[FFmpeg-cvslog] avfilter: add afwtdn filter
Paul B Mahol
git at videolan.org
Sat Jul 24 13:34:22 EEST 2021
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Sun May 31 10:22:23 2020 +0200| [6846d48fa64d077c0b5e7786d4a9d49a3d81025d] | committer: Paul B Mahol
avfilter: add afwtdn filter
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=6846d48fa64d077c0b5e7786d4a9d49a3d81025d
---
Changelog | 1 +
doc/filters.texi | 60 +++
libavfilter/Makefile | 1 +
libavfilter/af_afwtdn.c | 1349 ++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
6 files changed, 1413 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index 606a9d3edb..1037688682 100644
--- a/Changelog
+++ b/Changelog
@@ -8,6 +8,7 @@ version <next>:
- Argonaut Games CVG demuxer
- Argonaut Games CVG muxer
- Concatf protocol
+- afwtdn audio filter
version 4.4:
diff --git a/doc/filters.texi b/doc/filters.texi
index 1b1db2ebb8..7c1d3e49ae 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1493,6 +1493,66 @@ Default value is 1.0.
This filter supports the all above options as @ref{commands}.
+ at section afwtdn
+Reduce broadband noise from input samples using Wavelets.
+
+A description of the accepted options follows.
+
+ at table @option
+ at item sigma
+Set the noise sigma, allowed range is from 0 to 1.
+Default value is 0.
+This option controls strength of denoising applied to input samples.
+Most useful way to set this option is via decibels, eg. -45dB.
+
+ at item levels
+Set the number of wavelet levels of decomposition.
+Allowed range is from 1 to 12.
+Default value is 10.
+Setting this too low make denoising performance very poor.
+
+ at item wavet
+Set wavelet type for decomposition of input frame.
+They are sorted by number of coefficients, from lowest to highest.
+More coefficients means worse filtering speed, but overall better quality.
+Available wavelets are:
+
+ at table @samp
+ at item sym2
+ at item sym4
+ at item rbior68
+ at item deb10
+ at item sym10
+ at item coif5
+ at item bl3
+ at end table
+
+ at item percent
+Set percent of full denoising. Allowed range is from 0 to 100 percent.
+Default value is 85 percent or partial denoising.
+
+ at item profile
+If enabled, first input frame will be used as noise profile.
+If first frame samples contain non-noise performance will be very poor.
+
+ at item adaptive
+If enabled, input frames are analyzed for presence of noise.
+If noise is detected with high possibility then input frame profile will be
+used for processing following frames, until new noise frame is detected.
+
+ at item samples
+Set size of single frame in number of samples. Allowed range is from 512 to
+65536. Default frame size is 8192 samples.
+
+ at item softness
+Set softness applied inside thresholding function. Allowed range is from 0 to
+10. Default softness is 1.
+ at end table
+
+ at subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
@section agate
A gate is mainly used to reduce lower parts of a signal. This kind of signal
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 62ee3d7b67..49c0c8342b 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -53,6 +53,7 @@ OBJS-$(CONFIG_AFFTFILT_FILTER) += af_afftfilt.o
OBJS-$(CONFIG_AFIR_FILTER) += af_afir.o
OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
OBJS-$(CONFIG_AFREQSHIFT_FILTER) += af_afreqshift.o
+OBJS-$(CONFIG_AFWTDN_FILTER) += af_afwtdn.o
OBJS-$(CONFIG_AGATE_FILTER) += af_agate.o
OBJS-$(CONFIG_AIIR_FILTER) += af_aiir.o
OBJS-$(CONFIG_AINTEGRAL_FILTER) += af_aderivative.o
diff --git a/libavfilter/af_afwtdn.c b/libavfilter/af_afwtdn.c
new file mode 100644
index 0000000000..45217f1afe
--- /dev/null
+++ b/libavfilter/af_afwtdn.c
@@ -0,0 +1,1349 @@
+/*
+ * Copyright (c) 2020 Paul B Mahol
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <float.h>
+
+#include "libavutil/avassert.h"
+#include "libavutil/avstring.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "filters.h"
+#include "formats.h"
+
+enum WaveletTypes {
+ SYM2,
+ SYM4,
+ RBIOR68,
+ DEB10,
+ SYM10,
+ COIF5,
+ BL3,
+ NB_WAVELET_TYPES,
+};
+
+/*
+ * All wavelets coefficients are taken from: http://wavelets.pybytes.com/
+ */
+
+static const double bl3_lp[42] = {
+ 0.000146098, -0.000232304, -0.000285414, 0.000462093, 0.000559952,
+ -0.000927187, -0.001103748, 0.00188212, 0.002186714, -0.003882426,
+ -0.00435384, 0.008201477, 0.008685294, -0.017982291, -0.017176331,
+ 0.042068328, 0.032080869, -0.110036987, -0.050201753, 0.433923147,
+ 0.766130398, 0.433923147, -0.050201753, -0.110036987, 0.032080869,
+ 0.042068328, -0.017176331, -0.017982291, 0.008685294, 0.008201477,
+ -0.00435384, -0.003882426, 0.002186714, 0.00188212, -0.001103748,
+ -0.000927187, 0.000559952, 0.000462093, -0.000285414, -0.000232304,
+ 0.000146098, 0.0,
+};
+
+static const double bl3_hp[42] = {
+ 0.0, 0.000146098, 0.000232304, -0.000285414, -0.000462093, 0.000559952,
+ 0.000927187, -0.001103748, -0.00188212, 0.002186714, 0.003882426,
+ -0.00435384, -0.008201477, 0.008685294, 0.017982291, -0.017176331,
+ -0.042068328, 0.032080869, 0.110036987, -0.050201753, -0.433923147,
+ 0.766130398, -0.433923147, -0.050201753, 0.110036987, 0.032080869,
+ -0.042068328, -0.017176331, 0.017982291, 0.008685294, -0.008201477,
+ -0.00435384, 0.003882426, 0.002186714, -0.00188212, -0.001103748,
+ 0.000927187, 0.000559952, -0.000462093, -0.000285414, 0.000232304,
+ 0.000146098,
+};
+
+static const double bl3_ilp[42] = {
+ 0.0, 0.000146098, -0.000232304, -0.000285414, 0.000462093, 0.000559952,
+ -0.000927187, -0.001103748, 0.00188212, 0.002186714, -0.003882426,
+ -0.00435384, 0.008201477, 0.008685294, -0.017982291, -0.017176331,
+ 0.042068328, 0.032080869, -0.110036987, -0.050201753, 0.433923147,
