* commit '832a202c47a246ed15e3edc6b05dfcfa7d82c4b2':
protocols: make the list of protocols static
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This commit also disables the async fate test, because it
used internal APIs in a non-kosher way, which no longer
exists.
* commit '2758cdedfb7ac61f8b5e4861f99218b6fd43491d':
lavf: reorganize URLProtocols
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* commit 'e192cd9ce2b51c2e6919f2a78b1ce53e0024e728':
smoothstreamingenc: do not open the files as read+write
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* commit '7fbb3b5b9857276b4cd17b2a530c7e0880d2bc0a':
lavf: use the io_open callbacks for files opened from open_input() as well
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
FATE tests have been updated to patch. They do not differ in
any meaningful way.
* commit 'dc6527ed908e4d330738f139074455ffbe56a2de':
nutenc: do not use AVCodecContext.frame_size
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This also fixes reading gapless metadata when the entries do not start with the
mean atom. Such samples can be found here:
https://hydrogenaud.io/index.php/topic,93310.0.html
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Marton Balint <cus@passwd.hu>
https://developer.apple.com/library/mac/technotes/tn2174/_index.html
- Enabled creation of timecode tracks for MP4 in the same way as MOV.
- Used nmhd as media information header of timecode track of MP4 instead
of gmhd used in MOV, thus avoiding tcmi also, as recommended above.
- Bypassed adding source reference field for MP4, as suggested above.
Issue: https://trac.ffmpeg.org/ticket/4704
Signed-off-by: Syed Andaleeb Roomy <andaleebcse@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This broke packed_maindata.mp3.mp4
Its unknown to me what this commit would have fixed
Reviewed-by: James Almer <jamrial@gmail.com>
This reverts commit 79127dbbef, reversing
changes made to 9fad1ce7c9.
* commit '0d1229f1d2b8f26dd50c6be7917bb8ed8cb95364':
voc: Split ff_voc_get_packet into a separate file
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* commit 'b92962436bdcfae478c8598dca397a397762eef8':
mov: Fix the format specifier type for size
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This allows to copy information related to the stream ID from the demuxer
to the muxer, thus allowing for example to retain information related to
synchronous and asynchronous KLV data packets. This information is used
in the muxer when remuxing to distinguish the two kind of packets (if the
information is lacking, data packets are considered synchronous).
The fate reference changes are due to the use of
av_packet_merge_side_data(), which increases the size of the output
packet size, since side data is merged into the packet data.
Currently, AVStream contains an embedded AVCodecContext instance, which
is used by demuxers to export stream parameters to the caller and by
muxers to receive stream parameters from the caller. It is also used
internally as the codec context that is passed to parsers.
In addition, it is also widely used by the callers as the decoding (when
demuxer) or encoding (when muxing) context, though this has been
officially discouraged since Libav 11.
There are multiple important problems with this approach:
- the fields in AVCodecContext are in general one of
* stream parameters
* codec options
* codec state
However, it's not clear which ones are which. It is consequently
unclear which fields are a demuxer allowed to set or a muxer allowed to
read. This leads to erratic behaviour depending on whether decoding or
encoding is being performed or not (and whether it uses the AVStream
embedded codec context).
- various synchronization issues arising from the fact that the same
context is used by several different APIs (muxers/demuxers,
parsers, bitstream filters and encoders/decoders) simultaneously, with
there being no clear rules for who can modify what and the different
processes being typically delayed with respect to each other.
- avformat_find_stream_info() making it necessary to support opening
and closing a single codec context multiple times, thus
complicating the semantics of freeing various allocated objects in the
codec context.
Those problems are resolved by replacing the AVStream embedded codec
context with a newly added AVCodecParameters instance, which stores only
the stream parameters exported by the demuxers or read by the muxers.
Instead of a linked list constructed at av_register_all(), store them
in a constant array of pointers.
Since no registration is necessary now, this removes some global state
from lavf. This will also allow the urlprotocol layer caller to limit
the available protocols in a simple and flexible way in the following
commits.
Improves streaming compatibility with Windows Media Services. Also tested for
compatilbility in Windows Media Player, Windows Media ASF Viewer and VLC.
This version of the patch only writes exclusion among audio streams, therefore
choosing a subtitle language should be possible independently of audio language.
