Fixes out of array read
Fixes: 049fdf78565f1ce5665df236d90f8657/asan_heap-oob_10a5a97_1026_42f9d4855547329560f385768de2f3fb.wtv
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When ffmpeg exit by exception, start a new ffmpeg will
cover the old segment list, add this flag can continue
append the new segments into old hls segment list
Signed-off-by: LiuQi <liuqi@gosun.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This change relaxes the whitelist on reading color metadata in MOV/BMFF
containers. The whitelist on writing values is still in place.
As a consequence it also fixes an apparent bug in reading 'nclc' values.
The 'nclc' spec [1] is in harmony with ISO 23001-8 for the values it
lists, but the code getting removed was remapping 5->6 and 6->7 for
primaries, which is incorrect, and was remapping 6->5 for color matrix
("colorspace" in the code), which is equivalent but an unnecessary
inconsistency. This logic error doesn't appear in movenc.
Removing the whitelist allows proper conversion when the source codec
relies on the container for proper signaling of newer codepoints, such
as DNxHR and VP9. If converting to a codec or container that has updated
its spec to include the new codepoints, the metadata will be preserved.
If going back to MOV/BMFF, the output whitelist will still kick in, so
this won't result in out-of-spec files being created.
[1] https://developer.apple.com/library/mac/technotes/tn2162/_index.html
Signed-off-by: Steven Robertson <steven@strobe.cc>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes out of array read
Fixes: 13262c363a28da8d6bdcc472aed6e9dc/asan_heap-oob_cfb5e2_3733_31cf3fcc783295c34222eb070a784f84.3gp
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Instead of silently ignoring the content_type option in listen mode,
apply its value to the provided "Content-Type:" header.
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Instead of silently ignoring the headers option in listen mode, use
the provided headers.
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The values don't need to be hardcoded since the correct values are
returned by avs_bits_per_pixel.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Enable the MXF muxer to mux baseline H.264 video streams.
Signed-off-by: Matthias Hunstock <atze@fem.tu-ilmenau.de>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Also set a default_whitelist for mmsh and ffrtmphttp.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This will be used to allow writing file sequences using the tee output onto
multiple places in parallel
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If negative pts are possible for some codecs in ogg then the code needs to be
changed to use signed values.
Found-by: Thomas Guilbert <tguilbert@google.com>
Fixes: clusterfuzz_usan-2016-08-02
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Set the stream_id to 0xbd (private_stream_id_1). Tools seem to assume
that value, and this is consistent with MPEG TS specification (ITU-T
H.222.0 section 2.12.3).
Rev #2: Fixes doubled header writing, checked FATE running without errors
Rev #3: Fixed coding style
This commit addresses the following scenario:
we are using ffmpeg to transcode or remux mkv (or something else) to mkv. The result is being streamed on-the-fly to an HTML5 client (streaming starts while ffmpeg is still running). The problem here is that the client is unable to detect the duration because the duration is only written to the mkv at the end of the transcoding/remoxing process. In matroskaenc.c, the duration is only written during mkv_write_trailer but not during mkv_write_header.
The approach:
FFMPEG is currently putting quite some effort to estimate the durations of source streams, but in many cases the source stream durations are still left at 0 and these durations are nowhere mapped to or used for output streams. As much as I would have liked to deduct or estimate output durations based on input stream durations - I realized that this is a hard task (as Nicolas already mentioned in a previous conversation). It would involve changes to the duration calculation/estimation/deduction for input streams and propagating these durations to output streams or the output context in a correct way.
So I looked for a simple and small solution with better chances to get accepted. In webmdashenc.c I found that a duration is written during write_header and this duration is taken from the streams' metadata, so I decided for a similar approach.
And here's what it does:
At first it is checking the duration of the AVFormatContext. In typical cases this value is not set, but: It is set in cases where the user has specified a recording_time or an end_time via the -t or -to parameters.