+ 0.766130398, 0.433923147, -0.050201753, -0.110036987, 0.032080869,
+ 0.042068328, -0.017176331, -0.017982291, 0.008685294, 0.008201477,
+ -0.00435384, -0.003882426, 0.002186714, 0.00188212, -0.001103748,
+ -0.000927187, 0.000559952, 0.000462093, -0.000285414, -0.000232304,
+ 0.000146098,
+};
+
+static const double bl3_ihp[42] = {
+ 0.000146098, 0.000232304, -0.000285414, -0.000462093, 0.000559952,
+ 0.000927187, -0.001103748, -0.00188212, 0.002186714, 0.003882426,
+ -0.00435384, -0.008201477, 0.008685294, 0.017982291, -0.017176331,
+ -0.042068328, 0.032080869, 0.110036987, -0.050201753, -0.433923147,
+ 0.766130398, -0.433923147, -0.050201753, 0.110036987, 0.032080869,
+ -0.042068328, -0.017176331, 0.017982291, 0.008685294, -0.008201477,
+ -0.00435384, 0.003882426, 0.002186714, -0.00188212, -0.001103748,
+ 0.000927187, 0.000559952, -0.000462093, -0.000285414, 0.000232304,
+ 0.000146098,
+};
+
+static const double sym10_lp[20] = {
+ 0.0007701598091144901, 9.563267072289475e-05,
+ -0.008641299277022422, -0.0014653825813050513,
+ 0.0459272392310922, 0.011609893903711381,
+ -0.15949427888491757, -0.07088053578324385,
+ 0.47169066693843925, 0.7695100370211071,
+ 0.38382676106708546, -0.03553674047381755,
+ -0.0319900568824278, 0.04999497207737669,
+ 0.005764912033581909, -0.02035493981231129,
+ -0.0008043589320165449, 0.004593173585311828,
+ 5.7036083618494284e-05, -0.0004593294210046588,
+};
+
+static const double sym10_hp[20] = {
+ 0.0004593294210046588, 5.7036083618494284e-05,
+ -0.004593173585311828, -0.0008043589320165449,
+ 0.02035493981231129, 0.005764912033581909,
+ -0.04999497207737669, -0.0319900568824278,
+ 0.03553674047381755, 0.38382676106708546,
+ -0.7695100370211071, 0.47169066693843925,
+ 0.07088053578324385, -0.15949427888491757,
+ -0.011609893903711381, 0.0459272392310922,
+ 0.0014653825813050513, -0.008641299277022422,
+ -9.563267072289475e-05, 0.0007701598091144901,
+};
+
+static const double sym10_ilp[20] = {
+ -0.0004593294210046588, 5.7036083618494284e-05,
+ 0.004593173585311828, -0.0008043589320165449,
+ -0.02035493981231129, 0.005764912033581909,
+ 0.04999497207737669, -0.0319900568824278,
+ -0.03553674047381755, 0.38382676106708546,
+ 0.7695100370211071, 0.47169066693843925,
+ -0.07088053578324385, -0.15949427888491757,
+ 0.011609893903711381, 0.0459272392310922,
+ -0.0014653825813050513, -0.008641299277022422,
+ 9.563267072289475e-05, 0.0007701598091144901,
+};
+
+static const double sym10_ihp[20] = {
+ 0.0007701598091144901, -9.563267072289475e-05,
+ -0.008641299277022422, 0.0014653825813050513,
+ 0.0459272392310922, -0.011609893903711381,
+ -0.15949427888491757, 0.07088053578324385,
+ 0.47169066693843925, -0.7695100370211071,
+ 0.38382676106708546, 0.03553674047381755,
+ -0.0319900568824278, -0.04999497207737669,
+ 0.005764912033581909, 0.02035493981231129,
+ -0.0008043589320165449, -0.004593173585311828,
+ 5.7036083618494284e-05, 0.0004593294210046588,
+};
+
+static const double rbior68_lp[18] = {
+ 0.0, 0.0, 0.0, 0.0,
+ 0.014426282505624435, 0.014467504896790148,
+ -0.07872200106262882, -0.04036797903033992,
+ 0.41784910915027457, 0.7589077294536541,
+ 0.41784910915027457, -0.04036797903033992,
+ -0.07872200106262882, 0.014467504896790148,
+ 0.014426282505624435, 0.0, 0.0, 0.0,
+};
+
+static const double rbior68_hp[18] = {
+ -0.0019088317364812906, -0.0019142861290887667,
+ 0.016990639867602342, 0.01193456527972926,
+ -0.04973290349094079, -0.07726317316720414,
+ 0.09405920349573646, 0.4207962846098268,
+ -0.8259229974584023, 0.4207962846098268,
+ 0.09405920349573646, -0.07726317316720414,
+ -0.04973290349094079, 0.01193456527972926,
+ 0.016990639867602342, -0.0019142861290887667,
+ -0.0019088317364812906, 0.0,
+};
+
+static const double rbior68_ilp[18] = {
+ 0.0019088317364812906, -0.0019142861290887667,
+ -0.016990639867602342, 0.01193456527972926,
+ 0.04973290349094079, -0.07726317316720414,
+ -0.09405920349573646, 0.4207962846098268,
+ 0.8259229974584023, 0.4207962846098268,
+ -0.09405920349573646, -0.07726317316720414,
+ 0.04973290349094079, 0.01193456527972926,
+ -0.016990639867602342, -0.0019142861290887667,
+ 0.0019088317364812906, 0.0,
+};
+
+static const double rbior68_ihp[18] = {
+ 0.0, 0.0, 0.0, 0.0,
+ 0.014426282505624435, -0.014467504896790148,
+ -0.07872200106262882, 0.04036797903033992,
+ 0.41784910915027457, -0.7589077294536541,
+ 0.41784910915027457, 0.04036797903033992,
+ -0.07872200106262882, -0.014467504896790148,
+ 0.014426282505624435, 0.0, 0.0, 0.0,
+};
+
+static const double coif5_lp[30] = {
+ -9.517657273819165e-08, -1.6744288576823017e-07,
+ 2.0637618513646814e-06, 3.7346551751414047e-06,
+ -2.1315026809955787e-05, -4.134043227251251e-05,
+ 0.00014054114970203437, 0.00030225958181306315,
+ -0.0006381313430451114, -0.0016628637020130838,
+ 0.0024333732126576722, 0.006764185448053083,
+ -0.009164231162481846, -0.01976177894257264,
+ 0.03268357426711183, 0.0412892087501817,
+ -0.10557420870333893, -0.06203596396290357,
+ 0.4379916261718371, 0.7742896036529562,
+ 0.4215662066908515, -0.05204316317624377,
+ -0.09192001055969624, 0.02816802897093635,
+ 0.023408156785839195, -0.010131117519849788,
+ -0.004159358781386048, 0.0021782363581090178,
+ 0.00035858968789573785, -0.00021208083980379827,
+};
+
+static const double coif5_hp[30] = {
+ 0.00021208083980379827, 0.00035858968789573785,
+ -0.0021782363581090178, -0.004159358781386048,
+ 0.010131117519849788, 0.023408156785839195,
+ -0.02816802897093635, -0.09192001055969624,
+ 0.05204316317624377, 0.4215662066908515,
+ -0.7742896036529562, 0.4379916261718371,
+ 0.06203596396290357, -0.10557420870333893,
+ -0.0412892087501817, 0.03268357426711183,
+ 0.01976177894257264, -0.009164231162481846,
+ -0.006764185448053083, 0.0024333732126576722,