Signed-off-by: Marton Balint <cus@passwd.hu>
I discovered that ffserver streaming was broken (it seems like it has been since 20th November) and I opened a ticket for this (https://trac.ffmpeg.org/ticket/5250 <https://trac.ffmpeg.org/ticket/5250>).
I spent yesterday learning git bisect (with the kind help of cehoyos) to painstakingly track down the cause. This was made more difficult due to the presence of a segfault in ffserver during the period where the bug was introduced so I first had to identify when and how that was fixed and then retrospectively apply that fix again for each step of the second git bisect to find the actual bug.
Anyway, the fruits of my labour are the innocent looking patch below to correct a couple of typos and define a valid range for two variables.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fix cases where unknown data (data beyond p->buf_size) could produce a
higher ico probe score than if the unknown data was known and valid.
For example:
Header: OK, 2 frames
Frame 0: Unknown (offset points beyond end of probe buffer)
Frame 1: Invalid
Previously this example had a score of 25, even though the score would
be 1 if the unknown frame was known to be valid or 0 if it was known
to be invalid. For this example the score is now 1.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Some muxer might or might not fit incomplete mp3 frames in
their packets.
Bug-Id: 899
CC: libav-stable@libav.org
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This can be made more efficient, but first and the main goal of this change is to
store it at all
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
It is only used in a boolean context. Also clarify its documentation.
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Use with -use_localtime, and set -hls_segment_filename to a path which
contains a subdirectory i.e. /some/path/%Y%m%d/%Y%m%dT%H%M%S-%s.ts
This will mkdir the %Y%m%d-part of the path if it does not already
exist.
In addition, each filename in the playlist output will be prefixed with
this subdirectory (if playlist and segment shares the same base path).
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This commit is a no-op.
* commit '5eb562831b3a9bea8026c413ef1338e06450d005':
mov: Use the correct type for size
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This avoid "libavformat/genh.c:43:14: warning: variable coef_splitted set but not used"
Fewer warnings makes it easier to see new and important warnings
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Currently it is broken when explicitly using the file protocol, it uses an
insecure temporary file name, and in commit b4431c80 disabling the option by
default was already considered. Also it is not very consistent to have such an
option for one particular muxer.
Signed-off-by: Marton Balint <cus@passwd.hu>
This also deprecates our old duplicated callbacks.
* commit '9f61abc8111c7c43f49ca012e957a108b9cc7610':
lavf: allow custom IO for all files
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
The UUID is based on http://www.adobe.com/devnet/xmp.html
The patch is made according to XMP SPECIFICATION PART 3 - STORAGE IN
FILES See Table 8
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This fixes infinite loops due to seeking back.
Signed-off-by: Alexandra Hájková <alexandra@khirnov.net>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This fixes infinite loops due to seeking back.
Signed-off-by: Alexandra Hájková <alexandra@khirnov.net>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The loop can be very long, even though the file is very short.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Alexandra Hájková <alexandra@khirnov.net>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
asf_read_payload can unset eof_reached, so check it also before calling
that function.
This fixes infinite loops.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Alexandra Hájková <alexandra@khirnov.net>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Silences: CID1351343
The header is calculated by the code above the changed hunk, it is
thus asserted that the header is always correct.
Reviewed-by: "Ronald S. Bultje" <rsbultje@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This can provide a manual workaround for ticket #4230.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Marton Balint <cus@passwd.hu>
Checked compatiblity with VLC, Windows Media Player 12 and Windows Media ASF
Viewer 9 series.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Marton Balint <cus@passwd.hu>
Note to maintainers: update tools
Note to maintainers: set a default whitelist for your protocol
If that makes no sense then consider to set "none" and thus require the user to specify a white-list
for sub-protocols to be opened
Note, testing and checking for missing changes is needed
Reviewed-by: Andreas Cadhalpun <andreas.cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This reverts commit 8ed82d8174.
SMPTE S377-1-2009c defines in F.4.1 that the Video Line Map should
always be an array with two 32 bit integers as elements. This is
repeated in G.2.12 with actual examples for progressive content,
where the second value would always be 0.
Additionally, the IRT MXF analyser also lists this as the only
error in the MXF output from ffmpeg: https://mxf-analyser-cloud.irt.de
Reviewed-by: Tomas Härdin <tomas.hardin@codemill.se>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This can cause problems with urls that have arguments after the filename
This reverts commit b0c57206d5.
Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Due to this bug in VC++ 2015 Update 1:
https://connect.microsoft.com/VisualStudio/feedback/details/2291638
the 'key' array in ff_read_riff_info() ends up being not null
terminated which led to failures in a Chromium unit tests. Update 2
should have a fix, but until then it is important to avoid problems.
The index creation is O(N^2) with number of entries (typically thousands).
On a Pi this can take more than 60 seconds to execute for a recording of a few hours.
By replacing with an O(N) loop, this takes virtually zero time
Liked-by: Paul B Mahol <onemda@gmail.com>
Reviewed-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Decodes YUV 4:2:2 10-bit and RGB 12-bit files.
Older files with more subbands, skips, Bayer, alpha not supported.
Alpha requires addition of GBRAP12 pixel format.
Some muxer use the FLV field PreviousTagSize to be the sum of tag
length. Without this change, the flv demuxer think the file is broken
and the re-sync will fail.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This option can force the segmenter to only start a new segment if a packet
reaches the muxer within the specified duration after the segmenting clock
time, which makes it more resilient to backward local time jumps, such as leap
seconds or transition to standard time from daylight savings time.
Reviewed-by: Stefano Sabatini <stefasab@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
Libav, for some reason, merged this as a public API function. This will
aid in future merges.
A define is left for backwards compat, just in case some person
used it, since it is in a public header.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
commit "avpacket: Deprecate av_dup_packet" broke the use
av_interleaved_write_uncoded_frame as any input uncoded frame has an
invalid packet size that will crash when av_packet_ref tries to allocate
'size' new memory. Since the packet is a temporary created within mux.c
itself it can be used directly without needing a new ref.
Signed-off-by: Matt Oliver <protogonoi@gmail.com>
The purpose of this patch is to preserve timestamps when using ffmpeg for publishing RTMP streams, e.g. ffmpeg -i rtmp://source/stream -f flv rtmp://target/stream.
There is a setting "copyts" for that purpose. Unfortunately it doesn't work with FLV muxer because it has its own timestamp correction which makes global setting "copyts" ineffective.
This patch removes timestamp correction in FLV muxer. This means FLV will rely on ffmpeg timestamp correction which makes it possible to use copyts.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Possibly the check as a whole causes more problems than it helps, if so dont
hesitate to remove it
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Some (de)muxers open additional files beyond the main IO context.
Currently, they call avio_open() directly, which prevents the caller
from using custom IO for such streams.
This commit adds callbacks to AVFormatContext that default to
avio_open2()/avio_close(), but can be overridden by the caller. All
muxers and demuxers using AVIO are switched to using those callbacks
instead of calling avio_open()/avio_close() directly.
(de)muxers that use the URLProtocol layer directly instead of AVIO
remain unconverted for now. This should be fixed in later commits.
adds two new options that may be set via the dictionary:
- send_buffer_size
- recv_buffer_size
When present, setsockopt() is used with SO_SNDBUF and SO_RCVBUF to set
socket buffer sizes. I chose to make send and receive independent
because buffering requirements are often asymmetric.
Errors in setting the buffer size mean the socket will use its
default, so they are ignored.
There is no sanity checking on values, as the kernel/socket layers
already impose reasonable limits if asked for something crazy.
Rationale for enlarging receive buffers is to reduce susceptibility
to intermittent network delays/congestion. I added setting the send
buffer for symmetry.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This feature is mostly only used by NLE software, and is
both of dubious value being enabled by default, and a
possible security risk.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This feature is not know much or used much AFAIK, and it might be helpfull in
exploits.
No specific case is known where it can be used in an exploit though
subsequent commits depend on this commit though
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
With this, user applications which use custom IO and have set a IO context will not have
their already opened IO context ignored and glob/seq being interpreted
Comments and tests from maintainers of user apps are welcome!
Liked-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This feature is mostly only used by NLE software, and is
both of dubious value being enabled by default, and a
possible security risk.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
For http, this avoids spurious warnings about failed requests (e.g.
HTTP error 416 Requested Range Not Satisfiable), if the last packet
is truncated and the size read is bogus.
Signed-off-by: Martin Storsjö <martin@martin.st>
When loading a truncated flv file, it would previously try to do a seek to
the end of every packet read. For some input protocols (such as http), such
repeated seek attempts are cripple the reading performance.
