* commit 'b08569a23948db107e5e6175cd4c695427d5339d':
lavf: Replace the ASF demuxer
Conflicts:
Changelog
libavformat/asf.h
libavformat/asfdec.c
libavformat/version.h
tests/ref/fate/wmv8-drm-nodec
tests/ref/seek/lavf-asf
The rewritten demuxer is placed in a new file, the current demuxer is
left as default. Carl has tested both and the one working better is
default.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The old one is the result of the reverse engineering and guesswork.
The new one has been written following the now-available specification.
This work is part of Outreach Program for Women Summer 2014 activities
for the Libav project.
The fate references had to be changed because the old demuxer truncates
the last frame in some cases, the new one handles it properly.
The seek-test reference is changed because seeking works differently
in the new demuxer. When seeking, the packet is not read from the stream
directly, but it is rather constructed by the demuxer. That is why
position is -1 now in the reference.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Otherwise sm_size can be larger than size, which results in a negative
packet size.
Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Electronic Arts VP6 files may contain two video streams: one for the
primary video stream and another for the alpha mask. The file format
uses identical data structures for both streams.
Signed-off-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Result differs in pkt_duration and time_base.den for some reason.
Right now it tests only one example (adjusted to match the output).
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The code is simply broken, the read packets are not aligned to
the mp3 frames, the file end or the id3 tag thus this simply
cannot reliably find the ID3v1 tag to remove it
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '6ec688e1bc76dd93151cbca1c340162ae4b10d77':
mp3: enable packed main_data decoding in MP4
Conflicts:
libavcodec/mpegaudiodec_template.c
Only the parts needed to support the available sample are merged
the remaining error checks are left in place
Merged-by: Michael Niedermayer <michaelni@gmx.at>
14496-3 suggests packing main_data of MP3 that is usually scattered
into multiple frames due to bit reservoir.
However, after packing main_data into a access unit, bitrate index
in the MPEG audio frame header doesn't match with actual frame size.
In order to accept this, this patch removes unnecessary frame size
checking on mp3 decoder.
Also, mov demuxer was changed to use MP3 parser only on special cases
(QT MOV with specific sample description) to avoid re-packetizing.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The APIC description must be unique, and some ID3v2 tag writers add
spaces to write several APIC entries with the same description. The
trailing spaces simply serve as a way to disambiguate the description.
Do this so that API users do not have to special-case mp3 to fix this
cosmetic issue.
Requested-by: wm4
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Calling ffio_ensure_seekback() if ffio_init_checksum() has been called
on the same context can lead to out of bounds memory accesses and
crashes. The reason is that ffio_ensure_seekback() does not update
checksum_ptr after reallocating the buffer, resulting in a dangling
pointer.
This effectively fixes potential crashes when opening mp3 files.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Also fix typo found by Lou Logan:
Sacrifying -> Sacrificing
Reviewed-by: Lou Logan <lou@lrcd.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
In the TTA extradata re-construction the values are written with
avio_wl16 and if they don't fit into uint16_t, this triggers an
av_assert2 in avio_w8.
Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
And default to 8000 if it is invalid.
An invalid sample rate can trigger av_assert2 in av_rescale_rnd.
Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
This should make no difference as the byte is ignored
Found-by: tim nicholson <nichot20@yahoo.com>
Reviewed-by: tim nicholson <nichot20@yahoo.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Added HLS encryption with -hls_key_info_file <key_info_file> option. The
first line of key_info_file specifies the key URI written to the
playlist. The key URL is used to access the encryption key during
playback. The second line specifies the path to the key file used to
obtain the key during the encryption process. The key file is read as a
single packed array of 16 octets in binary format. The optional third
line specifies the initialization vector (IV) as a hexadecimal string to
be used instead of the segment sequence number (default) for encryption.
Changes to key_info_file will result in segment encryption with the new
key/IV and an entry in the playlist for the new key URI/IV.