+ 0.0016628637020130838, -0.0006381313430451114,
+ -0.00030225958181306315, 0.00014054114970203437,
+ 4.134043227251251e-05, -2.1315026809955787e-05,
+ -3.7346551751414047e-06, 2.0637618513646814e-06,
+ 1.6744288576823017e-07, -9.517657273819165e-08,
+};
+
+static const double coif5_ilp[30] = {
+ -0.00021208083980379827, 0.00035858968789573785,
+ 0.0021782363581090178, -0.004159358781386048,
+ -0.010131117519849788, 0.023408156785839195,
+ 0.02816802897093635, -0.09192001055969624,
+ -0.05204316317624377, 0.4215662066908515,
+ 0.7742896036529562, 0.4379916261718371,
+ -0.06203596396290357, -0.10557420870333893,
+ 0.0412892087501817, 0.03268357426711183,
+ -0.01976177894257264, -0.009164231162481846,
+ 0.006764185448053083, 0.0024333732126576722,
+ -0.0016628637020130838, -0.0006381313430451114,
+ 0.00030225958181306315, 0.00014054114970203437,
+ -4.134043227251251e-05, -2.1315026809955787e-05,
+ 3.7346551751414047e-06, 2.0637618513646814e-06,
+ -1.6744288576823017e-07, -9.517657273819165e-08,
+};
+
+static const double coif5_ihp[30] = {
+ -9.517657273819165e-08, 1.6744288576823017e-07,
+ 2.0637618513646814e-06, -3.7346551751414047e-06,
+ -2.1315026809955787e-05, 4.134043227251251e-05,
+ 0.00014054114970203437, -0.00030225958181306315,
+ -0.0006381313430451114, 0.0016628637020130838,
+ 0.0024333732126576722, -0.006764185448053083,
+ -0.009164231162481846, 0.01976177894257264,
+ 0.03268357426711183, -0.0412892087501817,
+ -0.10557420870333893, 0.06203596396290357,
+ 0.4379916261718371, -0.7742896036529562,
+ 0.4215662066908515, 0.05204316317624377,
+ -0.09192001055969624, -0.02816802897093635,
+ 0.023408156785839195, 0.010131117519849788,
+ -0.004159358781386048, -0.0021782363581090178,
+ 0.00035858968789573785, 0.00021208083980379827,
+};
+
+static const double deb10_lp[20] = {
+ -1.326420300235487e-05, 9.358867000108985e-05,
+ -0.0001164668549943862, -0.0006858566950046825,
+ 0.00199240529499085, 0.0013953517469940798,
+ -0.010733175482979604, 0.0036065535669883944,
+ 0.03321267405893324, -0.02945753682194567,
+ -0.07139414716586077, 0.09305736460380659,
+ 0.12736934033574265, -0.19594627437659665,
+ -0.24984642432648865, 0.2811723436604265,
+ 0.6884590394525921, 0.5272011889309198,
+ 0.18817680007762133, 0.026670057900950818,
+};
+
+static const double deb10_hp[20] = {
+ -0.026670057900950818, 0.18817680007762133,
+ -0.5272011889309198, 0.6884590394525921,
+ -0.2811723436604265, -0.24984642432648865,
+ 0.19594627437659665, 0.12736934033574265,
+ -0.09305736460380659, -0.07139414716586077,
+ 0.02945753682194567, 0.03321267405893324,
+ -0.0036065535669883944, -0.010733175482979604,
+ -0.0013953517469940798, 0.00199240529499085,
+ 0.0006858566950046825, -0.0001164668549943862,
+ -9.358867000108985e-05, -1.326420300235487e-05,
+};
+
+static const double deb10_ilp[20] = {
+ 0.026670057900950818, 0.18817680007762133,
+ 0.5272011889309198, 0.6884590394525921,
+ 0.2811723436604265, -0.24984642432648865,
+ -0.19594627437659665, 0.12736934033574265,
+ 0.09305736460380659, -0.07139414716586077,
+ -0.02945753682194567, 0.03321267405893324,
+ 0.0036065535669883944, -0.010733175482979604,
+ 0.0013953517469940798, 0.00199240529499085,
+ -0.0006858566950046825, -0.0001164668549943862,
+ 9.358867000108985e-05, -1.326420300235487e-05,
+};
+
+static const double deb10_ihp[20] = {
+ -1.326420300235487e-05, -9.358867000108985e-05,
+ -0.0001164668549943862, 0.0006858566950046825,
+ 0.00199240529499085, -0.0013953517469940798,
+ -0.010733175482979604, -0.0036065535669883944,
+ 0.03321267405893324, 0.02945753682194567,
+ -0.07139414716586077, -0.09305736460380659,
+ 0.12736934033574265, 0.19594627437659665,
+ -0.24984642432648865, -0.2811723436604265,
+ 0.6884590394525921, -0.5272011889309198,
+ 0.18817680007762133, -0.026670057900950818,
+};
+
+static const double sym4_lp[8] = {
+ -0.07576571478927333,
+ -0.02963552764599851,
+ 0.49761866763201545,
+ 0.8037387518059161,
+ 0.29785779560527736,
+ -0.09921954357684722,
+ -0.012603967262037833,
+ 0.0322231006040427,
+};
+
+static const double sym4_hp[8] = {
+ -0.0322231006040427,
+ -0.012603967262037833,
+ 0.09921954357684722,
+ 0.29785779560527736,
+ -0.8037387518059161,
+ 0.49761866763201545,
+ 0.02963552764599851,
+ -0.07576571478927333,
+};
+
+static const double sym4_ilp[8] = {
+ 0.0322231006040427,
+ -0.012603967262037833,
+ -0.09921954357684722,
+ 0.29785779560527736,
+ 0.8037387518059161,
+ 0.49761866763201545,
+ -0.02963552764599851,
+ -0.07576571478927333,
+};
+
+static const double sym4_ihp[8] = {
+ -0.07576571478927333,
+ 0.02963552764599851,
+ 0.49761866763201545,
+ -0.8037387518059161,
+ 0.29785779560527736,
+ 0.09921954357684722,
+ -0.012603967262037833,
+ -0.0322231006040427,
+};
+
+static const double sym2_lp[4] = {
+ -0.12940952255092145, 0.22414386804185735,
+ 0.836516303737469, 0.48296291314469025,
+};
+
+static const double sym2_hp[4] = {
+ -0.48296291314469025, 0.836516303737469,
+ -0.22414386804185735, -0.12940952255092145,
+};
+
+static const double sym2_ilp[4] = {
+ 0.48296291314469025, 0.836516303737469,
+ 0.22414386804185735, -0.12940952255092145,
+};
+
+static const double sym2_ihp[4] = {
+ -0.12940952255092145, -0.22414386804185735,
+ 0.836516303737469, -0.48296291314469025,
+};
+
+#define MAX_LEVELS 13
+
+typedef struct ChannelParams {
+ int *output_length;
+ int *filter_length;
+ double **output_coefs;
+ double **subbands_to_free;
+ double **filter_coefs;
+
+ int tempa_length;
+ int tempa_len_max;
+ int temp_in_length;
+ int temp_in_max_length;
+ int buffer_length;
+ int min_left_ext;
+ int max_left_ext;
+
+ double *tempa;
+ double *tempd;
+ double *temp_in;
+ double *buffer;
+ double *buffer2;
+ double *prev;
