This commit does for AVInputFormat what commit
59c9dc82f4 did for AVOutputFormat:
It adds a new type FFInputFormat, moves all the internals
of AVInputFormat to it and adds a now reduced AVInputFormat
as first member.
This does not affect/improve extensibility of both public
or private fields for demuxers (it is still a mess due to lavd).
This is possible since 50f34172e0
(which removed the last usage of an internal field of AVInputFormat
in fftools).
(Hint: tools/probetest.c accesses the internals of FFInputFormat
as well, but given that it is a testing tool this is not considered
a problem.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Unnecessary since acf63d5350adeae551d412db699f8ca03f7e76b9;
also avoids relocations.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
avio_seek() is called inside check(). Seeking to 'off' then seeking
to 'off + i' is unefficient, and it can loop 64 * 1024 times in the
worst case. When probe a malformed file over HTTP, it looks like
stucked forvever. ffio_ensure_seekback() doesn't solve the issue
when the stream is seekable but slow.
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
Do this by allocating AVStream together with the data that is
currently in AVStreamInternal; or rather: Put AVStream at the
beginning of a new structure called FFStream (which encompasses
more than just the internal fields and is a proper context in its own
right, hence the name) and remove AVStreamInternal altogether.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Do this by allocating AVFormatContext together with the data that is
currently in AVFormatInternal; or rather: Put AVFormatContext at the
beginning of a new structure called FFFormatContext (which encompasses
more than just the internal fields and is a proper context in its own
right, hence the name) and remove AVFormatInternal altogether.
The biggest simplifications occured in avformat_alloc_context(), where
one can now simply call avformat_free_context() in case of errors.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Currently AVIOContext's private fields are all over AVIOContext.
This commit moves them into a new structure in avio_internal.h instead.
Said structure contains the public AVIOContext as its first element
in order to avoid having to allocate a separate AVIOContextInternal
which is costly for those use cases where one just wants to access
an already existing buffer via the AVIOContext-API.
For these cases ffio_init_context() can't fail and always returned zero,
which was typically not checked. Therefore it has been made to not
return anything.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This is possible now that the next-API is gone.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Those are private fields, no reason to have them exposed in a public
header. Since there are some (semi-)public fields located after these,
even though this section is supposed to be private, keep some dummy
padding there until the next major bump to preserve ABI compatibility.
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.
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>
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>
This massively reduces the detection of random data as low score mp3
It may improve security by making it harder to read non multimedia data
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The number of bits from bit #m to #n is n - m plus 1.
Signed-off-by: Ingo Brückl <ib@wupperonline.de>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
MPEG Audio frame header must be 4 bytes. If we fail to read
4 bytes bail early to avoid Use-of-uninitialized-value msan error.
Reference https://crbug.com/666874.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
"Skipping 0 bytes of junk" is useless to the user, and essentially
indicates a NOP. At 0 bytes, this message is now pushed back to
the verbose log level.
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
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.
* commit '955aec3c7c7be39b659197e1ec379a09f2b7c41c':
mpegaudiodecheader: check the header in avpriv_mpegaudio_decode_header
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
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.
"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 2b3e9bbfb5 caused problems for a
certain API user:
https://code.google.com/p/chromium/issues/detail?id=537725https://code.google.com/p/chromium/issues/detail?id=542032
The problem seems rather arbitrary, because if there's junk, anything
can happen. In this case, the imperfect junk skipping just caused it to
read different junk, from what I can see.
We can improve the accuracy of junk detection by a lot by checking if 2
consecutive frames use the same configuration. While in theory it might
be completely fine for the 1st frame to have a different format than the
2nd frame, it's exceedingly unlikely, and I can't think of a legitimate
use-case.
This is approximately the same mpg123 does for junk skipping. The
set of compared header bits is the same as the libavcodec mp3 parser
uses for similar purposes.
When AVFMT_FLAG_FAST_SEEK is specified, make MP3 seek operation as
fast as possible.
When no "-usetoc" is specified, the default operation is using TOC
if available; otherwise, uses linear interpolation. This is useful
when seeking a large MP3 file with no TOC available. One example is
Podcast, many MP3 files are large, but no CBR/VBR tags. Most of
them are actually CBR. Even in VBR cases, this option sacrifices the
accuracy of playback time in exchange for responsiveness.
Andreas Rheinhardt