Key info file format:
<key URI>
<key file path>
<IV> (optional)
Example key URIs:
http://server/file.key
/path/to/file.key
file.key
Example key file paths:
file.key
/path/to/file.key
Example IV:
0123456789ABCDEF0123456789ABCDEF
Example:
ffmpeg -f lavfi -i testsrc -c:v h264 -hls_key_info_file file.keyinfo
foo.m3u8
file.keyinfo:
http://server/file.key
/path/to/file.key
0123456789ABCDEF0123456789ABCDEF
Example shell script:
BASE_URL=${1:-'.'}
openssl rand 16 > file.key
echo $BASE_URL/file.key > file.keyinfo
echo file.key >> file.keyinfo
echo $(openssl rand -hex 16) >> file.keyinfo
ffmpeg -f lavfi -re -i testsrc -c:v h264 -hls_flags delete_segments \
-hls_key_info_file file.keyinfo out.m3u8
--
Signed-off-by: Christian Suloway <csuloway@globaleagleent.com>
Signed-off-by: Dan Dennedy <dan@dennedy.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
GNUTLS_SHUT_RDWR means GnuTLS will keep waiting for the server's
termination reply. But since we don't shutdown the TCP connection at
this point yet, GnuTLS will just keep skipping actual data from the
server, which basically is perceived as hang.
Use GNUTLS_SHUT_WR instead, which doesn't have this problem.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
GNUTLS_SHUT_RDWR means GnuTLS will keep waiting for the server's
termination reply. But since we don't shutdown the TCP connection at
this point yet, GnuTLS will just keep skipping actual data from the
server, which basically is perceived as hang.
Use GNUTLS_SHUT_WR instead, which doesn't have this problem.
Signed-off-by: Martin Storsjö <martin@martin.st>
Neccessary -> Necessary
formated -> formatted
thee -> the
eventhough -> even though
seperately -> separately
Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
* commit 'a7ac1a7b94447f33ae95be4d6d186e2775977f91':
flv: Name an enum and use its type
Conflicts:
libavformat/flvdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Some files have SeekHead elements with broken IDs. They mismatch with
the ID of the destination element. These files are written by
"IDMmkvlib0.1" (as identified by the MuxingApp and WritingApp elements),
and the SeekHead IDs are actually endian-swapped.
This confuses the SeekHead logic of the demuxer. It will read some
elements twice, because the SeekHead ID is used to identify and remember
already read elements. With the file at hand, the stream list was
duplicated by reading the Tracks element twice.
Fix this by rejecting invalid EBML IDs in SeekHead entries. (This fix is
relatively specific to the broken file at hand, and doesn't protect
against some other cases of broken SeekHead, such as valid but
mismatching target element IDs.)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
There is no support for non integer sample rates, using doubles/floats currently could
only lead to rounding differences between platforms
Previous version Reviewed-by: Mark Harris <mark.hsj@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Make the logic in libavformat/hevc.c parse_rps align with libavcodec/hevc_ps.c ff_hevc_decode_short_term_rps
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is required for the (not yet in git) private stream detection/export,
no other testcase known
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Send a footer to correctly close client sockets.
This fixes network errors in client applications.
Signed-off-by: Stephan Holljes <klaxa1337@googlemail.com>
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
display_matrix_size is only initialized when av_stream_get_side_data()
returns a side data pointer. The code is safe since the only effect this
has is setting the display_matrix pointer to NULL which it was already
anyway.
