[FFmpeg-cvslog] avfilter: add audio frequency and phase shift filters
Paul B Mahol
git at videolan.org
Tue Oct 20 21:43:41 EEST 2020
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Sat Oct 17 20:34:08 2020 +0200| [637c154a5048f60d0b59d35941d4d528edf56370] | committer: Paul B Mahol
avfilter: add audio frequency and phase shift filters
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=637c154a5048f60d0b59d35941d4d528edf56370
---
Changelog | 1 +
doc/filters.texi | 30 ++++
libavfilter/Makefile | 2 +
libavfilter/af_afreqshift.c | 379 ++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 2 +
libavfilter/version.h | 2 +-
6 files changed, 415 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index 996ed2ca51..21a4be731b 100644
--- a/Changelog
+++ b/Changelog
@@ -36,6 +36,7 @@ version <next>:
- AVS3 video decoder via libuavs3d
- Cintel RAW decoder
- VDPAU accelerated VP9 10/12bit decoding
+- afreqshift and aphaseshift filters
version 4.3:
diff --git a/doc/filters.texi b/doc/filters.texi
index 809761a43e..f664f0824d 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1314,6 +1314,21 @@ Force the output to either unsigned 8-bit or signed 16-bit stereo
aformat=sample_fmts=u8|s16:channel_layouts=stereo
@end example
+ at section afreqshift
+Apply frequency shift to input audio samples.
+
+The filter accepts the following options:
+
+ at table @option
+ at item shift
+Specify frequency shift. Allowed range is -INT_MAX to INT_MAX.
+Default value is 0.0.
+ at end table
+
+ at subsection Commands
+
+This filter supports the above option as @ref{commands}.
+
@section agate
A gate is mainly used to reduce lower parts of a signal. This kind of signal
@@ -2064,6 +2079,21 @@ It accepts the following values:
@end table
@end table
+ at section aphaseshift
+Apply phase shift to input audio samples.
+
+The filter accepts the following options:
+
+ at table @option
+ at item shift
+Specify phase shift. Allowed range is from -1.0 to 1.0.
+Default value is 0.0.
+ at end table
+
+ at subsection Commands
+
+This filter supports the above option as @ref{commands}.
+
@section apulsator
Audio pulsator is something between an autopanner and a tremolo.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index e6d3c283da..efdea39ccc 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -50,6 +50,7 @@ OBJS-$(CONFIG_AFFTDN_FILTER) += af_afftdn.o
OBJS-$(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_AGATE_FILTER) += af_agate.o
OBJS-$(CONFIG_AIIR_FILTER) += af_aiir.o
OBJS-$(CONFIG_AINTEGRAL_FILTER) += af_aderivative.o
@@ -68,6 +69,7 @@ OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
OBJS-$(CONFIG_APAD_FILTER) += af_apad.o
OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o
OBJS-$(CONFIG_APHASER_FILTER) += af_aphaser.o generate_wave_table.o
+OBJS-$(CONFIG_APHASESHIFT_FILTER) += af_afreqshift.o
OBJS-$(CONFIG_APULSATOR_FILTER) += af_apulsator.o
OBJS-$(CONFIG_AREALTIME_FILTER) += f_realtime.o
OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
diff --git a/libavfilter/af_afreqshift.c b/libavfilter/af_afreqshift.c
new file mode 100644
index 0000000000..e83575813d
--- /dev/null
+++ b/libavfilter/af_afreqshift.c
@@ -0,0 +1,379 @@
+/*
+ * Copyright (c) Paul B Mahol
+ * Copyright (c) Laurent de Soras, 2005
+ *
+ * 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 "libavutil/channel_layout.h"
+#include "libavutil/ffmath.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "formats.h"
+
+#define NB_COEFS 16
+
+typedef struct AFreqShift {
+ const AVClass *class;
+
+ double shift;
+
+ double c[NB_COEFS];
+
+ int64_t in_samples;
+
+ AVFrame *i1, *o1;
+ AVFrame *i2, *o2;
+
+ void (*filter_channel)(AVFilterContext *ctx,
+ int nb_samples,
+ int sample_rate,
+ const double *src, double *dst,
+ double *i1, double *o1,
+ double *i2, double *o2);