Then it is looking for a DURATION metadata field in the metadata of the output context (AVFormatContext::metadata). This would only exist in case the user has explicitly specified a metadata DURATION value from the command line.
Then it is iterating all streams looking for a "DURATION" metadata (this works unless the option "-map_metadata -1" has been specified) and determines the maximum value.
The precendence is as follows: 1. Use duration of AVFormatContext - 2. Use explicitly specified metadata duration value - 3. Use maximum (mapped) metadata duration over all streams.
To test this:
1. With explicit recording time:
ffmpeg -i file:"src.mkv" -loglevel debug -t 01:38:36.000 -y "dest.mkv"
2. Take duration from metadata specified via command line parameters:
ffmpeg -i file:"src.mkv" -loglevel debug -map_metadata -1 -metadata Duration="01:14:33.00" -y "dest.mkv"
3. Take duration from mapped input metadata:
ffmpeg -i file:"src.mkv" -loglevel debug -y "dest.mkv"
Regression risk:
Very low IMO because it only affects the header while ffmpeg is still running. When ffmpeg completes the process, the duration is rewritten to the header with the usual value (same like without this commit).
Signed-off-by: SoftWorkz <softworkz@hotmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
HLS demuxer calls the subdemuxer avformat_find_stream_info() while
overriding the subdemuxer AVFMTCTX_NOHEADER flag by clearing it.
However, this prevents some streams in some MPEG TS streams from being
detected properly.
Simply removing the clearing of the flag would cause the inner
avformat_find_stream_info() call to take longer in some cases, without
a way to control it.
To fix the issue, do not clear the flag but propagate it to HLS demuxer.
To avoid the above-mentioned mandatory delay, the call to
avformat_find_stream_info() is dropped except in the HLS ID3 timestamped
case. The HLS demuxer user should be calling avformat_find_stream_info()
on the HLS demuxer if it wants to find the stream info.
The main streams are now created dynamically after read_header time if
the subdemuxer uses AVFMTCTX_NOHEADER (mpegts).
Subdemuxer avformat_find_stream_info() is still called for the HLS ID3
timestamped case as the HLS demuxer needs to know the packet durations
to properly interleave ID3 timestamped streams with MPEG TS streams on
sub-segment level.
Fixes ticket #4930.
Creation of main demuxer streams from subdemuxer streams is moved to
update_streams_from_subdemuxer() which can be called repeatedly.
There should be no functional changes.
Commit 81306fd4bdf ("hls: eliminate ffurl_* usage", merged in d0fc5de3a6)
changed the hls demuxer to use AVIOContext instead of URLContext for its
HTTP requests.
HLS demuxer uses the "offset" option of the http demuxer, requesting
the initial file offset for the I/O (http URLProtocol uses the "Range:"
HTTP header to try to accommodate that).
However, the code in libavformat/aviobuf.c seems to be doing its own
accounting for the current file offset (AVIOContext.pos), with the
assumption that the initial offset is always zero.
HLS demuxer does an explicit seek after open_url to account for cases
where the "offset" was not effective (due to the URL being a local file
or the HTTP server not obeying it), which should be a no-op in case the
file offset is already at that position.
However, since aviobuf.c code thinks the starting offset is 0, this
doesn't work properly.
This breaks retrieval of ranged media segments.
To fix the regression, just drop the seek call from the HLS demuxer when
the HTTP(S) protocol is used.
Commit 9200514ad8 ("lavf: replace AVStream.codec with
AVStream.codecpar") merged in commit 6f69f7a8bf changed
avformat_find_stream_info() to put the extradata it got from
st->parser->parser->split() to st->internal->avctx instead of st->codec
(extradata in st->internal->avctx will be later copied to st->codecpar).
However, in the same function, the "is stream ready?" check was changed
to check for extradata in st->codecpar instead of st->codec, even
though st->codecpar is not yet updated at that point.
