.
This change also fixes a broken assertion in wav_parse_bext_string that
was off-by-one, though no known current usage of that method hits that
broken case.
Chromium bug: 987270
Signed-off-by: Matt Wolenetz <wolenetz@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Replace the same code logic with ensure_playlist(), it's will
help reusable blocks of code.
Reviewed-by: Jun Zhao <barryjzhao@tencent.com>
Signed-off-by: vacingfang <vacingfang@tencent.com>
av_packet_unref have reseted the AVPacket, so don't need to call
reset_packet after that.
Reviewed-by: Steven Liu <lq@chinaffmpeg.org>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
mpc8_q4_syms is an array of int8_t that is initialized using
values not in the range of an int8_t and that is only accessed via
a pointer to uint8_t in ff_init_vlc_sparse. The latter applies to all
the other *_bits and *_syms tables in mpc8huff.h, so make them all
unsigned.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Some of the VP3 arrays (namely vp31_intra_y_dequant, vp31_intra_c_dequant
and vp31_inter_dequant) are currently declared as array of (const) int8_t
despite them being only used to directly initialize an array of uint8_t.
vp31_inter_dequant even contains the value 128 which is not
representible in int8_t and might generate overflow warnings by
compilers.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
why change .4 to .25, it's for:
one scenecut(pkt_pts=20040) isn't detected by 0.4 threshold
why not change to 0.3 instead of 0.25:
it will miss the scenecut(pkt_pts=20040) after applying the next
patch which enables yuvj420
for fate testing, it's better to catch all scenecut scenes.
Reviewed-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
This patch does not make other pixel formats usable yet to make sure the test
result is the same with rgb32 format.
Reviewed-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
A value of zero for sample_count in trun box is not
prohibited by 14496-12 section 8.8.8. 4a9d32baca
disallowed this which led the demuxer to error out
when reading the header of valid files.
@@ -105,12 +105,14 @@ It takes a single signed native-endian 16-bit raw audio stream of at most 2 chan
@table @option
@item silence_threshold
Threshold for detecting silence, ranges from -1 to 32767. -1 disables silence detection and
is required for use with the AcoustID service. Default is -1.
Threshold for detecting silence. Range is from -1 to 32767, where -1 disables
silence detection. Silence detection can only be used with version 3 of the
algorithm.
Silence detection must be disabled for use with the AcoustID service. Default is -1.
@item algorithm
Version of algorithm to fingerprint with. Range is 0 to 4. Version 2 requires that silence
detection be enabled. Default is 1.
Version of algorithm to fingerprint with. Range is 0 to 4.
Version 3 enables silence detection. Default is 1.
@item fp_format
Format to output the fingerprint as. Accepts the following options:
@@ -234,8 +236,10 @@ This is a deprecated option to set the segment length in microseconds, use @var{
@item seg_duration @var{duration}
Set the segment length in seconds (fractional value can be set). The value is
treated as average segment duration when @var{use_template} is enabled and
@var{use_timeline} is disabled and as minimum segment duration for all the other
use cases.
@item frag_duration @var{duration}
Set the length in seconds of fragments within segments (fractional value can be set).
@item frag_type @var{type}
Set the type of interval for fragmentation.
@item window_size @var{size}
Set the maximum number of segments kept in the manifest.
@item extra_window_size @var{size}
@@ -275,6 +279,15 @@ of the adaptation sets and a,b,c,d and e are the indices of the mapped streams.
To map all video (or audio) streams to an AdaptationSet, "v" (or "a") can be used as stream identifier instead of IDs.
When no assignment is defined, this defaults to an AdaptationSet for each stream.
Optional syntax is "id=x,seg_duration=x,frag_duration=x,frag_type=type,descriptor=descriptor_string,streams=a,b,c id=y,seg_duration=y,frag_type=type,streams=d,e" and so on,
descriptor is useful to the scheme defined by ISO/IEC 23009-1:2014/Amd.2:2015.
For example, -adaptation_sets "id=0,descriptor=<SupplementalProperty schemeIdUri=\"urn:mpeg:dash:srd:2014\" value=\"0,0,0,1,1,2,2\"/>,streams=v".
Please note that descriptor string should be a self-closing xml tag.
seg_duration, frag_duration and frag_type override the global option values for each adaptation set.