+ double *overlap;
+} ChannelParams;
+
+typedef struct AudioFWTDNContext {
+ const AVClass *class;
+
+ double sigma;
+ double percent;
+ double softness;
+
+ uint64_t sn;
+ int64_t eof_pts;
+
+ int wavelet_type;
+ int channels;
+ int nb_samples;
+ int levels;
+ int wavelet_length;
+ int need_profile;
+ int got_profile;
+ int adaptive;
+
+ int delay;
+ int drop_samples;
+ int padd_samples;
+ int overlap_length;
+ int prev_length;
+ ChannelParams *cp;
+
+ const double *lp, *hp;
+ const double *ilp, *ihp;
+
+ AVFrame *stddev, *absmean, *filter;
+ AVFrame *new_stddev, *new_absmean;
+
+ int (*filter_channel)(AVFilterContext *ctx, void *arg, int ch, int nb_jobs);
+} AudioFWTDNContext;
+
+#define OFFSET(x) offsetof(AudioFWTDNContext, x)
+#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+#define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
+
+static const AVOption afwtdn_options[] = {
+ { "sigma", "set noise sigma", OFFSET(sigma), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, AFR },
+ { "levels", "set number of wavelet levels", OFFSET(levels), AV_OPT_TYPE_INT, {.i64=10}, 1, MAX_LEVELS-1, AF },
+ { "wavet", "set wavelet type", OFFSET(wavelet_type), AV_OPT_TYPE_INT, {.i64=SYM10}, 0, NB_WAVELET_TYPES - 1, AF, "wavet" },
+ { "sym2", "sym2", 0, AV_OPT_TYPE_CONST, {.i64=SYM2}, 0, 0, AF, "wavet" },
+ { "sym4", "sym4", 0, AV_OPT_TYPE_CONST, {.i64=SYM4}, 0, 0, AF, "wavet" },
+ { "rbior68", "rbior68", 0, AV_OPT_TYPE_CONST, {.i64=RBIOR68}, 0, 0, AF, "wavet" },
+ { "deb10", "deb10", 0, AV_OPT_TYPE_CONST, {.i64=DEB10}, 0, 0, AF, "wavet" },
+ { "sym10", "sym10", 0, AV_OPT_TYPE_CONST, {.i64=SYM10}, 0, 0, AF, "wavet" },
+ { "coif5", "coif5", 0, AV_OPT_TYPE_CONST, {.i64=COIF5}, 0, 0, AF, "wavet" },
+ { "bl3", "bl3", 0, AV_OPT_TYPE_CONST, {.i64=BL3}, 0, 0, AF, "wavet" },
+ { "percent", "set percent of full denoising", OFFSET(percent),AV_OPT_TYPE_DOUBLE, {.dbl=85}, 0, 100, AFR },
+ { "profile", "profile noise", OFFSET(need_profile), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AFR },
+ { "adaptive", "adaptive profiling of noise", OFFSET(adaptive), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AFR },
+ { "samples", "set frame size in number of samples", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64=8192}, 512, 65536, AF },
+ { "softness", "set thresholding softness", OFFSET(softness), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 10, AFR },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(afwtdn);
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats = NULL;
+ AVFilterChannelLayouts *layouts = NULL;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_DBLP,
+ AV_SAMPLE_FMT_NONE
+ };
+ int ret;
+
+ formats = ff_make_format_list(sample_fmts);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_formats(ctx, formats);
+ if (ret < 0)
+ return ret;
+
+ layouts = ff_all_channel_counts();
+ if (!layouts)
+ return AVERROR(ENOMEM);
+
+ ret = ff_set_common_channel_layouts(ctx, layouts);
+ if (ret < 0)
+ return ret;
+
+ formats = ff_all_samplerates();
+ return ff_set_common_samplerates(ctx, formats);
+}
+
+#define pow2(x) (1U << (x))
+#define mod_pow2(x, power_of_two) ((x) & ((power_of_two) - 1))
+
+static void conv_down(double *in, int in_length, double *low, double *high,
+ int out_length, const double *lp, const double *hp,
+ int wavelet_length, int skip,
+ double *buffer, int buffer_length)
+{
+ double thigh = 0.0, tlow = 0.0;
+ int buff_idx = 1 + skip;
+
+ memcpy(buffer, in, buff_idx * sizeof(*buffer));
+ memset(buffer + buff_idx, 0, (buffer_length - buff_idx) * sizeof(*buffer));
+
+ for (int i = 0; i < out_length - 1; i++) {
+ double thigh = 0.0, tlow = 0.0;
+
+ for (int j = 0; j < wavelet_length; j++) {
+ const int idx = mod_pow2(-j + buff_idx - 1, buffer_length);
+ const double btemp = buffer[idx];
+
+ thigh += btemp * hp[j];
+ tlow += btemp * lp[j];
+ }
+
+ high[i] = thigh;
+ low[i] = tlow;
+ buffer[buff_idx++] = in[2 * i + 1 + skip];
+ buffer[buff_idx++] = in[2 * i + 2 + skip];
+ buff_idx = mod_pow2(buff_idx, buffer_length);
+ }
+
+ for (int i = 0; i < wavelet_length; i++) {
+ const int idx = mod_pow2(-i + buff_idx - 1, buffer_length);
+ const double btemp = buffer[idx];
+
+ thigh += btemp * hp[i];
+ tlow += btemp * lp[i];
+ }
+
+ high[out_length - 1] = thigh;
+ low[out_length - 1] = tlow;
+}
+
+static int left_ext(int wavelet_length, int levels, uint64_t sn)
+{
+ if (!sn)
+ return 0;
+ return (pow2(levels) - 1) * (wavelet_length - 2) + mod_pow2(sn, pow2(levels));
+}
+
+static int nb_coefs(int length, int level, uint64_t sn)
+{
+ const int pow2_level = pow2(level);
+
+ return (sn + length) / pow2_level - sn / pow2_level;
+}
+
+static int reallocate_inputs(double **out, int *out_length,
+ int in_length, int levels, int ch, uint64_t sn)
+{
+ const int temp_length = nb_coefs(in_length, levels, sn);
+
+ for (int level = 0; level < levels; level++) {
+ const int temp_length = nb_coefs(in_length, level + 1, sn);
+
+ if (temp_length > out_length[level]) {
+ av_freep(&out[level]);
+ out_length[level] = 0;
+
+ out[level] = av_calloc(temp_length + 1, sizeof(**out));
+ if (!out[level])
+ return AVERROR(ENOMEM);
+ out_length[level] = temp_length + 1;
+ }
+
+ memset(out[level] + temp_length, 0,
+ (out_length[level] - temp_length) * sizeof(**out));
+ out_length[level] = temp_length;
+ }
+
+ if (temp_length > out_length[levels]) {
+ av_freep(&out[levels]);
+ out_length[levels] = 0;
+
+ out[levels] = av_calloc(temp_length + 1, sizeof(**out));
+ if (!out[levels])
+ return AVERROR(ENOMEM);
+ out_length[levels] = temp_length + 1;
+ }
+
+ memset(out[levels] + temp_length, 0,
+ (out_length[levels] - temp_length) * sizeof(**out));
+ out_length[levels] = temp_length;
+
+ return 0;
+}
+