+} AFreqShift;
+
+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);
+}
+
+static void pfilter_channel(AVFilterContext *ctx,
+ int nb_samples,
+ int sample_rate,
+ const double *src, double *dst,
+ double *i1, double *o1,
+ double *i2, double *o2)
+{
+ AFreqShift *s = ctx->priv;
+ double *c = s->c;
+ double shift = s->shift * M_PI;
+ double cos_theta = cos(shift);
+ double sin_theta = sin(shift);
+
+ for (int n = 0; n < nb_samples; n++) {
+ double xn1 = src[n], xn2 = src[n];
+ double I, Q;
+
+ for (int j = 0; j < NB_COEFS / 2; j++) {
+ I = c[j] * (xn1 + o2[j]) - i2[j];
+ i2[j] = i1[j];
+ i1[j] = xn1;
+ o2[j] = o1[j];
+ o1[j] = I;
+ xn1 = I;
+ }
+
+ for (int j = NB_COEFS / 2; j < NB_COEFS; j++) {
+ Q = c[j] * (xn2 + o2[j]) - i2[j];
+ i2[j] = i1[j];
+ i1[j] = xn2;
+ o2[j] = o1[j];
+ o1[j] = Q;
+ xn2 = Q;
+ }
+ Q = o2[NB_COEFS - 1];
+
+ dst[n] = I * cos_theta - Q * sin_theta;
+ }
+}
+
+static void ffilter_channel(AVFilterContext *ctx,
+ int nb_samples,
+ int sample_rate,
+ const double *src, double *dst,
+ double *i1, double *o1,
+ double *i2, double *o2)
+{
+ AFreqShift *s = ctx->priv;
+ double *c = s->c;
+ double ts = 1. / sample_rate;
+ double shift = s->shift;
+ int64_t N = s->in_samples;
+
+ for (int n = 0; n < nb_samples; n++) {
+ double xn1 = src[n], xn2 = src[n];
+ double I, Q, theta;
+
+ for (int j = 0; j < NB_COEFS / 2; j++) {
+ I = c[j] * (xn1 + o2[j]) - i2[j];
+ i2[j] = i1[j];
+ i1[j] = xn1;
+ o2[j] = o1[j];
+ o1[j] = I;
+ xn1 = I;
+ }
+
+ for (int j = NB_COEFS / 2; j < NB_COEFS; j++) {
+ Q = c[j] * (xn2 + o2[j]) - i2[j];
+ i2[j] = i1[j];
+ i1[j] = xn2;
+ o2[j] = o1[j];
+ o1[j] = Q;
+ xn2 = Q;
+ }
+ Q = o2[NB_COEFS - 1];
+
+ theta = 2. * M_PI * fmod(shift * (N + n) * ts, 1.);
+ dst[n] = I * cos(theta) - Q * sin(theta);
+ }
+}
+
+static void compute_transition_param(double *K, double *Q, double transition)
+{
+ double kksqrt, e, e2, e4, k, q;
+
+ k = tan((1. - transition * 2.) * M_PI / 4.);
+ k *= k;
+ kksqrt = pow(1 - k * k, 0.25);
+ e = 0.5 * (1. - kksqrt) / (1. + kksqrt);
+ e2 = e * e;
+ e4 = e2 * e2;
+ q = e * (1. + e4 * (2. + e4 * (15. + 150. * e4)));
+
+ *Q = q;
+ *K = k;
+}
+
+static double ipowp(double x, int64_t n)
+{
+ double z = 1.;
+
+ while (n != 0) {
+ if (n & 1)
+ z *= x;
+ n >>= 1;
+ x *= x;
+ }
+
+ return z;
+}
+
+static double compute_acc_num(double q, int order, int c)
+{
+ int64_t i = 0;
+ int j = 1;
+ double acc = 0.;
+ double q_ii1;
+
+ do {
+ q_ii1 = ipowp(q, i * (i + 1));
+ q_ii1 *= sin((i * 2 + 1) * c * M_PI / order) * j;
+ acc += q_ii1;
+
+ j = -j;
+ i++;
+ } while (fabs(q_ii1) > 1e-100);
+
+ return acc;
+}
+
+static double compute_acc_den(double q, int order, int c)
+{
+ int64_t i = 1;
+ int j = -1;
+ double acc = 0.;
+ double q_i2;
+
+ do {
+ q_i2 = ipowp(q, i * i);
+ q_i2 *= cos(i * 2 * c * M_PI / order) * j;
+ acc += q_i2;
+
+ j = -j;
+ i++;
+ } while (fabs(q_i2) > 1e-100);
+
+ return acc;
+}
+
+static double compute_coef(int index, double k, double q, int order)
+{
+ const int c = index + 1;
+ const double num = compute_acc_num(q, order, c) * pow(q, 0.25);
+ const double den = compute_acc_den(q, order, c) + 0.5;
+ const double ww = num / den;
+ const double wwsq = ww * ww;
+
+ const double x = sqrt((1 - wwsq * k) * (1 - wwsq / k)) / (1 + wwsq);
+ const double coef = (1 - x) / (1 + x);
+
+ return coef;
+}
+
+static void compute_coefs(double *coef_arr, int nbr_coefs, double transition)
+{
+ const int order = nbr_coefs * 2 + 1;
+ double k, q;
+
+ compute_transition_param(&k, &q, transition);
+
+ for (int n = 0; n < nbr_coefs; n++)
+ coef_arr[(n / 2) + (n & 1) * nbr_coefs / 2] = compute_coef(n, k, q, order);
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AFreqShift *s = ctx->priv;
+
+ compute_coefs(s->c, NB_COEFS, 2. * 20. / inlink->sample_rate);
+
+ s->i1 = ff_get_audio_buffer(inlink, NB_COEFS);