Signed-off-by: Martin Storsjö <martin@martin.st>
Confine the 0x00000000 to 'raw '/'twos' fourcc mapping to old version 0
sound sample descriptions, since they are the only valid sample
descriptions for this type of mapping.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This fixes infinite loops due to seeking back.
Reviewed-by: Alexandra Hájková <alexandra.khirnova@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
asf_read_payload can unset eof_reached, so check it also before calling
that function.
This fixes infinite loops.
Reviewed-by: Alexandra Hájková <alexandra.khirnova@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
This fixes infinite loops due to seeking back.
Reviewed-by: Alexandra Hájková <alexandra.khirnova@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
This fixes infinite loops.
Reviewed-by: Alexandra Hájková <alexandra.khirnova@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
The loop can be very long, even though the file is very short.
Reviewed-by: Alexandra Hájková <alexandra.khirnova@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Both avio_skip and detect_unknown_subobject use int64_t for the size
parameter.
This fixes a segmentation fault due to infinite recursion.
Reviewed-by: Alexandra Hájková <alexandra.khirnova@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Otherwise invalid values are used unchecked in the next run.
This can cause NULL pointer dereferencing.
Reviewed-by: Alexandra Hájková <alexandra.khirnova@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Fixes out of array access
Fixes: asan_heap-oob_445b39_1741_d00eb645ab48eb2203b4a04a5b997103.ivr
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes runtime error: null pointer passed as argument 2, which is
declared to never be null
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Fixes out of array read
Fixes: 03c406ec9530e594a074ce2979f8a1f0/asan_heap-oob_7dec26_4664_37c52495b2870a2eaac65f53958e76c1.flac
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Use av_realloc() rather than av_malloc() when normalizing noncompliant
private data in get_qt_codec().
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Either disabling or init'ing secure memory is required after the use
of gcry_check_version. From a look at the functions rtmpdh uses, I
noticed none require the use of secure memory, so we disable it [1][2].
This resolves some errors returned by rtmpdh code with uninitialized
gcrypt, especifically:
Fatal: failed to create the RNG lock: Invalid argument
FATAL: failed to acquire the FSM lock in libgrypt: Invalid argument
Version "1.5.4" was arbitrarily chosen. An older version probably works
as well, but I couldn't compile older versions to test on my machine.
[1]
https://gnupg.org/documentation/manuals/gcrypt/Initializing-the-library.html
[2]
https://www.gnupg.org/documentation/manuals/gcrypt/Controlling-the-library.html
Signed-off-by: Ricardo Constantino <wiiaboo@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
It is used to store the difference between pointers, so ptrdiff_t is the
correct type.
This prevents potential overflows.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
This fixes segmentation faults due to out of bounds writes, when
color_start is interpreted as negative number.
This regression was introduced in commit 57631f.
Reviewed-by: Mats Peterson <matsp888@yahoo.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
This patch adds a new static function get_qt_codec() that takes care of
the initial retrieval of the fourcc and codec ID for A_QUICKTIME and
V_QUICKTIME. It also normalizes noncompliant private data found in some
older files that incorrectly starts with the fourcc by expanding/shifting
the data by 4 bytes, and storing the data size at the start. This is
necessary in order for the rest of the code in the A_QUICKTIME and
V_QUICKTIME blocks (and most likely other code as well) to correctly
parse the private data.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
In many older QuickTime files, the audio format, or "fourcc", is
0x00000000. The QuickTime File Format Specification states the following
regarding this situation:
"This format descriptor should not be used, but may be found in some
files. Samples are assumed to be stored in either 'raw ' or 'twos'
format, depending on the sample size field in the sound description."
MPlayer handles this logic by itself, but FFmpeg/FFplay currently does
not.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
In many older QuickTime files, the audio format, or "fourcc", is
0x00000000. The QuickTime File Format Specification states the following
regarding this situation:
"This format descriptor should not be used, but may be found in some
files. Samples are assumed to be stored in either 'raw ' or 'twos'
format, depending on the sample size field in the sound description."
MPlayer handles this logic by itself, but FFmpeg/FFplay currently does
not.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Both avio_skip and detect_unknown_subobject use int64_t for the size
parameter.
This fixes a segmentation fault due to infinite recursion.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Alexandra Hájková <alexandra.khirnova@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Otherwise invalid values are used unchecked in the next run.