+static int max_left_zeros_inverse(int levels, int level, int wavelet_length)
+{
+ return (pow2(levels - level) - 1) * (wavelet_length - 1);
+}
+
+static int reallocate_outputs(AudioFWTDNContext *s,
+ double **out, int *out_length,
+ int in_length, int levels, int ch, uint64_t sn)
+{
+ ChannelParams *cp = &s->cp[ch];
+ int temp_length = 0;
+ int add = 0;
+
+ for (int level = 0; level < levels; level++) {
+ temp_length = nb_coefs(in_length, level + 1, sn);
+ if (temp_length > out_length[level]) {
+ av_freep(&cp->subbands_to_free[level]);
+ out_length[level] = 0;
+
+ add = max_left_zeros_inverse(levels, level + 1, s->wavelet_length);
+ cp->subbands_to_free[level] = av_calloc(add + temp_length + 1, sizeof(**out));
+ if (!cp->subbands_to_free[level])
+ return AVERROR(ENOMEM);
+ out_length[level] = add + temp_length + 1;
+ out[level] = cp->subbands_to_free[level] + add;
+ }
+
+ memset(out[level] + temp_length, 0,
+ FFMAX(out_length[level] - temp_length - add, 0) * sizeof(**out));
+ out_length[level] = temp_length;
+ }
+
+ temp_length = nb_coefs(in_length, levels, sn);
+ if (temp_length > out_length[levels]) {
+ av_freep(&cp->subbands_to_free[levels]);
+ out_length[levels] = 0;
+
+ cp->subbands_to_free[levels] = av_calloc(temp_length + 1, sizeof(**out));
+ if (!cp->subbands_to_free[levels])
+ return AVERROR(ENOMEM);
+ out_length[levels] = temp_length + 1;
+ out[levels] = cp->subbands_to_free[levels];
+ }
+
+ memset(out[levels] + temp_length, 0,
+ (out_length[levels] - temp_length) * sizeof(**out));
+ out_length[levels] = temp_length;
+
+ return 0;
+}
+
+static int discard_left_ext(int wavelet_length, int levels, int level, uint64_t sn)
+{
+ if (levels == level || sn == 0)
+ return 0;
+ return (pow2(levels - level) - 1) * (wavelet_length - 2) + mod_pow2(sn, pow2(levels)) / pow2(level);
+}
+
+static int forward(AudioFWTDNContext *s,
+ const double *in, int in_length,
+ double **out, int *out_length, int ch, uint64_t sn)
+{
+ ChannelParams *cp = &s->cp[ch];
+ int levels = s->levels;
+ int skip = sn ? s->wavelet_length - 1 : 1;
+ int leftext, ret;
+
+ ret = reallocate_inputs(out, out_length, in_length, levels, ch, sn);
+ if (ret < 0)
+ return ret;
+ ret = reallocate_outputs(s, cp->filter_coefs, cp->filter_length,
+ in_length, levels, ch, sn);
+ if (ret < 0)
+ return ret;
+
+ leftext = left_ext(s->wavelet_length, levels, sn);
+
+ if (cp->temp_in_max_length < in_length + cp->max_left_ext + skip) {
+ av_freep(&cp->temp_in);
+ cp->temp_in_max_length = in_length + cp->max_left_ext + skip;
+ cp->temp_in = av_calloc(cp->temp_in_max_length, sizeof(*cp->temp_in));
+ if (!cp->temp_in) {
+ cp->temp_in_max_length = 0;
+ return AVERROR(ENOMEM);
+ }
+ }
+
+ memset(cp->temp_in, 0, cp->temp_in_max_length * sizeof(*cp->temp_in));
+ cp->temp_in_length = in_length + leftext;
+
+ if (leftext)
+ memcpy(cp->temp_in, cp->prev + s->prev_length - leftext, leftext * sizeof(*cp->temp_in));
+ memcpy(cp->temp_in + leftext, in, in_length * sizeof(*in));
+
+ if (levels == 1) {
+ conv_down(cp->temp_in, cp->temp_in_length, out[1], out[0], out_length[1],
+ s->lp, s->hp, s->wavelet_length, skip,
+ cp->buffer, cp->buffer_length);
+ } else {
+ int discard = discard_left_ext(s->wavelet_length, levels, 1, sn);
+ int tempa_length_prev;
+
+ if (cp->tempa_len_max < (in_length + cp->max_left_ext + s->wavelet_length - 1) / 2) {
+ av_freep(&cp->tempa);
+ av_freep(&cp->tempd);
+ cp->tempa_len_max = (in_length + cp->max_left_ext + s->wavelet_length - 1) / 2;
+ cp->tempa = av_calloc(cp->tempa_len_max, sizeof(*cp->tempa));
+ cp->tempd = av_calloc(cp->tempa_len_max, sizeof(*cp->tempd));
+ if (!cp->tempa || !cp->tempd) {
+ cp->tempa_len_max = 0;
+ return AVERROR(ENOMEM);
+ }
+ }
+
+ memset(cp->tempa, 0, cp->tempa_len_max * sizeof(*cp->tempa));
+ memset(cp->tempd, 0, cp->tempa_len_max * sizeof(*cp->tempd));
+
+ cp->tempa_length = out_length[0] + discard;
+ conv_down(cp->temp_in, cp->temp_in_length,
+ cp->tempa, cp->tempd, cp->tempa_length,
+ s->lp, s->hp, s->wavelet_length, skip,
+ cp->buffer, cp->buffer_length);
+ memcpy(out[0], cp->tempd + discard, out_length[0] * sizeof(**out));
+ tempa_length_prev = cp->tempa_length;
+
+ for (int level = 1; level < levels - 1; level++) {
+ if (out_length[level] == 0)
+ return 0;
+ discard = discard_left_ext(s->wavelet_length, levels, level + 1, sn);
+ cp->tempa_length = out_length[level] + discard;
+ conv_down(cp->tempa, tempa_length_prev,
+ cp->tempa, cp->tempd, cp->tempa_length,
+ s->lp, s->hp, s->wavelet_length, skip,
+ cp->buffer, cp->buffer_length);
+ memcpy(out[level], cp->tempd + discard, out_length[level] * sizeof(**out));
+ tempa_length_prev = cp->tempa_length;
+ }
+
+ if (out_length[levels] == 0)
+ return 0;
+ conv_down(cp->tempa, cp->tempa_length, out[levels], out[levels - 1], out_length[levels],
+ s->lp, s->hp, s->wavelet_length, skip,
+ cp->buffer, cp->buffer_length);
+ }
+
+ if (s->prev_length < in_length) {
+ memcpy(cp->prev, in + in_length - cp->max_left_ext, cp->max_left_ext * sizeof(*cp->prev));
+ } else {
+ memmove(cp->prev, cp->prev + in_length, (s->prev_length - in_length) * sizeof(*cp->prev));
+ memcpy(cp->prev + s->prev_length - in_length, in, in_length * sizeof(*cp->prev));
+ }
+
+ return 0;
+}
+
+static void conv_up(double *low, double *high, int in_length, double *out, int out_length,
+ const double *lp, const double *hp, int filter_length,
+ double *buffer, double *buffer2, int buffer_length)
+{
+ int shift = 0, buff_idx = 0, in_idx = 0;
+
+ memset(buffer, 0, buffer_length * sizeof(*buffer));
+ memset(buffer2, 0, buffer_length * sizeof(*buffer2));
+
+ for (int i = 0; i < out_length; i++) {
+ double sum = 0.0;
+
+ if ((i & 1) == 0) {
+ if (in_idx < in_length) {
+ buffer[buff_idx] = low[in_idx];