Extradata retrieved from split() is therefore not considered anymore,
and avformat_find_stream_info() will therefore needlessly continue
probing in some cases.
Fix that by checking for the extradata at st->internal->avctx where it
is actually put.
It's a small and simple function that can be inlined.
This removes one private symbol and should reduce object dependencies with the next
major bump
Signed-off-by: James Almer <jamrial@gmail.com>
This ensures that AV_NOPTS_VALUE value is handled
correctly.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Jan Sebechlebsky <sebechlebskyjan@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
When seeking a file where codec delay is greater than 0, the timecode
can become negative after offsetting by the codec delay. Failing to cast
to a signed int64 will cause the check against skip_to_timecode to evaluate
true for these negative values. This breaks the "skip_to" seek mechanism.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fix the confusion around the used time base.
Check size returned from avio_size()
Signed-off-by: Jörn Heusipp <osmanx@problemloesungsmaschine.de>
Signed-off-by: Josh de Kock <josh@itanimul.li>
The header was never installed and the function is only used in libavformat
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Add ff_format_output_open utility function to wrap
io_open callback of AVFormatContext structure.
Signed-off-by: Jan Sebechlebsky <sebechlebskyjan@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
This will add support for flushing by writing NULL
packet to the tee muxer, which propagates the action
to slave muxers as expected.
Signed-off-by: Jan Sebechlebsky <sebechlebskyjan@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
Use ff_stream_encode_params_copy() to copy encoding-related
fields (parameters) of stream.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Jan Sebechlebsky <sebechlebskyjan@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
because the BSF logic was re-factored into a shareable
function and both av_write_frame and av_interleaved_write_frame use it it
Signed-off-by: LiuQi <liuqi@gosun.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit '76729970049fe95659346503f7401a5d869f9959':
mov: Implement support for multiple sample description tables
Notes:
* The sc->stsc_data[index].id checks have been moved from the mov_read_stsc
to mov_read_packet before the value is used in mov_change_extradata to
not break playback of samples with broken stsc entries (see sample of
ticket #1918).
* sc->stsc_index is now checked against sc->stsc_count - 1 before it
is incremented so it remains lesser than sc->stsc_count. Fixes a crash
with:
./ffmpeg -i matrixbench_mpeg2.mpg -t 1 -frag_duration 200k test.mov
./ffprobe -show_packets test.mov
Merged-by: Matthieu Bouron <matthieu.bouron@stupeflix.com>
This allows simpler selection of flac in ogg from the command line,
while following the RFC 5334 recommendation[1] for the oga extension.
[1] https://tools.ietf.org/html/rfc5334#section-10.3
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
support split hls segment at duration set by hls_time
Signed-off-by: LiuQi <liuqi@gosun.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This fixes part of Ticket5676
This fixes kodi, mpv, chromium and ffplay build against 3.0 and linked to 3.1
This is a similar ABI fix to 1eb43af1a0
Approved-by: BBB
Approved-by: jamrial
Approved-by: BtbN
Approved-by: nevcairiel
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This ensures the AVStream->codec entry is kept in sync when new streams are
discovered mid-playback or changes to the context occur from other sources.
