The assumption that avcodec_send_packet makes regarding decoders
consuming the entire packet is not true if the codec supports
truncated decoding mode and the truncated flag is turned on.
Steps to reproduce:
./ffmpeg_g -flags truncated \
-i "http://samples.ffmpeg.org/MPEG2/test-ebu-422.40000.pakets.ts" \
-c:v ffv1 -c:a copy -y /tmp/truncated.nut
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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.
It's container level information on some formats (Matroska, MXF, yuv4mpeg), so
it should be printed at higher log levels than debug.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
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.
This makes it easier to use the lowres option when dealing with input
files in different codecs. If the codec doesn't support lowres=1 for
instance, it will throw a warning and use lowres=0 instead of erroring
out completely.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Currently it's exported as AVFrame.pkt_pts, which is also the only use
for that field. The reason it is done like this is that lavc used to
export various codec-specific "timing" information in AVFrame.pts, which
is not done anymore.
Since it is confusing to the callers to have a separate field which is
used only for decoder timestamps and nothing else, deprecate pkt_pts and
use just AVFrame.pts everywhere.
The deprecated avcodec_decode_video2() and avcodec_decode_audio4()
functions called av_packet_split_side_data() on the input packets. This
is required for packets produced by libavformat with the
AVFMT_FLAG_KEEP_SIDE_DATA flag unset (which is unfortunately the
default).
The new API didn't do this yet, although it didn't matter as no decoder
supports the new API yet. The emulation layer for the old API calls the
old API functions, which took care of the splitting. Add this code to
the new API codec entrypoints too, because we shouldn't send essentially
corrupted data to decoders.
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>
The bits_per_raw_sample represents the number of bits of precision per sample.
The field is added at the logical place, not at the end as the code was just
recently added
This fixes the regression about losing the audio sample precision information
The change in the fate test checksum un-does the change from the merge
Previous version reviewed by: wm4 <nfxjfg@googlemail.com>
Previous version reviewed by: Dominik 'Rathann' Mierzejewski <dominik@greysector.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Paul B Mahol
Pavel Koshevoy
Anton Khirnov
Michael Niedermayer
Vittorio Giovara
Sasi Inguva
Rodger Combs
James Almer
Kieran Kunhya
wm4
Marton Balint
Jon Toohill
Aman Gupta
Andrey Turkin
Peter Ross
Clément Bœsch
Hendrik Leppkes