+ s->o1 = ff_get_audio_buffer(inlink, NB_COEFS);
+ s->i2 = ff_get_audio_buffer(inlink, NB_COEFS);
+ s->o2 = ff_get_audio_buffer(inlink, NB_COEFS);
+ if (!s->i1 || !s->o1 || !s->i2 || !s->o2)
+ return AVERROR(ENOMEM);
+
+ if (!strcmp(ctx->filter->name, "afreqshift"))
+ s->filter_channel = ffilter_channel;
+ else
+ s->filter_channel = pfilter_channel;
+
+ return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AVFilterLink *outlink = ctx->outputs[0];
+ AFreqShift *s = ctx->priv;
+ AVFrame *out;
+
+ if (av_frame_is_writable(in)) {
+ out = in;
+ } else {
+ out = ff_get_audio_buffer(outlink, in->nb_samples);
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+ av_frame_copy_props(out, in);
+ }
+
+ for (int ch = 0; ch < in->channels; ch++) {
+ s->filter_channel(ctx, in->nb_samples,
+ in->sample_rate,
+ (const double *)in->extended_data[ch],
+ (double *)out->extended_data[ch],
+ (double *)s->i1->extended_data[ch],
+ (double *)s->o1->extended_data[ch],
+ (double *)s->i2->extended_data[ch],
+ (double *)s->o2->extended_data[ch]);
+ }
+
+ s->in_samples += in->nb_samples;
+
+ if (out != in)
+ av_frame_free(&in);
+ return ff_filter_frame(outlink, out);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ AFreqShift *s = ctx->priv;
+
+ av_frame_free(&s->i1);
+ av_frame_free(&s->o1);
+ av_frame_free(&s->i2);
+ av_frame_free(&s->o2);
+}
+
+#define OFFSET(x) offsetof(AFreqShift, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
+
+static const AVOption afreqshift_options[] = {
+ { "shift", "set frequency shift", OFFSET(shift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -INT_MAX, INT_MAX, FLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(afreqshift);
+
+static const AVFilterPad inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ .config_props = config_input,
+ },
+ { NULL }
+};
+
+static const AVFilterPad outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_afreqshift = {
+ .name = "afreqshift",
+ .description = NULL_IF_CONFIG_SMALL("Apply frequency shifting to input audio."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(AFreqShift),
+ .priv_class = &afreqshift_class,
+ .uninit = uninit,
+ .inputs = inputs,
+ .outputs = outputs,
+ .process_command = ff_filter_process_command,
+};
+
+static const AVOption aphaseshift_options[] = {
+ { "shift", "set phase shift", OFFSET(shift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1.0, 1.0, FLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(aphaseshift);
+
+AVFilter ff_af_aphaseshift = {
+ .name = "aphaseshift",
+ .description = NULL_IF_CONFIG_SMALL("Apply phase shifting to input audio."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(AFreqShift),
+ .priv_class = &aphaseshift_class,
+ .uninit = uninit,
+ .inputs = inputs,
+ .outputs = outputs,
+ .process_command = ff_filter_process_command,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index fa91e608e4..801c53f7c0 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -43,6 +43,7 @@ extern AVFilter ff_af_afftdn;
extern AVFilter ff_af_afftfilt;
extern AVFilter ff_af_afir;
extern AVFilter ff_af_aformat;
+extern AVFilter ff_af_afreqshift;
extern AVFilter ff_af_agate;
extern AVFilter ff_af_aiir;
extern AVFilter ff_af_aintegral;
@@ -61,6 +62,7 @@ extern AVFilter ff_af_anull;
extern AVFilter ff_af_apad;
extern AVFilter ff_af_aperms;
extern AVFilter ff_af_aphaser;
+extern AVFilter ff_af_aphaseshift;
extern AVFilter ff_af_apulsator;
extern AVFilter ff_af_arealtime;
extern AVFilter ff_af_aresample;
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 308fbe07c3..b8ba489da7 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 7
-#define LIBAVFILTER_VERSION_MINOR 87
+#define LIBAVFILTER_VERSION_MINOR 88
#define LIBAVFILTER_VERSION_MICRO 100
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