The slice index expected by D3D11VA is the one from the texture not from the
array or texture/slices.
In VLC the slices we provide the decoder don't start from 0 and thus pictures
appear in bogus order. With possible crashes and corruptions when using an
invalid index.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Right now, if we attempt to use cuvid in a media player and then
try to seek, the decoder will happily pass out whatever frames were
already in flight before the seek.
There is both the output queue in our code and some number of frames
within the cuvid decoder that need to be accounted for.
cuvid doesn't support flush, so our only choice is to do a brute-force
re-creation of the decoder, which also implies re-creating the parser,
but this is fine.
The only subtlty is that there is sanity check code in decoder
initialisation that wants to make sure the HWContextFrame hasn't already
been initialised. This is a fair check to do at the beginning but not
after a flush, so it has to be made conditional.
Signed-off-by: Philip Langdale <philipl@overt.org>
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
cuvid/nvdecode also supports mpeg1, mpeg2, h.263/mpeg4-asp and mjpeg.
It should, in theory, also support wmv3 via the vc1 support, given
that vdpau supports this. However, it failed to play wmv3 samples
which vdpau played correctly, so I'm not sure what to make of it.
Signed-off-by: Philip Langdale <philipl@overt.org>
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
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.
This avoids enabling and building the x264rgb encoder when its actually not supported and
thus would not work
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This function is supposed to "reset" a codec context to a clean state so
that it can be opened again. The only reason it exists is to allow using
AVStream.codec as a decoding context (after it was already
opened/used/closed by avformat_find_stream_info()). Since that behaviour
is now deprecated, there is no reason for this function to exist
anymore.
Since AVCodecContext contains a lot of complex state, copying a codec
context is not a well-defined operation. The purpose for which it is
typically used (which is well-defined) is copying the stream parameters
from one codec context to another. That is now possible with through the
AVCodecParameters API. Therefore, there is no reason for
avcodec_copy_context() to exist.
Register mmaldec as mpeg2 decoder. Supporting mpeg2 in mmaldec is just a
matter of setting the correct MMAL_ENCODING on the input port. To ease the
addition of further supported mmal codecs a macro is introduced to generate
the decoder and decoder class structs.
Signed-off-by: Julian Scheel <julian@jusst.de>
Signed-off-by: wm4 <nfxjfg@googlemail.com>
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.
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>
Matthieu Bouron
Timo Rothenpieler
Steve Lhomme
Sasi Inguva
Carl Eugen Hoyos
Yogender Gupta
Rostislav Pehlivanov
Philip Langdale
Oliver Collyer
Paul B Mahol
Jon Toohill
Jan Sebechlebsky
Martin Storsjö
Anton Khirnov
Vittorio Giovara
Neil Birkbeck
Michael Niedermayer
Diego Biurrun
Julian Scheel
wm4