Fixes trac 5678.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit 'b668662939de3a02454cfc9ba3e6d10b87527a40':
get_bits: Move BITSTREAM_READER_LE definition before all relevant #includes
The merge commit also includes changes for libavcodec/interplayacm.c and
libavcodec/truemotion2rt.c
Merged-by: Clément Bœsch <clement@stupeflix.com>
* commit '3fdffc032e8ea5676bc0c2551b900c0dc887835b':
rtsp: Use avcodec_descriptor_get instead of avcodec_find_decoder
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
Add -movflags use_metadata_tags to the mov muxer. This will cause
the muxer to write all metadata to the file in the keys and mtda
atoms.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit 'f12a705ee570e16ca692c66b62821a2dbdf82566':
movenc: Factorize a function for finding a metadata entry and the associated language
Merged-by: Clément Bœsch <clement@stupeflix.com>
* commit 'd34826c33d401929b2ff8aee161fd39ad0a73613':
mov: Add a comment referring to the standard that defines the loci box
Merged-by: Clément Bœsch <clement@stupeflix.com>
* commit 'a79aafd0b4d37eda6f15dc68e6509d4e815290c9':
movenc: Add a test for VFR with b-frames, with a duration change at a fragment end
Merged-by: Matthieu Bouron <matthieu.bouron@stupeflix.com>
* commit 'e1eb0fc960163402bbb4e630185790488f7d28ed':
movenc: Use packets in interleaving queues for the duration at the end of fragments
Merged-by: Matthieu Bouron <matthieu.bouron@stupeflix.com>
* commit 'db7968bff4851c2be79b15b2cb2ae747424d2fca':
avio: Allow custom IO users to get labels for the output bytestream
Merged-by: Matthieu Bouron <matthieu.bouron@stupeflix.com>
* commit '0c4468dc185fa8b9e7d6add914595c5e928b24fd':
stereo3d: Add API to get name from value or value from name
Merged-by: Clément Bœsch <clement@stupeflix.com>
Implement variable sized big-endian integers, since these are found
in files created by ARRI cameras.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit '393596f9d51134d6e45d81ae129223f4faea1232':
mpegtsenc: stop impersonating ses in sdt
This commit also includes the needed FATE updates later spotted by
Martin Storsjö and fixed in 34effe816f on
Libav side.
Merged-by: Clément Bœsch <u@pkh.me>
The internal avctx bitrate is copied into codecpar after estimate_timings()
Fixes Ticket5646
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
this removes the need to probe to discover aac streams
inside mpegts containers, thus speeding up initial playback.
Reviewed-by: Hendrik Leppkes <h.leppkes@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit '74d98d1b0e0e7af444c933ea3c472494de3ce6f2':
mpegts: Validate the SL Packet Header Configuration
See e630ca5111
Our local timestamp_len > 64 is adjusted to > 63 to match the Libav
check and the actual specifications (14496-1, 10.2.2).
There is no need to request a sample as it violates the specifications
and such a file would likely be the result of a crafted/fuzzed sample.
On the other hand, the clipping of the value is kept for extra safety.
Merged-by: Clément Bœsch <clement@stupeflix.com>
It has no use afterwards and freeing it before calling ff_flac_parse_picture()
may help prevent OOM issues on memory constrained scenarios.
Signed-off-by: James Almer <jamrial@gmail.com>
We haven't had a stable release since the packet_gap addition, so probably it
is worth reworking the option to something that makes more sense to the end
user. Also add burst_bits option to specify maximum length of bit bursts.
Signed-off-by: Marton Balint <cus@passwd.hu>
This function needs to return false, or data in the additional tables
will be skipped, and the decoder will not be able to decode frames
associated with them.
Store data from each stsd in a separate extradata buffer, keep track of
the stsc index for read and seek operations, switch buffers when the
index differs. Decoder is notified with an AV_PKT_DATA_NEW_EXTRADATA
packet side data.
Since H264 supports this notification, and can be reset midstream, enable
this feature only for multiple avcC's. All other stsd types (such as
hvc1 and hev1) need decoder-side changes, so they are left disabled for
now.
This is implemented only in non-fragmented MOVs.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
The stat struct is defined to stati64, which requires using the appropriate wstati/stati functions as well.
Fixes a whole bunch of compiler warnings as well as build breakage with the decklink avdevice.
Fixes trac #5640
We still only support one single layer though, but this allows
receiving streams that have this structure present even for
single layer streams.
Signed-off-by: Martin Storsjö <martin@martin.st>
Docs clearly states that av_write_trailer should only be called if
avformat_write_header was successful, therefore we have to deinit if we return
failure.
Signed-off-by: Marton Balint <cus@passwd.hu>