For example, -adaptation_sets "id=0,seg_duration=2,frag_duration=1,frag_type=duration,streams=v id=1,seg_duration=2,frag_type=none,streams=a"
type_id marks an adaptation set as containing streams meant to be used for Trick Mode for the referenced adaptation set.
For example, -adaptation_sets "id=0,seg_duration=2,frag_type=none,streams=0 id=1,seg_duration=10,frag_type=none,trick_id=0,streams=1"
@item timeout @var{timeout}
Set timeout for socket I/O operations. Applicable only for HTTP output.
@item index_correction @var{index_correction}
@@ -320,9 +333,37 @@ This option will also try to comply with the above open spec, till Apple's spec
Applicable only when @var{streaming} and @var{hls_playlist} options are enabled.
This is an experimental feature.
@item ldash @var{ldash}
Enable Low-latency Dash by constraining the presence and values of some elements.
Publish master playlist repeatedly every after specified number of segment intervals.
@item write_prft @var{write_prft}
Write Producer Reference Time elements on supported streams. This also enables writing
prft boxes in the underlying muxer. Applicable only when the @var{utc_url} option is enabled.
It's set to auto by default, in which case the muxer will attempt to enable it only in modes
that require it.
@item mpd_profile @var{mpd_profile}
Set one or more manifest profiles.
@item http_opts @var{http_opts}
A :-separated list of key=value options to pass to the underlying HTTP
protocol. Applicable only for HTTP output.
@item target_latency @var{target_latency}
Set an intended target latency in seconds (fractional value can be set) for serving. Applicable only when @var{streaming} and @var{write_prft} options are enabled.
This is an informative fields clients can use to measure the latency of the service.
@item min_playback_rate @var{min_playback_rate}
Set the minimum playback rate indicated as appropriate for the purposes of automatically
adjusting playback latency and buffer occupancy during normal playback by clients.
@item max_playback_rate @var{max_playback_rate}
Set the maximum playback rate indicated as appropriate for the purposes of automatically
adjusting playback latency and buffer occupancy during normal playback by clients.
@end table
@anchor{framecrc}
@@ -607,6 +648,9 @@ Set the starting sequence numbers according to @var{start_number} option value.
@item epoch
The start number will be the seconds since epoch (1970-01-01 00:00:00)
@item epoch_us
The start number will be the microseconds since epoch (1970-01-01 00:00:00)
@item datetime
The start number will be based on the current date/time as YYYYmmddHHMMSS. e.g. 20161231235759.
@@ -803,6 +847,9 @@ fmp4 files may be used in HLS version 7 and above.
@item hls_fmp4_init_filename @var{filename}
Set filename to the fragment files header file, default filename is @file{init.mp4}.
@item hls_fmp4_init_resend
Resend init file after m3u8 file refresh every time, default is @var{0}.
When @code{var_stream_map} is set with two or more variant streams, the
@var{filename} pattern must contain the string "%v", this string specifies
the position of variant stream index in the generated init file names.
@@ -898,8 +945,8 @@ serving up segments can be configured to reject requests to *.tmp to prevent acc
before they have been added to the m3u8 playlist. This flag also affects how m3u8 playlist files are created.
If this flag is set, all playlist files will written into temporary file and renamed after they are complete, similarly as segments are handled.
But playlists with @code{file} protocol and with type (@code{hls_playlist_type}) other than @code{vod}
are always written into temporary file regardles of this flag. Master playlist files (@code{master_pl_name}), if any, with @code{file} protocol,
are always written into temporary file regardles of this flag if @code{master_pl_publish_rate} value is other than zero.
are always written into temporary file regardless of this flag. Master playlist files (@code{master_pl_name}), if any, with @code{file} protocol,
are always written into temporary file regardless of this flag if @code{master_pl_publish_rate} value is other than zero.
@end table
@@ -1029,6 +1076,21 @@ have and language is named ENG, the other audio language is named CHN.
By default, a single hls variant containing all the encoded streams is created.
av_log(NULL, AV_LOG_WARNING, "Multiple %s options specified for stream %d, only the last option '-%s%s%s "SPECIFIER_OPT_FMT_##type"' will be used.\n",\
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