This can cause NULL pointer dereferencing.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Alexandra Hájková <alexandra.khirnova@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This fixes NULL pointer dereferencing if the codec is forced to
adpcm_thp even though a different one was detected.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
This fixes NULL pointer dereferencing.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
The QuickTime File Format Specification states the following:
"Depth: A 16-bit integer that indicates the pixel depth of the
compressed image. Values of 1, 2, 4, 8 ,16, 24, and 32 indicate the
depth of color images. The value 32 should be used only if the image
contains an alpha channel. Values of 34, 36, and 40 indicate 2-, 4-, and
8-bit grayscale, respectively, for grayscale images."
There is no mention of value 33, i.e. 1-bit video (0x01) with the
greyscale bit (0x20) set. I therefore suggest that we ignore the
greyscale bit when processing 1-bit video. Another reason to do this is
that the first 1-bit sample file below will be displayed properly with
blue colors in QuickTime in Windows or Mac *in spite of* the greyscale
bit being set.
Also, QuickTime in Windows or Mac ignores the greyscale bit if the
video sample description contains a palette, regardless of bit depth.
This is undocumented behaviour, but I think we should do the same, and
it seems pretty logical after all, since one wouldn't really bother
putting a customized palette into a grayscale file anyway. See the
second 8-bit sample file below, which has the greyscale bit set, and
which contains a palette in the video sample description. In Windows or
Mac, it will be displayed with the palette in the sample description, in
spite of the greyscale bit being set.
Sample file 1 (1-bit QuickTime Animation):
https://drive.google.com/open?id=0B3_pEBoLs0faTThSek1EeXQ0ZHM
Earth Spin 1-bit qtrle orig.mov
Sample file 2 (8-bit QuickTime Animation):
https://drive.google.com/open?id=0B3_pEBoLs0fad2s0V1YzUWo5aDA
quiz-palette+gs.mov
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This fixes segmentation faults caused by passing a packet_ptr of NULL to
memcpy.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
This preempts potential bugs if this is changed and the indention
ends up different from C interpretation
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This fixes ubsan runtime error: left shift by 8 places cannot be
represented in type 'int'
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
* commit '69a68593ce5684409c3c4dd9a901bfd8b16925b1':
Remove stray line breaks from avpriv_{report_missing_feature|request_samples}
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
Also use a unsigned constant for the shift calculation, as 1 << 31 is
undefined for int32_t. This is also fixed oggparsetheora.
This fixes ubsan runtime error: shift exponent is too large for
32-bit type 'int'
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Fixes Ticket #5032
The samples in Ticket #5032 is using \r\r\n as line breaks. Since we
already are handling \r, or \n, or \r\n as line breaks, \r\n\n will be
considered as a double line breaks. This is an issue because
ff_subtitles_read_text_chunk() will as a result stop extracting a chunk
after just one line.
So instead of parsing the SRT by "chunks" (which means splitting every
double LB), this new parser is detecting timing lines, and split the
events on this basis. While this sounds safe and simple, it needs to
take into account the event number preceding the timing line while
handling situations such as:
- event number starting at 0 or actually any number instead of 1
- event numbers not being ordered at all
- event number being followed by text garbage (this really happened,
see Ticket #4898)
- event payload containing one or multiple number (a protagonist saying
a count-down, a date or whatever) which could be confused with a
chapter number
- event number being empty (see Ticket #2167)
- all kind of weird line breaks can appear randomly like wild pokémons
- untrustable line breaks (Ticket #5032)
The sample madness.srt tries to sum up most of this into one sample,
ticket5032-rrn.srt is the file containing \r\r\n line breaks. and
empty-events-2167.srt contains empty events.
* commit '64f8c439fd663fec4d57ac21af572d498fe21f7a':
rtmpproto: Include the full path as app when "slist=" is found
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* commit '2d0432d918a71468419b7ac1e543ab3b399d3d37':
vocdec: put the code not shared with other demuxers under appropriate ifdef
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
This unbreaks muxing-encoding
Example:
ffmpeg -i matrixbench_mpeg2.mpg new.avi
-rw-r----- 1 michael michael 226035354 Jan 1 16:27 new.avi
-rw-r----- 1 michael michael 10016802 Jan 1 16:28 ref.avi
Also av_get_audio_frame_duration() itself uses frame_size
This reverts commit 29e6606e9b, reversing
changes made to 53448461a7.
