This commit changes a few things about the noise substitution
logic:
- Brings back the quantization factor (reduced to 3) during
scalefactor index calculations.
- Rejects any zeroed bands. They should be inaudiable and it's
a waste transmitting the scalefactor indices for these.
- Uses swb_offsets instead of incrementing a 'start' with every
window group size.
- Rejects all PNS during short windows.
Overall improves quality. There was a plan to use the lfg system
to create the random numbers instead of using whatever the decoder
uses but for now this works fine. Entropy is far from important here.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit once again improves the PNS implementation by scaling the
thresholds with frequency. The thresholds get looser as the frequency
increases since higher frequencies are basically noise to human ears.
Also, this introduces quantization error correction for PNS. Should
the error be too much, no PNS will be used. The energy_ratio is used
to regulate the actual encoded PNS energy: if the generated PNS
energy is higher than the energy from the psy system, energy_ratio
is used to correct it so that hopefully once requantized and
transmitted the value in the decoder will be closer to what the
encoder has.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This was an oversight when the IS system was being first implemented.
The ener01 part was largely a result of trial and error and the fact
that the sum of coef0 and coef1 could result in a zero was
overlooked. Once ener01 turns to zero it's used to divide the left
channel energy which doesn't turn out so well as it fills IS[]
with -nan's and inf's which in turn confused the quantize_band_cost.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
TNS had both IS and PNS switched on when it makes more sense
to have them both off.
Prediction had a redundant argument.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
IS and PNS increase quality a ton so as a result the PSNR changed.
Disable the extensions and keep the tests separate such that there
will be no red herrings if one test fails.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Without this fate-filter-join failes with
FF_API_GET_CHANNEL_LAYOUT_COMPAT disabled.
Reviewed-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
This fixes fate with FF_API_LAVF_BITEXACT disabled.
Reviewed-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.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>
Works only for flv, h263 and huffyuv decoders.
Makes only one pass through the file (this should be changed to two passes)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This fixes fate with FF_API_REQUEST_CHANNELS disabled.
Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Works only with video stream.
First pass without seeking -- counts crcs of a frames and store it in an array.
After that it seeks a lot in different places and checks if crcs of these frames and crcs of frames in array are the same.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit '58c3720a3cc71142b5d48d8ccdc9213f9a66cd33':
fate: Make sure a corner-case for ASF is covered
Adjusted fate ref to match the different timebase of the ffasf demuxer
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
Compute individual stream durations in matroska muxer.
Write them as string tags in the same format as mkvmerge tool does.
Signed-off-by: Sasi Inguva <isasi@google.com>
* commit 'a0797950527120c85263c910eb6ba08fddcfdcb3':
fate/mp3: specify the number of output samples instead of filesize
Merged-by: Michael Niedermayer <michael@niedermayer.cc>
Avoid clipping due to quantization noise to produce audible
artifacts, by detecting near-clipping signals and both attenuating
them a little and encoding escape-encoded bands (usually the
loudest) rounding towards zero instead of nearest, which tends to
decrease overall energy and thus clipping.
Currently fate tests measure numerical error so this change makes
tests using asynth (which are near clipping) report higher error
not less, because of window attenuation. Yet, they sound better,
not worse (albeit subtle, other samples aren't subtle at all).
Only measuring psychoacoustically weighted error would make for
a representative test, so that will be left for a future patch.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
