Up until now, ff_alloc_packet2() has a min_size parameter:
It is supposed to be a lower bound on the final size of the packet
to allocate. If it is not too far from the upper bound (namely,
if it is at least half the upper bound), then ff_alloc_packet2()
already allocates the final, already refcounted packet; if it is
not, then the packet is not refcounted and its data only points to
a buffer owned by the AVCodecContext (in this case, the packet will
be made refcounted in encode_simple_internal() in libavcodec/encode.c).
The goal of this was to avoid data copies and intermediate buffers
if one has a precise lower bound.
Yet those encoders for which precise lower bounds exist have recently
been switched to ff_get_encode_buffer() (which automatically allocates
final buffers), leaving only two encoders to actually set the min_size
to something else than zero (namely aliaspixenc and hapenc). Both of
these encoders use a very low lower bound that is not helpful in any
nontrivial case.
This commit therefore removes the min_size parameter as well as the
codepath in ff_alloc_packet2() for the allocation of final buffers.
Furthermore, the function has been renamed to ff_alloc_packet() and
moved to encode.h alongside ff_get_encode_buffer().
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
AVCodecContext.extradata is freed generically by libavcodec for
encoders, so it is unnecessary for an encoder to do it on its own.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Given that the AVCodec.next pointer has now been removed, most of the
AVCodecs are not modified at all any more and can therefore be made
const (as this patch does); the only exceptions are the very few codecs
for external libraries that have a init_static_data callback.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The issue is that the afq still has samples as on init it counts
the overlap used as a delay to adjust the PTS it generates, hence
we can't rely on it right after init.
So just check to see if any frames have been encoded. frame_number
can't be anything but 0 right after init and can only be set by lavc.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit implements a psychoacoustic system for the native Opus
encoder. Its unlike any other psychoacoustic system known since its
capable of using a lookahead to make better choices on how to treat the
current frame and how many bits to allocate for it (and future frames).
Also, whilst the main bulk of the analysis function has to run in a
single thread, the per-frame anaylsis functions does not modify the main
psychoacoustic context, so in the future it will be fairly trivial to
run those as slice threads.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Squelches the following compiler warnings:
libavcodec/opusenc.c:1051:16: warning: format specifies type 'long' but
the argument has type 'long long' [-Wformat]
avctx->bit_rate/1000, clipped_rate/1000);
^~~~~~~~~~~~~~~~~~~~
libavcodec/opusenc.c:1051:38: warning: format specifies type 'long' but
the argument has type 'long long' [-Wformat]
avctx->bit_rate/1000, clipped_rate/1000);
^~~~~~~~~~~~~~~~~
This marks the first time anyone has written an Opus encoder without
using any libopus code. The aim of the encoder is to prove how far
the format can go by writing the craziest encoder for it.
Right now the encoder's basic, it only supports CBR encoding, however
internally every single feature the CELT layer has is implemented
(except the pitch pre-filter which needs to work well with the rest of
whatever gets implemented). Psychoacoustic and rate control systems are
under development.
The encoder takes in frames of 120 samples and depending on the value of
opus_delay the plan is to use the extra buffered frames as lookahead.
Right now the encoder will pick the nearest largest legal frame size and
won't use the lookahead, but that'll change once there's a
psychoacoustic system.
Even though its a pretty basic encoder its already outperforming
any other native encoder FFmpeg has by a huge amount.
The PVQ search algorithm is faster and more accurate than libopus's
algorithm so the encoder's performance is close to that of libopus
at zero complexity (libopus has more SIMD).
The algorithm might be ported to libopus or other codecs using PVQ in
the future.
The encoder still has a few minor bugs, like desyncs at ultra low
bitrates (below 9kbps with 20ms frames).
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Anton Khirnov
Andreas Rheinhardt
Andreas Rheinhardt
Lynne
Rostislav Pehlivanov
Kyle Swanson
Michael Niedermayer