* commit '955aec3c7c7be39b659197e1ec379a09f2b7c41c':
mpegaudiodecheader: check the header in avpriv_mpegaudio_decode_header
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* commit '6bf4c1d71199b92894f24db6386ed5070e590a16':
r3d: do not create the audio stream until we know the sample rate
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* commit '9f1eccb97bf8894cb18b14f642500686505ef186':
ff_parse_specific_params: do not use AVCodecContext.frame_size
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
support reading encrypted mp4 using aes-ctr, conforming to ISO/IEC
23001-7.
a new parameter was added:
- decryption_key - 128 bit decryption key (hex)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This fixes crashes caused by out-of-bounds writes.
Reviewed-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
This commit for qtpalette.c and qtpalette.h adds 1-bit video to the
"palettized video" category, since if the video sample description
contains a palette, the two colors in the palette can be any color, not
necessarily black & white.
Unfortunately, I've noticed that the qtrle (QuickTime Animation) decoder
blindly assumes that 1-bit video is black & white. I don't have enough
knowledge about the decoder to fix this, though.
Below is a link to a sample 1-bit QuickTime Animation clip of a rotating
earth that uses blueish colors, and they will be correctly rendered in
QuickTime, but not in FFmpeg (which will use black & white).
https://drive.google.com/open?id=0B3_pEBoLs0faUlItWm9KaGJSTEE
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This solves the problem discussed in https://ffmpeg.org/pipermail/ffmpeg-devel/2015-September/179238.html
by allowing AVCodec::write_header to be delayed until after packets have been
run through required bitstream filters in order to generate global extradata.
It also provides a mechanism by which a muxer can add a bitstream filter to a
stream automatically, rather than prompting the user to do so.
Palettized QuickTime video in Matroska has hitherto not been recognized
whatsoever, and the "palette" used has been completely random.
The patch for matroskadec.c fixes this issue by adding a palette side
data packet in matroska_deliver_packet(), much in the same way as it's
done in mov.c.
Video samples for testing are available at
https://drive.google.com/open?id=0B3_pEBoLs0faWElmM2FnLTZYNlk.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This consists mainly of moving the palette handling from
the mov_parse_stsd_video() function to a new ff_get_qtpalette() function
in the new file qtpalette.c, which will be shared by both matroskadec.c and
mov.c.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Previous "currently size is limited" didn't give away
much in terms of useful info.
Signed-off-by: Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
So far an AC-3 elementary stream is refered to in the PMT according to
System A (ATSC). An E-AC-3 ES in contrast is embedded the System B (DVB) way.
To fix this inconsistency, this commit changes the default E-AC-3 behaviour to
use the ATSC way, too. Furthermore a new flag is added to optionally select the
DVB way (regarding both codecs and possible further differences in the future).
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes a specific srt sample, which has an event with negative duration.
libavcodec will convert an event with negative duration to an ASS event
which will be displayed forever, which is not wanted here.
Treat negative duration always as unknown duration instead, and show it
until the next subtitle event.
A negative bits_per_coded_sample doesn't make sense.
If it is too large, the size calculation for av_get_packet overflows,
resulting in allocation of a too small buffer.
Also make sure width and height are sane.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
This fixes NULL pointer dereferencing.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
If it is negative, it can cause the byte position to move backwards in
avio_skip, which in turn makes sm_size negative and thus size larger
than the size of the packet buffer, causing invalid writes in avio_read.
Also fix potential overflow of avio_tell(bc) + value_len.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Fixes ubsan runtime error: null pointer passed as argument 2, which is
declared to never be null
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Since the parser was merged back almost 2 months ago this is the first
time the bitstream of the container has been updated.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* commit 'e02de9df4b218bd6e1e927b67fd4075741545688':
lavc: export Dirac parsing API used by the ogg demuxer as public
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
A negative codec_id cannot be handled by the found_decoder API of
AVStream->info: if the codec_id is not recognized, found_decoder is set
to -codec_id, which has to be '<0' according to the API documentation.
This can cause NULL pointer dereferencing in try_decode_frame.
Also make sure the codec_type matches the expected one for codec_id.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
support writing encrypted mp4 using aes-ctr, conforming to ISO/IEC
23001-7.