The flag was set unintentionally and the code will break if a NULL
packet is passed in.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b14086ca38efa1a86cb0f0c6aa147b05f698877b':
mkv: Correctly report the latest packet had been flushed
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The first check is done without the AVIOContext, so alloc it only if said check succeeds
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
The first check is done without the AVIOContext, so alloc it only if said check succeeds
Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: James Almer <jamrial@gmail.com>
* commit 'caf7be30b11288c498fae67be4741bfbf083d977':
mpjpgdec: free AVIOContext leak on early probe fail
Conflicts:
libavformat/mpjpegdec.c
See: 34d278f983, this was mistakenly reimplemented, also see ffmpeg IRC log of today
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This can be useful for debugging, or in scenarios where the user
doesn't want to use the system's DNS settings for whatever reason.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Together with the next commit this prevents non-PCM S302M from being selected unless either
it can be decoded or the user selects passthrough/copy
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Per matroska Block Structure [1], for keyframes 0th bit of the flag
should not be set (unlike SimpleBlocks). For Blocks, keyframes is
inferred by the absence of ReferenceBlock element (as done by
matroskadec). This CL writes the flag correctly and inserts the
ReferenceBlock element for non-keyframes. The timestamp inserted is
that of the immediately preceding frame (which is true for VP8 and VP9
- the only 2 codecs using the matroska block element as of now). It
also considers all non-video frames (audio, subtitles, metadata) to
be keyframes.
[1] http://www.matroska.org/technical/specs/index.html#block_structure
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is the maximum rate possible based on the frame size limit of MXF D-10
Previous version reviewed by tim nicholson <nichot20@yahoo.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
mpz_import and mpz_export were added in GMP 4.1, in 2002.
This simplifies the DH code by clarifying that it only uses pure
bignum functions, no other parts of nettle/hogweed.
Signed-off-by: Martin Storsjö <martin@martin.st>
If avio_read fails, the buffer can contain uninitialized data.
This fixes 'Conditional jump or move depends on uninitialised value(s)'
valgrind warnings, and addresses a few memleaks.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
In this case the mov demuxer can return a large number of empty packets.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Otherwise the loop can take a lot of time if num_descr is very large.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
OSX does not know MSG_NOSIGNAL. BSD (which OSX is based on) has got
the socket option SO_NOSIGPIPE (even if modern BSDs also support
MSG_NOSIGNAL).
Signed-off-by: Martin Storsjö <martin@martin.st>
OSX does not know MSG_NOSIGNAL, and provides its own non-standard
mechanism instead. I guess Apple hates standards.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b2f0f37d242f1194fe1f886557cf6cefdf98caf6':
rtmpdh: Generate the whole private exponent using av_get_random_seed() with nettle/gmp
Merged-by: Michael Niedermayer <michaelni@gmx.at>
A negative sample duration is invalid according to the spec, but there
are samples that use it for the DTS calculation, e.g.:
http://files.1f0.de/samples/mp4-negative-stts-problem.mp4
These currently get out of A/V sync.
Also change the logging type to AV_LOG_WARNING, because decoding the
sample can continue.
Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Users have no means to find out from a failure how to make it work
or is it preferred to check and print a warning for h264 concat without auto_convert ?
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'e9e86d9ef637f5a600c76b352ffe5a82b71b25d1':
rtmpdh: Create sufficiently long private keys for gcrypt/nettle
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '8016a1bd3b60e917e1b12748dd80c06c3462c286':
rtmpdh: Remove an unnecessary check in the gcrypt/nettle dh_compute_key
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '063f7467e4d14ab7fe01b2845dab60cc75df8b53':
rtmpdh: Add fate test for the DH handshake routine
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '0508faaa11bf7507ffdd655aee57c9dc5a8203f4':
rtmpdh: Pass the actual buffer size of the output secret key
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '9f1b3050d9e31e9283d818f3640f3460ac8cfb5b':
rtmpdh: Check the output buffer size in the openssl version of dh_compute_key
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '127d813bcb5705202b7100cf1eccd1e26d72ba14':
rtmpdh: Fix a local variable name in the nettle/gcrypt codepath
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '78efc69e7c990226f4b913721ef1b308ca5bfa04':
rtmpdh: Make sure ret is initialized in the nettle version of bn_hex2bn
Merged-by: Michael Niedermayer <michaelni@gmx.at>
There was a misunderstanding betewen bits and bytes for the parameter
value for generating random big numbers.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'd4d90504a687d2c0ef77ccf11d831f24dcff9cf1':
tls_gnutls: Add missing includes for the gcrypt thread safety callbacks
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* cehoyos/master:
lavc/x264: Support bgr0 as input pix_fmt.
lavf: Use av_codec_get_tag2() in avformat_query_codec().