+ buffer2[buff_idx] = high[in_idx++];
+ } else {
+ buffer[buff_idx] = 0;
+ buffer2[buff_idx] = 0;
+ }
+ buff_idx++;
+ if (buff_idx >= buffer_length)
+ buff_idx = 0;
+ shift = 0;
+ }
+
+ for (int j = 0; j < (filter_length - shift + 1) / 2; j++) {
+ const int idx = mod_pow2(-j + buff_idx - 1, buffer_length);
+
+ sum += buffer[idx] * lp[j * 2 + shift] + buffer2[idx] * hp[j * 2 + shift];
+ }
+ out[i] = sum;
+ shift = 1;
+ }
+}
+
+static int append_left_ext(int wavelet_length, int levels, int level, uint64_t sn)
+{
+ if (levels == level)
+ return 0;
+
+ return (pow2(levels - level) - 1) * (wavelet_length - 2) +
+ mod_pow2(sn, pow2(levels)) / pow2(level);
+}
+
+static int inverse(AudioFWTDNContext *s,
+ double **in, int *in_length,
+ double *out, int out_length, int ch, uint64_t sn)
+{
+ ChannelParams *cp = &s->cp[ch];
+ const int levels = s->levels;
+ int leftext = left_ext(s->wavelet_length, levels, sn);
+ int temp_skip = 0;
+
+ if (sn == 0)
+ temp_skip = cp->min_left_ext;
+
+ memset(out, 0, out_length * sizeof(*out));
+
+ if (cp->temp_in_max_length < out_length + cp->max_left_ext + s->wavelet_length - 1) {
+ av_freep(&cp->temp_in);
+ cp->temp_in_max_length = out_length + cp->max_left_ext + s->wavelet_length - 1;
+ cp->temp_in = av_calloc(cp->temp_in_max_length, sizeof(*cp->temp_in));
+ if (!cp->temp_in) {
+ cp->temp_in_max_length = 0;
+ return AVERROR(ENOMEM);
+ }
+ }
+
+ memset(cp->temp_in, 0, cp->temp_in_max_length * sizeof(*cp->temp_in));
+ cp->temp_in_length = out_length + cp->max_left_ext;
+
+ if (levels == 1) {
+ conv_up(in[1], in[0], in_length[1], cp->temp_in, cp->temp_in_length,
+ s->ilp, s->ihp, s->wavelet_length,
+ cp->buffer, cp->buffer2, cp->buffer_length);
+ memcpy(out + cp->max_left_ext - leftext, cp->temp_in + temp_skip,
+ FFMAX(0, out_length - (cp->max_left_ext - leftext)) * sizeof(*out));
+ } else {
+ double *hp1, *hp2;
+ int add, add2;
+
+ if (cp->tempa_len_max < (out_length + cp->max_left_ext + s->wavelet_length - 1) / 2) {
+ av_freep(&cp->tempa);
+ cp->tempa_len_max = (out_length + cp->max_left_ext + s->wavelet_length - 1) / 2;
+ cp->tempa = av_calloc(cp->tempa_len_max, sizeof(*cp->tempa));
+ if (!cp->tempa) {
+ cp->tempa_len_max = 0;
+ return AVERROR(ENOMEM);
+ }
+ }
+
+ memset(cp->tempa, 0, cp->tempa_len_max * sizeof(*cp->tempa));
+
+ hp1 = levels & 1 ? cp->temp_in : cp->tempa;
+ hp2 = levels & 1 ? cp->tempa : cp->temp_in;
+
+ add = append_left_ext(s->wavelet_length, levels, levels - 1, sn);
+ conv_up(in[levels], in[levels - 1], in_length[levels], hp1, in_length[levels - 2] + add,
+ s->ilp, s->ihp, s->wavelet_length, cp->buffer, cp->buffer2, cp->buffer_length);
+
+ for (int level = levels - 1; level > 1; level--) {
+ add2 = append_left_ext(s->wavelet_length, levels, level - 1, sn);
+ add = append_left_ext(s->wavelet_length, levels, level, sn);
+ conv_up(hp1, in[level - 1] - add, in_length[level - 1] + add,
+ hp2, in_length[level - 2] + add2,
+ s->ilp, s->ihp, s->wavelet_length,
+ cp->buffer, cp->buffer2, cp->buffer_length);
+ FFSWAP(double *, hp1, hp2);
+ }
+
+ add = append_left_ext(s->wavelet_length, levels, 1, sn);
+ conv_up(hp1, in[0] - add, in_length[0] + add, cp->temp_in, cp->temp_in_length,
+ s->ilp, s->ihp, s->wavelet_length,
+ cp->buffer, cp->buffer2, cp->buffer_length);
+ }
+
+ memset(cp->temp_in, 0, temp_skip * sizeof(*cp->temp_in));
+ if (s->overlap_length <= out_length) {
+ memcpy(out + cp->max_left_ext - leftext, cp->temp_in + temp_skip,
+ FFMAX(0, out_length - (cp->max_left_ext - leftext)) * sizeof(*out));
+ for (int i = 0;i < FFMIN(s->overlap_length, out_length); i++)
+ out[i] += cp->overlap[i];
+
+ memcpy(cp->overlap, cp->temp_in + out_length - (cp->max_left_ext - leftext),
+ s->overlap_length * sizeof(*cp->overlap));
+ } else {
+ for (int i = 0;i < s->overlap_length - (cp->max_left_ext - leftext); i++)
+ cp->overlap[i + cp->max_left_ext - leftext] += cp->temp_in[i];
+ memcpy(out, cp->overlap, out_length * sizeof(*out));
+ memmove(cp->overlap, cp->overlap + out_length,
+ (s->overlap_length - out_length) * sizeof(*cp->overlap));
+ memcpy(cp->overlap + s->overlap_length - out_length, cp->temp_in + leftext,
+ out_length * sizeof(*cp->overlap));
+ }
+
+ return 0;
+}
+
+static int next_pow2(int in)
+{
+ return 1 << (av_log2(in) + 1);
+}
+
+static void denoise_level(double *out, const double *in,
+ const double *filter,
+ double percent, int length)
+{
+ const double x = percent * 0.01;
+ const double y = 1.0 - x;
+
+ for (int i = 0; i < length; i++)
+ out[i] = x * filter[i] + in[i] * y;
+}
+
+static double sqr(double in)
+{
+ return in * in;
+}
+
+static double measure_mean(const double *in, int length)
+{
+ double sum = 0.0;
+
+ for (int i = 0; i < length; i++)
+ sum += in[i];
+
+ return sum / length;
+}
+
+static double measure_absmean(const double *in, int length)
+{
+ double sum = 0.0;
+
+ for (int i = 0; i < length; i++)
+ sum += fabs(in[i]);
+
+ return sum / length;
+}
+
+static double measure_stddev(const double *in, int length, double mean)
+{
+ double sum = 0.;
+
+ for (int i = 0; i < length; i++) {
+ sum += sqr(in[i] - mean);
+ }
+
+ return sqrt(sum / length);
+}
+
+static void noise_filter(const double stddev, const double *in,
+ double *out, double absmean, double softness,
+ double new_stddev, int length)
+{
+ for (int i = 0; i < length; i++) {
+ if (new_stddev <= stddev)
+ out[i] = 0.0;
+ else if (fabs(in[i]) <= absmean)
+ out[i] = 0.0;
+ else
+ out[i] = in[i] - FFSIGN(in[i]) * absmean / exp(3.0 * softness * (fabs(in[i]) - absmean) / absmean);
+ }
+}
+
+typedef struct ThreadData {
+ AVFrame *in, *out;
+} ThreadData;
+
+static int filter_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
+{
+ AudioFWTDNContext *s = ctx->priv;
+ ThreadData *td = arg;
