It also made no sense to actually make the filter span the entire
window including the first band of the next window.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Tests fails on some ARM builds but it's close enough so it's okay.
NEON, half-precision floats, rounding errors, who knows.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit introduces a test for AAC-Main prediction
which was just reworked in this series of commits.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Pulses are already on the way so expect to see the list
gone in the close future.
TNS is already of sufficiently high quality to be enabled
by default (but isn't yet, so you too can help by testing!).
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit abandons the way the specifications state to
quantize the coefficients, makes use of the new LPC float
functions and is much better.
The original way of converting non-normalized float samples
to int32_t which out LPC system expects was wrong and it was
wrong to assume the coefficients that are generated are also
valid. It was essentially a full garbage-in, garbage-out
system and it definitely shows when looking at spectrals
and listening. The high frequencies were very overattenuated.
The new LPC function performs the analysis directly.
The specifications state to quantize the coefficients into
four bit index values using an asin() function which of course
had to have ugly ternary operators because the function turns
negative if the coefficients are negative which when encoding
causes invalid bitstream to get generated.
This deviates from this by using the direct TNS tables, which
are fairly small since you only have 4 bits at most for index
values. The LPC values are directly quantized against the tables
and are then used to perform filtering after the requantization,
which simply fetches the array values.
The end result is that TNS works much better now and doesn't
attenuate anything but the actual signal, e.g. TNS removes
quantization errors and does it's job correctly now.
It might be enabled by default soon since it doesn't hurt and
helps reduce nastyness at low bitrates.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit removes the array which was made redundant with
the last commit. The current prediction system gets the
quantization error directly (and without the single-frame delay)
in the search_for_pred function.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit completely alters the algorithm of prediction.
The original commit which introduced prediction was completely
incorrect to even remotely care about what the actual coefficients
contain or whether any options were enabled. Not my actual fault.
This commit treats prediction the way the decoder does and expects
to do: like lossy encryption. Everything related to prediction now
happens at the very end but just before quantization and encoding
of coefficients. On the decoder side, prediction happens before
anything has had a chance to even access the coefficients.
Also the original implementation had problems because it actually
touched the band_type of special bands which already had their
scalefactor indices marked and it's a wonder the asserion wasn't
triggered when transmitting those.
Overall, this now drastically increases audio quality and you should
think about enabling it if you don't plan on playing anything encoded
on really old low power ultra-embedded devices since they might not
support decoding of prediction or AAC-Main. Though the specifications
were written ages ago and as times change so do the FLOPS.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This was missed when the original commits were done. FF_PROFILE_UNKNOWN
is what's in avctx->profile when no audio profile is specified.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit simply duplicates the functionality of ff_lpc_calc_coefs()
for the case of a Levinson-Durbin LPC with the only difference being
that floating point samples are accepted and the resulting coefficients
are raw and unquantized.
The motivation behind doing this is the fact that the AAC encoder
requires LPC in TNS and LTP and converting non-normalized floating
point coefficients to int32_t using SWR and again back for the LPC
coefficients was very impractical.
The current LPC interfaces were designed for int32_t in mind possibly
because FLAC and ALAC use this type for most internal operations.
The mathematics in case of floats remains of course identical.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit simply moves the TNS tables to a more appropriate
aactab.h since then they can be accessed by both the decoder
and encoder.
The encoder _shouldn't_ normally need the tables since the
specs describe a specific quantization process, but the exact
reason for this can be seen in the TNS commit following.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
At least for vdpau, the hwaccel init code tries to check the video
profile and ensure that there is a matching vdpau profile available.
If it can't find a match, it will fail to initialise.
In the case of wmv3/vc1, I observed initialisation to fail all the
time. It turns out that this is due to the hwaccel being initialised
very early in the codec init, before the profile has been extracted
and set.
Conceptually, it's a simple fix to reorder the init code, but it gets
messy really fast because ff_get_format(), which is what implicitly
trigger hwaccel init, is called multiple times through various shared
init calls from h263, etc. It's incredibly hard to prove to my own
satisfaction that it's safe to move the vc1 specific init code
ahead of this generic code, but all the vc1 fate tests pass, and I've
visually inspected a couple of samples and things seem correct.
Signed-off-by: Philip Langdale <philipl@overt.org>
This prevents breaking existing command lines in case the "ab" default is removed from libavcodec
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The amv one probably looks suspicious, but since it's an intra-only
codec, I couldn't possibly imagine what it would use the edge for,
and the vsynth fate result doesn't change, so it's probably OK.
The current algorithm is just "try all the combinations, and pick the best".
It's not very fast either, probably due to a lot of copying, but will do for
an initial implementation.
Signed-off-by: Donny Yang <work@kota.moe>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Better message that ffplay is not going to be built by printing out what
will be built.
Based on a patch by Moritz Barsnick.
Signed-off-by: rogerdpack <rogerpack2005@gmail.com>
Reviewed-by: Ganesh Ajjanagadde <gajjanag@mit.edu>
Signed-off-by: Timothy Gu <timothygu99@gmail.com>
Assert on `avctx->codec->encode2` to avoid a SEGFAULT on the subsequent
function call.
avcodec_encode_video2() uses a similar assertion.
Calling the wrong function on a stream is a serious inconsistency
which could at other places be potentially dangerous and exploitable,
it is thus safer to stop execution and not continue with such
inconsistency after returning an error.
Commit-message-extended-by commiter
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The wiki, Ticket1464, and Ticket3970 warn about the usage of GCC 4.2.
This fixes Ticket3970.
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
* commit '413d4e54a9bffe2d0afdc6d8a80f516e5df6a421':
nvenc: Properly free the fifos
Not merged, ffmpeg's nvenc doesn't use these fifos.
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* commit '2157df425bd909854fd4afcec4aa3311d8a3b31b':
hlsenc: Support outputting specific versions
Not merged, the version is auto-selected in ffmpeg based on enabled features.
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