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Move the OpenSSL and GnuTLS implementations to their own files. Other
than the connection code (including options) and some boilerplate, no
code is actually shared.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Apparently it can happen that a mp3 file has junk data between id3 tag
and actual mp3 data. Skip this to avoid outputting nonsense timestamps.
(Two packets had the same timestamps, because the mp3 parser failed to
compute a frame duration.)
In this case, the junk consisted of 1044 bytes of zero, which
incidentally is the same size as normal mp3 frames in this stream. I
suspect the mp3 was edited with some tool which wiped the Xing/LAME
headers. Data near the end of the file suggests it was encoded with
"LAME3.97", but the normal Xing/LAME headers are missing. So this could
be "normal". mpg123 also attempts to skip at least 64KB of junk data by
scanning for headers.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Move the OpenSSL and GnuTLS implementations to their own files. Other
than the connection code (including options) and some boilerplate, no
code is actually shared.
Signed-off-by: Martin Storsjö <martin@martin.st>
In this case the mov demuxer can return a large number of empty packets.
Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
If avio_read fails, the buffer can contain uninitialized data.
This fixes 'Conditional jump or move depends on uninitialised value(s)'
valgrind warnings.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
If it fails, the buffers can be (partially) uninitialized.
This fixes 'Conditional jump or move depends on uninitialised value(s)'
valgrind warnings.
Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Prevents read of uninitialized variable
Based on patch by: Andreas Cadhalpun <andreas.cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Previous version Reviewed-by: tim nicholson <nichot20@yahoo.com>
Previous version Reviewed-by: Tomas Härdin <tomas.hardin@codemill.se>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is the 1st patch in preparation for using WebPAnimEncoder API for encoding
and muxing WebP images.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Since the underlying URLContext read functions are used,
they handle interruption, without having to handle it at
this level.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids hijacking the fd, by reading using the normal
URLContext functions instead. This allowing reading data that has
been buffered in the underlying URLContext.
This avoids using the libraries own send functions that can
cause SIGPIPE.
The fd is still used for polling the lowlevel socket, for
waiting for retries.
Signed-off-by: Martin Storsjö <martin@martin.st>
These loops can take a lot of time if count is very large.
Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
This can unnecessarily waste a lot of time.
Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
nut->last_syncpoint_pos doesn't necessarily change between resync
attempts, so find_any_startcode can return the same startcode again.
Thus remember where the last resync happened and don't try to resync
before that.
This can't be done locally in nut_read_packet, because this wouldn't
prevent infinite resync loops, where after the resync a packet is
returned and while reading a following packet the resync happens again.
Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Added in 361702660d. Modified version that
doesn't use this label merged in 55231323b0,
thus obsoleting this label.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes a memleak if read_kuki_chunk is executed more than once.
Reviewed-by: Carl Eugen Hoyos <cehoyos@ag.or.at>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
If avio_read fails, the buffer can contain uninitialized values.
Reviewed-by: Carl Eugen Hoyos <cehoyos@ag.or.at>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Since len is an unsigned int, the comparison is currently treated as
unsigned and thus ignores all errors from avio_read.
Thus cast len to int, which is unproblematic, because at that point len
is between 0 and 4.
This fixes 'Conditional jump or move depends on uninitialised value'
valgrind warnings in is_tag.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b8d2630c5327d2818d05c8a48be0417905d8e0fd':
dashenc: Reduce the segment duration if cutting out parts with edit lists
Merged-by: Michael Niedermayer <michaelni@gmx.at>
id should be an integer, not a string. It is also optional, so use
contentType instead which is the proper attribute for these values.
This addresses ticket #4545, fixing an MPD validation error.
