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Currently, it is accessed via a pointer (ff_celt_window)
exported from opustab.h which points inside a static array
(ff_celt_window_padded) in opustab.h. Instead export
ff_celt_window_padded directly and make opustab.h
a static const pointer pointing inside ff_celt_window_padded.
Also mark all the declarations in opustab.h as hidden,
so that the compiler knows that ff_celt_window has a fixed
offset from the code even when compiling position-independent
code.
Reviewed-by: Lynne <dev@lynne.ee>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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Reviewed-by: Lynne <dev@lynne.ee>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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Move ROUND_MUL* macros to their only users and the Celt
macros to opus_celt.h. Also improve the other headers
a bit while at it.
Reviewed-by: Lynne <dev@lynne.ee>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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AVCodec.channel_layouts is deprecated and Clang (unlike GCC)
warns when setting this field in a codec definition.
Fortunately, Clang (unlike GCC) allows to use
FF_DISABLE_DEPRECATION_WARNINGS inside a definition (of an FFCodec),
so that one can create simple macros to set AVCodec.channel_layouts
that also suppress deprecation warnings for Clang.
(Notice that some of the codec definitions were already
inside FF_DISABLE/ENABLE_DEPRECATION_WARNINGS (that were not
guarded by FF_API_OLD_CHANNEL_LAYOUT); these have been removed.
Also notice that setting AVCodec.channel_layouts was not guarded
by FF_API_OLD_CHANNEL_LAYOUT either, so testing disabling it
it without removing all the codeblocks would not have worked.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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This commit changes both the encoder and decoder to use the new lavu/tx code,
which has faster C transforms and more assembly optimizations.
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It reduces typing: Before this patch, there were 105 codecs
whose long_name-definition exceeded the 80 char line length
limit. Now there are only nine of them.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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Up until now, these encoders received non-refcounted packets
(whose data was owned by the corresponding AVCodecContext)
from ff_alloc_packet(); these packets were made refcounted lateron
by av_packet_make_refcounted() generically.
This commit makes these encoders accept user-supplied buffers by
replacing av_packet_make_refcounted() with an equivalent function
that is based upon get_encode_buffer().
(I am pretty certain that one can also set the flag for mpegvideo-
based encoders, but I want to double-check this later. What is certain
is that it reallocates the buffer owned by the AVCodecContext
which should maybe be moved to encode.c, so that proresenc_kostya.c
and ttaenc.c can make use of it, too.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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and remove FF_CODEC_CAP_INIT_THREADSAFE
All our native codecs are already init-threadsafe
(only wrappers for external libraries and hwaccels
are typically not marked as init-threadsafe yet),
so it is only natural for this to also be the default state.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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This is possible, because every given FFCodec has to implement
exactly one of these. Doing so decreases sizeof(FFCodec) and
therefore decreases the size of the binary.
Notice that in case of position-independent code the decrease
is in .data.rel.ro, so that this translates to decreased
memory consumption.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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This structure is no longer declared in a public header,
so using an FF-prefix is more appropriate.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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Up until now, codec.h contains both public and private parts
of AVCodec. This exposes the internals of AVCodec to users
and leads them into the temptation of actually using them
and forces us to forward-declare structures and types that
users can't use at all.
This commit changes this by adding a new structure FFCodec to
codec_internal.h that extends AVCodec, i.e. contains the public
AVCodec as first member; the private fields of AVCodec are moved
to this structure, leaving codec.h clean.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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Also move FF_CODEC_TAGS_END as well as struct AVCodecDefault.
This reduces the amount of files that have to include internal.h
(which comes with quite a lot of indirect inclusions), as e.g.
most encoders don't need it. It is furthemore in preparation
for moving the private part of AVCodec out of the public codec.h.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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Signed-off-by: James Almer <jamrial@gmail.com>
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Also include channel_layout.h directly wherever used.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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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>
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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>
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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>
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They are not properly namespaced and not intended for public use.
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By popular request.
Does the same as in libopusenc.
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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>
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Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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There's no difference apart from which entropy coding functions get called.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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No point in having the same code twice to do exactly the same thing.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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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>
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Just put the subtraction in the table.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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Also fixes transient windowing
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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vector_fmul_reverse requires padding the window at the front
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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Its only use is to adjust the aac psychoacoustic/filter system which
isn't used here.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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Fixes valgrind warnings, didn't affect anything since it was only used
for resynthesis.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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Saves around 5kbps.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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Apparently its only use is to enable comfort noise/error recovery.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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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);
^~~~~~~~~~~~~~~~~
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Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Fixes CID1400584
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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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>
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