3 new parameters were added:
- encryption_scheme - allowed values are none (default) and cenc-aes-ctr
- encryption_key - 128 bit encryption key (hex)
- encryption_kid - 128 bit encryption key identifier (hex)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
So far an AC-3 elementary stream is refered to in the PMT according to
System A (ATSC). However System B (DVB) has a different way to signal an AC-3
ES within the PMT. This different way can be enabled by a new flag. The flag is
more generally named 'system_b' as there are further differences between ATSC
and DVB (e.g. the signalling of E-AC-3) which should then also be covered by it
in the future.
Bug-Id: 73
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Similar to 33fefdb44.
Fix trac ticket #4921.
Signed-off-by: Nicolas George <george@nsup.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The current muxer behaviour is to create streams in read_header() based
on the audio/video presence flags, but fill in the stream parameters
later when we actually get some packets for them. This is rather shady,
since other demuxers set the stream parameters immediately when the
stream is created and do not touch the stream codec context after that.
Change the flv demuxer to behave in the same way as other similar
demuxers -- create the streams only when we get a packet for them.
Almost all the places from which this function is called already check
the header manually and in the two that don't (the mp3 muxer) the check
should not cause any problems.
It will not be set unless the muxing codec context is also the encoding
context, which is discouraged. When the frame size is not known from
av_get_audio_frame_duration(), the fallback should still be good enough.
It will not be set if the stream codec context is not the encoding
context. Use av_get_audio_frame_duration() instead, it should work for
all audio codecs supported by the muxer.
* commit '5f3a081b42b84404a40a52c80ef7a354cf048c56':
avi: Spin out the logic to position to the next non-interleaved stream
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
matroskaenc applies divisors to the display width/height when generating
stereo content. This patch adds the corresponding multipliers to matroskadec
so that the original sample aspect ratio can be recovered.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
* commit 'a0fa6d06b848f26b16ba12f0a9a4a85b93ab8022':
matroska: Warn when metadata references a non-existent element
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
It is required to call va_end for each invocation of va_start within the
same function.
Fixes: CID 1341583.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
Checking the codec context parameters to find out this information is
far too unreliable to be useful, so it is safer to assume B-frames are
always present.
QuickTime metadata can come after trak data. Add indicator for which trak is being parsed (-1 if none) so that global metadata after the trak can be parsed.
Signed-off-by: Neil Birkbeck <neil.birkbeck@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If size is zero, avio_get_str fails, leaving the buffer uninitialized.
This causes invalid reads in av_set_options_string.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Otherwise the ffm demuxer can return the same packets endlessly, if
there is no server attached.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
ts->mux_rate is int (signed 32-bit) type. The period calculations
will start to overflow when mux_rate > 5mbps. This fixes overflows
by converting first to 64-bit type.
Fixes#5044.
Signed-off-by: Timo Teräs <timo.teras@iki.fi>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This code is disabled by default so not to regress endpoints sending invalid MIME, but can be enabled via AVOption 'strict_mime_boundary'
Signed-off-by: Alex Agranovsky <alex@sighthound.com>
matroskaenc.c applies divisors to the display width/height when generating
stereo content. This patch adds the corresponding multipliers to matroskadec.c
so that the original sample aspect ratio can be recovered.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
"Fast seek" uses linear interpolation to find the position of the
requested seek time. For CBR this is more direct than using the
mp3 TOC and bypassing the TOC avoids problems with TOC precision.
(see https://crbug.com/545914#c13)
For VBR, fast seek is not precise, so continue to prefer the TOC
when available (the lesser of two evils).
Also, some re-ordering of the logic in mp3_seek to simplify and
give usetoc=1 precedence over fastseek flag.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
* commit '12b14382861fbf19378afcddaa19cd9a949a86a3':
udp: Provide additional information on getaddrinfo failure
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* commit '8375dc1dd101d51baa430f34c0bcadfa37873896':
asfdec: handle the case when the stream index has an invalid value better
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
Fixes probing of truehd/mlp files with a lot of frames in between the
major sync frames. The spec allows a distance of up to 128 frames in
between major sync frames, which leads to the probing code not reaching
the desired score.
Avoid decoding a frame to get the codec parameters while the codec
supports FF_CODEC_CAP_SKIP_FRAME_FILL_PARAM. This is particulary useful
to avoid decoding twice images (once in avformat_find_stream_info and
once when the actual decode is made).
If duration is still AV_NOPTS_VALUE when opening the next file, we can assume
that outpoint is not set.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Marton Balint <cus@passwd.hu>
