So, all frames and errors are correctly reported in order.
Also limit the numbers of error during draining to prevent infinite loop.
This fix fate failure with THREADS>=4:
make fate-h264-attachment-631 THREADS=4
This also reverts a755b725ec.
Suggested-by: wm4, Ronald S. Bultje, Marton Balint
Reviewed-by: w4 <nfxjfg@googlemail.com>
Reviewed-by: Ronald S. Bultje <rsbultje@gmail.com>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
Allows to get a more realistic total bitrate (and estimated file size)
in avi_write_header. Previously a static default value of 200k was
assumed.
Adds an internal helper function for bitrate guessing.
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Apparently the demuxer outputs the wrong padding for HE-AAC (based on
the raw sample rate, or so). aacdec contains a hack to adjust the muxer
padding accordingly before it's used to trim the decoder output. This
modified the packet side data, which in combination with the old
decoding API would change the packet the user passed to the decoder.
This is clearly not allowed, and it breaks running some gapless fate
tests with "-fflags +keepside" applied (without keepside, the packet
metadata is typically newly allocated, essentially making a copy and not
modifying the user's input packet).
This should probably be fixed in the demuxer (and consequently also the
muxer), but for now only fix the immediate problem.
Regression since 946ed78f5f (2012).
Currently, the new decoding API is pretty much just a wrapper around the
old deprecated one. This is problematic, since it interferes with making
full use of the flexibility added by the new API. The old API should
also be removed at some future point.
Reorganize the code so that the new send_packet/receive_frame functions
call the actual decoding directly and change the old deprecated
avcodec_decode_* functions into wrappers around the new API.
The new internal API for decoders is now changing as well. Before this
commit, it mirrors the public API, so the decoders need to implement
send_packet() and receive_frame() callbacks. This turns out to require
awkward constructs in both the decoders and the generic code. After this
commit, the decoders only implement the receive_frame() callback and
call a new internal function, ff_decode_get_packet() to obtain input
data, in the same manner to how the bitstream filters now work.
avcodec will now always make a reference to the input packet, which means
that non-refcounted input packets will be copied. Keeping the previous
behaviour, where this copy could sometimes be avoided, would make the
code significantly more complex and fragile for only dubious gains,
since packets are typically small and everyone who cares about
performance should use refcounted packets anyway.
The current code stores a pointer to the packet passed to the decoder,
which is then used during get_buffer() for timestamps and side data
passthrough. However, since this is a pointer to user data which we do
not own, storing it is potentially dangerous. It is also ill defined for
the new decoding API with split input/output.
Fix this problem by making an explicit internally owned copy of the
packet properties.
With the new decode API, doing this in ffmpeg.c is impractical. There
was resistance against removing the warning, so put it into libavcodec.
Not bothering with reducing the warning to verbose log level for
subsequent wanrings. The warning should be rare, and only happen when
developing new codecs for the old API.
Includes a change suggested by Michael Niedermayer.
Until now, the decoding API was restricted to outputting 0 or 1 frames
per input packet. It also enforces a somewhat rigid dataflow in general.
This new API seeks to relax these restrictions by decoupling input and
output. Instead of doing a single call on each decode step, which may
consume the packet and may produce output, the new API requires the user
to send input first, and then ask for output.
For now, there are no codecs supporting this API. The API can work with
codecs using the old API, and most code added here is to make them
interoperate. The reverse is not possible, although for audio it might.
From Libav commit 05f66706d1.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Until now, the decoding API was restricted to outputting 0 or 1 frames
per input packet. It also enforces a somewhat rigid dataflow in general.
This new API seeks to relax these restrictions by decoupling input and
output. Instead of doing a single call on each decode step, which may
consume the packet and may produce output, the new API requires the user
to send input first, and then ask for output.
For now, there are no codecs supporting this API. The API can work with
codecs using the old API, and most code added here is to make them
interoperate. The reverse is not possible, although for audio it might.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
It is only used inside libavcodec.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
The generic code in utils.c sets the AVFrame.pkt_dts field from the
packet it was supposedly decoded. This does not have to be true for a
fully asynchronous decoder like mmaldec. It could be overwritten with an
incorrect value. Even if the decoder doesn't determine the DTS (but sets
it to AV_NOPTS_VALUE), it's impossible to determine a correct value in
utils.c.
Decoders can now be marked with FF_CODEC_CAP_SETS_PKT_DTS, in which case
utils.c won't overwrite the field. The decoders are expected to set this
field (even if they only set it to AV_NOPTS_VALUE).
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This parameter can be used to inform the allocation code about how much
downsizing might occur, and can be used to optimize how to allocate the
packet
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The stats are a superset of the quality factor, also allowing the picture type and encoder "PSNR" stats to be exported
This also replaces the native by fixed little endian order for the affected side data
AV_PKT_DATA_QUALITY_FACTOR is left as a synonym of AV_PKT_DATA_QUALITY_STATS
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
They are used by dnxhd and mpegvideo_enc exclusively, move them to codec
private options instead.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
This field is designed for marking codec properties useful to lavc internals.
Two internal capabilities are added:
- FF_CODEC_CAP_INIT_THREADSAFE: codec can be opened without locks;
- FF_CODEC_CAP_INIT_CLEANUP: codec frees memory if initialization fails.
Signed-off-by: James Almer <jamrial@gmail.com>
Reviewed-by: "Ronald S. Bultje" <rsbultje@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Michael Niedermayer
James Almer
Muhammad Faiz
Tobias Rapp
Carl Eugen Hoyos
wm4
Anton Khirnov
Thierry Foucu
Andrey Turkin
Ganesh Ajjanagadde
Luca Barbato
Matthieu Bouron
Andreas Cadhalpun
Derek Buitenhuis
Hendrik Leppkes
Ronald S. Bultje
Vittorio Giovara
Michael Niedermayer
Martin Storsjö
Justin Ruggles