diff options
| -rw-r--r-- | Changelog | 1 | ||||
| -rw-r--r-- | doc/filters.texi | 30 | ||||
| -rw-r--r-- | libavfilter/Makefile | 2 | ||||
| -rw-r--r-- | libavfilter/af_afreqshift.c | 379 | ||||
| -rw-r--r-- | libavfilter/allfilters.c | 2 | ||||
| -rw-r--r-- | libavfilter/version.h | 2 |
6 files changed, 415 insertions, 1 deletions
@@ -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 +@section afreqshift +Apply frequency shift to input audio samples. + +The filter accepts the following options: + +@table @option +@item shift +Specify frequency shift. Allowed range is -INT_MAX to INT_MAX. +Default value is 0.0. +@end table + +@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 +@section aphaseshift +Apply phase shift to input audio samples. + +The filter accepts the following options: + +@table @option +@item shift +Specify phase shift. Allowed range is from -1.0 to 1.0. +Default value is 0.0. +@end table + +@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 |