+ AVFrame *in = td->in;
+ AVFrame *out = td->out;
+ ChannelParams *cp = &s->cp[ch];
+ const double *src = (const double *)(in->extended_data[ch]);
+ double *dst = (double *)out->extended_data[ch];
+ double *absmean = (double *)s->absmean->extended_data[ch];
+ double *new_absmean = (double *)s->new_absmean->extended_data[ch];
+ double *stddev = (double *)s->stddev->extended_data[ch];
+ double *new_stddev = (double *)s->new_stddev->extended_data[ch];
+ double *filter = (double *)s->filter->extended_data[ch];
+ double is_noise = 0.0;
+ int ret;
+
+ ret = forward(s, src, in->nb_samples, cp->output_coefs, cp->output_length, ch, s->sn);
+ if (ret < 0)
+ return ret;
+
+ if (!s->got_profile && s->need_profile) {
+ for (int level = 0; level <= s->levels; level++) {
+ const int length = cp->output_length[level];
+ const double scale = sqrt(2.0 * log(length));
+
+ stddev[level] = measure_stddev(cp->output_coefs[level], length,
+ measure_mean(cp->output_coefs[level], length)) * scale;
+ absmean[level] = measure_absmean(cp->output_coefs[level], length) * scale;
+ }
+ } else if (!s->got_profile && !s->need_profile && !s->adaptive) {
+ for (int level = 0; level <= s->levels; level++) {
+ const int length = cp->output_length[level];
+ const double scale = sqrt(2.0 * log(length));
+
+ stddev[level] = 0.5 * s->sigma * scale;
+ absmean[level] = 0.5 * s->sigma * scale;
+ }
+ }
+
+ for (int level = 0; level <= s->levels; level++) {
+ const int length = cp->output_length[level];
+ double vad;
+
+ new_stddev[level] = measure_stddev(cp->output_coefs[level], length,
+ measure_mean(cp->output_coefs[level], length));
+ new_absmean[level] = measure_absmean(cp->output_coefs[level], length);
+ if (new_absmean[level] <= FLT_EPSILON)
+ vad = 1.0;
+ else
+ vad = new_stddev[level] / new_absmean[level];
+ if (level < s->levels)
+ is_noise += sqr(vad - 1.232);
+ }
+
+ is_noise *= in->sample_rate;
+ is_noise /= s->nb_samples;
+ for (int level = 0; level <= s->levels; level++) {
+ const int length = cp->output_length[level];
+ const double scale = sqrt(2.0 * log(length));
+
+ if (is_noise < 0.05 && s->adaptive) {
+ stddev[level] = new_stddev[level] * scale;
+ absmean[level] = new_absmean[level] * scale;
+ }
+
+ noise_filter(stddev[level], cp->output_coefs[level], filter, absmean[level],
+ s->softness, new_stddev[level], length);
+ denoise_level(cp->filter_coefs[level], cp->output_coefs[level], filter, s->percent, length);
+ }
+
+ ret = inverse(s, cp->filter_coefs, cp->filter_length, dst, out->nb_samples, ch, s->sn);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AudioFWTDNContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+ ThreadData td;
+ AVFrame *out;
+ int eof = in == NULL;
+
+ out = ff_get_audio_buffer(outlink, s->nb_samples);
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+ if (in) {
+ av_frame_copy_props(out, in);
+ s->eof_pts = in->pts + in->nb_samples;
+ }
+ if (eof)
+ out->pts = s->eof_pts - s->padd_samples;
+
+ if (!in || in->nb_samples < s->nb_samples) {
+ AVFrame *new_in = ff_get_audio_buffer(outlink, s->nb_samples);
+
+ if (!new_in) {
+ av_frame_free(&in);
+ av_frame_free(&out);
+ return AVERROR(ENOMEM);
+ }
+ if (in)
+ av_frame_copy_props(new_in, in);
+
+ s->padd_samples -= s->nb_samples - (in ? in->nb_samples: 0);
+ if (in)
+ av_samples_copy(new_in->extended_data, in->extended_data, 0, 0,
+ in->nb_samples, in->channels, in->format);
+ av_frame_free(&in);
+ in = new_in;
+ }
+
+ td.in = in;
+ td.out = out;
+ ctx->internal->execute(ctx, s->filter_channel, &td, NULL, inlink->channels);
+ if (s->need_profile)
+ s->got_profile = 1;
+
+ s->sn += s->nb_samples;
+
+ if (s->drop_samples >= in->nb_samples) {
+ s->drop_samples -= in->nb_samples;
+ s->delay += in->nb_samples;
+ av_frame_free(&in);
+ av_frame_free(&out);
+ FF_FILTER_FORWARD_STATUS(inlink, outlink);
+ FF_FILTER_FORWARD_WANTED(outlink, inlink);
+ return 0;
+ } else if (s->drop_samples > 0) {
+ for (int ch = 0; ch < out->channels; ch++) {
+ memmove(out->extended_data[ch],
+ out->extended_data[ch] + s->drop_samples * sizeof(double),
+ (in->nb_samples - s->drop_samples) * sizeof(double));
+ }
+
+ out->nb_samples = in->nb_samples - s->drop_samples;
+ out->pts = in->pts - av_rescale_q(s->delay, (AVRational){1, outlink->sample_rate}, outlink->time_base);
+ s->delay += s->drop_samples;
+ s->drop_samples = 0;
+ } else {
+ if (s->padd_samples < 0 && eof) {
+ out->nb_samples += s->padd_samples;
+ s->padd_samples = 0;
+ }
+ if (!eof)
+ out->pts = in->pts - av_rescale_q(s->delay, (AVRational){1, outlink->sample_rate}, outlink->time_base);
+ }
+
+ av_frame_free(&in);
+ return ff_filter_frame(outlink, out);
+}
+
+static int max_left_ext(int wavelet_length, int levels)
+{
+ return (pow2(levels) - 1) * (wavelet_length - 1);
+}
+
+static int min_left_ext(int wavelet_length, int levels)
+{
+ return (pow2(levels) - 1) * (wavelet_length - 2);
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AudioFWTDNContext *s = ctx->priv;
+
+ switch (s->wavelet_type) {
+ case SYM2:
+ s->wavelet_length = 4;
+ s->lp = sym2_lp;
+ s->hp = sym2_hp;
+ s->ilp = sym2_ilp;
+ s->ihp = sym2_ihp;
+ break;
+ case SYM4:
+ s->wavelet_length = 8;
+ s->lp = sym4_lp;
+ s->hp = sym4_hp;
+ s->ilp = sym4_ilp;
+ s->ihp = sym4_ihp;
+ break;
+ case RBIOR68:
+ s->wavelet_length = 18;
+ s->lp = rbior68_lp;
+ s->hp = rbior68_hp;
+ s->ilp = rbior68_ilp;
+ s->ihp = rbior68_ihp;
+ break;
+ case DEB10:
+ s->wavelet_length = 20;
+ s->lp = deb10_lp;
+ s->hp = deb10_hp;
+ s->ilp = deb10_ilp;
+ s->ihp = deb10_ihp;
+ case SYM10:
+ s->wavelet_length = 20;
+ s->lp = sym10_lp;
+ s->hp = sym10_hp;
+ s->ilp = sym10_ilp;
+ s->ihp = sym10_ihp;
+ break;
+ case COIF5:
+ s->wavelet_length = 30;
+ s->lp = coif5_lp;