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '254f3daba4271c1918d9a7ad155b1442ef93ed29':
nut: Make sure to clean up on read_header failure
Conflicts:
libavformat/nutdec.c
See: 361702660d
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This makes sure that the time + duration of the first segment
matches the start time of the next segment for e.g. AAC audio
with encoder delay.
Signed-off-by: Martin Storsjö <martin@martin.st>
id should be an integer, not a string. It is also optional, so use
contentType instead which is the proper attribute for these values.
This fixes an MPD validation error.
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This currently works for most users because
avformat_open_input sets it, but this patch fixes any
applications not using that function.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
APIC tags always have a description. Tag writers obviously leave it
empty if there is no description. In this case, libavformat would export
"" as title. Do not set the title instead.
APIC tags always have a description. Tag writers obviously leave it
empty if there is no description. In this case, libavformat would export
"" as title. Do not set the title instead.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '7b734ee55dbb8476d7ad63c7daf55c534cf82d5d':
lavf: Open PICT images with Quickdraw
Conflicts:
tests/fate/image.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If max in clean_index is set to a negative ast->sample_size, the
following loop never ends:
while (max < 1024)
max += max;
Thus set ast->sample_size to 0 if it would otherwise be negative.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If bit_rate is negative, it can trigger an av_assert2 in av_rescale_rnd.
Since av_rescale returns int64_t, but st->codec_bit_rate is int, it can
also overflow into a negative value.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes a segmentation fault.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
index_scale is set to matroska->time_scale of type uint64_t.
When index_scale is int, the assignment can overflow and e.g. result
in index_scale = 0. This causes a floating point exception due to the
division by index_scale.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fix a duplicate memory allocation. priv_data should be allocated
in line 64 call to avformat_alloc_output_context2 since we pass
the correct AVFormat to it. This removes the duplicate allocation.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Use dyn_duf to write chunks so that we create the actual chunk
file only after the entire chunk data is available. This will help
not confuse other software looking at the chunk file (e.g. a web
server) by seeing a zero length file when ffmpeg is writing into
it.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes Detecting AAC with such descriptor if the parts needed for detection
are later in the stream
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '82de8d71118f4eafd6a43e9ea9169bd411793798':
mpegts: Update the PSI/SI table only if the version change
Conflicts:
libavformat/mpegts.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The existing check has two problems:
1) i + count can overflow, so that the check '< 256' returns true.
2) In the (i == 'N') case occurs a j-- so that the loop runs once more.
This can trigger the assertion 'nut->header_len[0] == 0' or cause
segmentation faults or infinite hangs.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If a PAT is finished while a PMT section filter is opened but
not yet finished, the PMT section filter is closed and all
the received data is discarded.
This is usually not an issue but some multiplexers (With very
quick PAT/PMT repetition settings) consistently emit a PMT
section start, then a PAT, and then the rest of the PMT,
causing the aforementioned behavior to result in no PMT being
finished.
In the most pathologic situation the stream information are lost
and the probe fallback miscategorizes subtitles as mp3 audio.
Avoid the issue through eliminating redundant PSI/SI table
updates by checking their version field, which is required by
the standard to be incremented on every change no matter how
minor.
CC: libav-stable@libav.org
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
A negative frame rate triggers an av_assert2 in av_rescale_rnd.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Check extended sync word for 16-bit LE and BE core streams to reduce
probability of alias sync detection. Previously sync word extension was
checked only for 14-bit streams.
This follows up the similar change in avcodec/dca_parser.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Remove the direct profile from UTCTiming element. Per DASH spec,
direct profile value should be the time at which the request was
made to the server and not the time at which the manifest was
written. So ffmpeg cannot write this value. This patch removes
the direct profile and write the UTCTiming element with the http
profile only if a URL is passed as a parameter. Update the fate
test to reflect this change.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fix a bug where the chunk muxer doesn't write the very first audio
packet (with pts == 0).
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b90adb0aba073f9c1b4abca852119947393ced4c':
rtsp: Make sure we don't write too many transport entries into a fixed-size array
Merged-by: Michael Niedermayer <michaelni@gmx.at>