+ s->hp = coif5_hp;
+ s->ilp = coif5_ilp;
+ s->ihp = coif5_ihp;
+ break;
+ case BL3:
+ s->wavelet_length = 42;
+ s->lp = bl3_lp;
+ s->hp = bl3_hp;
+ s->ilp = bl3_ilp;
+ s->ihp = bl3_ihp;
+ break;
+ default:
+ av_assert0(0);
+ }
+
+ s->levels = FFMIN(s->levels, lrint(log(s->nb_samples / (s->wavelet_length - 1.0)) / M_LN2));
+ av_log(ctx, AV_LOG_VERBOSE, "levels: %d\n", s->levels);
+ s->filter_channel = filter_channel;
+
+ s->stddev = ff_get_audio_buffer(outlink, MAX_LEVELS);
+ s->new_stddev = ff_get_audio_buffer(outlink, MAX_LEVELS);
+ s->filter = ff_get_audio_buffer(outlink, s->nb_samples);
+ s->absmean = ff_get_audio_buffer(outlink, MAX_LEVELS);
+ s->new_absmean = ff_get_audio_buffer(outlink, MAX_LEVELS);
+ if (!s->stddev || !s->absmean || !s->filter ||
+ !s->new_stddev || !s->new_absmean)
+ return AVERROR(ENOMEM);
+
+ s->channels = outlink->channels;
+ s->overlap_length = max_left_ext(s->wavelet_length, s->levels);
+ s->prev_length = s->overlap_length;
+ s->drop_samples = s->overlap_length;
+ s->padd_samples = s->overlap_length;
+ s->sn = 1;
+
+ s->cp = av_calloc(s->channels, sizeof(*s->cp));
+ if (!s->cp)
+ return AVERROR(ENOMEM);
+
+ for (int ch = 0; ch < s->channels; ch++) {
+ ChannelParams *cp = &s->cp[ch];
+
+ cp->output_coefs = av_calloc(s->levels + 1, sizeof(*cp->output_coefs));
+ cp->filter_coefs = av_calloc(s->levels + 1, sizeof(*cp->filter_coefs));
+ cp->output_length = av_calloc(s->levels + 1, sizeof(*cp->output_length));
+ cp->filter_length = av_calloc(s->levels + 1, sizeof(*cp->filter_length));
+ cp->buffer_length = next_pow2(s->wavelet_length);
+ cp->buffer = av_calloc(cp->buffer_length, sizeof(*cp->buffer));
+ cp->buffer2 = av_calloc(cp->buffer_length, sizeof(*cp->buffer2));
+ cp->subbands_to_free = av_calloc(s->levels + 1, sizeof(*cp->subbands_to_free));
+ cp->prev = av_calloc(s->prev_length, sizeof(*cp->prev));
+ cp->overlap = av_calloc(s->overlap_length, sizeof(*cp->overlap));
+ cp->max_left_ext = max_left_ext(s->wavelet_length, s->levels);
+ cp->min_left_ext = min_left_ext(s->wavelet_length, s->levels);
+ if (!cp->output_coefs || !cp->filter_coefs || !cp->output_length ||
+ !cp->filter_length || !cp->subbands_to_free || !cp->prev || !cp->overlap ||
+ !cp->buffer || !cp->buffer2)
+ return AVERROR(ENOMEM);
+ }
+
+ return 0;
+}
+
+static int activate(AVFilterContext *ctx)
+{
+ AVFilterLink *inlink = ctx->inputs[0];
+ AVFilterLink *outlink = ctx->outputs[0];
+ AudioFWTDNContext *s = ctx->priv;
+ AVFrame *in = NULL;
+ int ret, status;
+ int64_t pts;
+
+ FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
+
+ ret = ff_inlink_consume_samples(inlink, s->nb_samples, s->nb_samples, &in);
+ if (ret < 0)
+ return ret;
+ if (ret > 0)
+ return filter_frame(inlink, in);
+
+ if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
+ if (status == AVERROR_EOF) {
+ while (s->padd_samples != 0) {
+ ret = filter_frame(inlink, NULL);
+ if (ret < 0)
+ return ret;
+ }
+ ff_outlink_set_status(outlink, status, pts);
+ return ret;
+ }
+ }
+ FF_FILTER_FORWARD_WANTED(outlink, inlink);
+
+ return FFERROR_NOT_READY;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ AudioFWTDNContext *s = ctx->priv;
+
+ av_frame_free(&s->filter);
+ av_frame_free(&s->new_stddev);
+ av_frame_free(&s->stddev);
+ av_frame_free(&s->new_absmean);
+ av_frame_free(&s->absmean);
+
+ for (int ch = 0; s->cp && ch < s->channels; ch++) {
+ ChannelParams *cp = &s->cp[ch];
+
+ av_freep(&cp->tempa);
+ av_freep(&cp->tempd);
+ av_freep(&cp->temp_in);
+ av_freep(&cp->buffer);
+ av_freep(&cp->buffer2);
+ av_freep(&cp->prev);
+ av_freep(&cp->overlap);
+
+ av_freep(&cp->output_length);
+ av_freep(&cp->filter_length);
+
+ if (cp->output_coefs) {
+ for (int level = 0; level <= s->levels; level++)
+ av_freep(&cp->output_coefs[level]);
+ }
+
+ if (cp->subbands_to_free) {
+ for (int level = 0; level <= s->levels; level++)
+ av_freep(&cp->subbands_to_free[level]);
+ }
+
+ av_freep(&cp->subbands_to_free);
+ av_freep(&cp->output_coefs);
+ av_freep(&cp->filter_coefs);
+ }
+
+ av_freep(&s->cp);
+}
+
+static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
+ char *res, int res_len, int flags)
+{
+ AudioFWTDNContext *s = ctx->priv;
+ int ret;
+
+ ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
+ if (ret < 0)
+ return ret;
+
+ if (!strcmp(cmd, "profile") && s->need_profile)
+ s->got_profile = 0;
+
+ return 0;
+}
+
+static const AVFilterPad inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+static const AVFilterPad outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output,
+ },
+ { NULL }
+};
+
+const AVFilter ff_af_afwtdn = {
+ .name = "afwtdn",
+ .description = NULL_IF_CONFIG_SMALL("Denoise audio stream using Wavelets."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(AudioFWTDNContext),
+ .priv_class = &afwtdn_class,
+ .activate = activate,
+ .uninit = uninit,
+ .inputs = inputs,
+ .outputs = outputs,
+ .process_command = process_command,
+ .flags = AVFILTER_FLAG_SLICE_THREADS,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index c6afef835f..ae74f9c891 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -46,6 +46,7 @@ extern const AVFilter ff_af_afftfilt;
extern const AVFilter ff_af_afir;
extern const AVFilter ff_af_aformat;
extern const AVFilter ff_af_afreqshift;
+extern const AVFilter ff_af_afwtdn;
extern const AVFilter ff_af_agate;
extern const AVFilter ff_af_aiir;
extern const AVFilter ff_af_aintegral;
diff --git a/libavfilter/version.h b/libavfilter/version.h
index fbb81ef31c..75cd10dccd 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 8
-#define LIBAVFILTER_VERSION_MINOR 0
+#define LIBAVFILTER_VERSION_MINOR 1
#define LIBAVFILTER_VERSION_MICRO 103
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