diff options
| -rw-r--r-- | Changelog | 1 | ||||
| -rw-r--r-- | doc/filters.texi | 56 | ||||
| -rw-r--r-- | libavfilter/Makefile | 1 | ||||
| -rw-r--r-- | libavfilter/af_aexciter.c | 317 | ||||
| -rw-r--r-- | libavfilter/allfilters.c | 1 | ||||
| -rw-r--r-- | libavfilter/version.h | 2 |
6 files changed, 377 insertions, 1 deletions
@@ -69,6 +69,7 @@ version <next>: - xbm_pipe demuxer - colorize filter - CRI parser +- aexciter audio filter version 4.3: diff --git a/doc/filters.texi b/doc/filters.texi index dc42927f6a..85052aa18d 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -1003,6 +1003,62 @@ aeval=val(0)|-val(1) @end example @end itemize +@section aexciter + +An exciter is used to produce high sound that is not present in the +original signal. This is done by creating harmonic distortions of the +signal which are restricted in range and added to the original signal. +An Exciter raises the upper end of an audio signal without simply raising +the higher frequencies like an equalizer would do to create a more +"crisp" or "brilliant" sound. + +The filter accepts the following options: + +@table @option +@item level_in +Set input level prior processing of signal. +Allowed range is from 0 to 64. +Default value is 1. + +@item level_out +Set output level after processing of signal. +Allowed range is from 0 to 64. +Default value is 1. + +@item amount +Set the amount of harmonics added to original signal. +Allowed range is from 0 to 64. +Default value is 1. + +@item drive +Set the amount of newly created harmonics. +Allowed range is from 0.1 to 10. +Default value is 8.5. + +@item blend +Set the octave of newly created harmonics. +Allowed range is from -10 to 10. +Default value is 0. + +@item freq +Set the lower frequency limit of producing harmonics in Hz. +Allowed range is from 2000 to 12000 Hz. +Default is 7500 Hz. + +@item ceil +Set the upper frequency limit of producing harmonics. +Allowed range is from 9999 to 20000 Hz. +If value is lower than 10000 Hz no limit is applied. + +@item listen +Mute the original signal and output only added harmonics. +By default is disabled. +@end table + +@subsection Commands + +This filter supports the all above options as @ref{commands}. + @anchor{afade} @section afade diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 3ec28df411..607a09287a 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -46,6 +46,7 @@ OBJS-$(CONFIG_ADERIVATIVE_FILTER) += af_aderivative.o OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o OBJS-$(CONFIG_AEMPHASIS_FILTER) += af_aemphasis.o OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o +OBJS-$(CONFIG_AEXCITER_FILTER) += af_aexciter.o OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o OBJS-$(CONFIG_AFFTDN_FILTER) += af_afftdn.o OBJS-$(CONFIG_AFFTFILT_FILTER) += af_afftfilt.o diff --git a/libavfilter/af_aexciter.c b/libavfilter/af_aexciter.c new file mode 100644 index 0000000000..f09c99984c --- /dev/null +++ b/libavfilter/af_aexciter.c @@ -0,0 +1,317 @@ +/* + * Copyright (c) Markus Schmidt and Christian Holschuh + * + * 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/opt.h" +#include "avfilter.h" +#include "internal.h" +#include "audio.h" + +typedef struct ChannelParams { + double blend_old, drive_old; + double rdrive, rbdr, kpa, kpb, kna, knb, ap, + an, imr, kc, srct, sq, pwrq; + double prev_med, prev_out; + + double hp[5], lp[5]; + double hw[4][2], lw[2][2]; +} ChannelParams; + +typedef struct AExciterContext { + const AVClass *class; + + double level_in; + double level_out; + double amount; + double drive; + double blend; + double freq; + double ceil; + int listen; + + ChannelParams *cp; +} AExciterContext; + +#define OFFSET(x) offsetof(AExciterContext, x) +#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM + +static const AVOption aexciter_options[] = { + { "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A }, + { "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A }, + { "amount", "set amount", OFFSET(amount), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A }, + { "drive", "set harmonics", OFFSET(drive), AV_OPT_TYPE_DOUBLE, {.dbl=8.5}, 0.1, 10, A }, + { "blend", "set blend harmonics", OFFSET(blend), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -10, 10, A }, + { "freq", "set scope", OFFSET(freq), AV_OPT_TYPE_DOUBLE, {.dbl=7500}, 2000, 12000, A }, + { "ceil", "set ceiling", OFFSET(ceil), AV_OPT_TYPE_DOUBLE, {.dbl=9999}, 9999, 20000, A }, + { "listen", "enable listen mode", OFFSET(listen), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(aexciter); + +static inline double M(double x) +{ + return (fabs(x) > 0.00000001) ? x : 0.0; +} + +static inline double D(double x) +{ + x = fabs(x); + + return (x > 0.00000001) ? sqrt(x) : 0.0; +} + +static void set_params(ChannelParams *p, + double blend, double drive, + double srate, double freq, + double ceil) +{ + double a0, a1, a2, b0, b1, b2, w0, alpha; + + p->rdrive = 12.0 / drive; + p->rbdr = p->rdrive / (10.5 - blend) * 780.0 / 33.0; + p->kpa = D(2.0 * (p->rdrive*p->rdrive) - 1.0) + 1.0; + p->kpb = (2.0 - p->kpa) / 2.0; + p->ap = ((p->rdrive*p->rdrive) - p->kpa + 1.0) / 2.0; + p->kc = p->kpa / D(2.0 * D(2.0 * (p->rdrive*p->rdrive) - 1.0) - 2.0 * p->rdrive*p->rdrive); + + p->srct = (0.1 * srate) / (0.1 * srate + 1.0); + p->sq = p->kc*p->kc + 1.0; + p->knb = -1.0 * p->rbdr / D(p->sq); + p->kna = 2.0 * p->kc * p->rbdr / D(p->sq); + p->an = p->rbdr*p->rbdr / p->sq; + p->imr = 2.0 * p->knb + D(2.0 * p->kna + 4.0 * p->an - 1.0); + p->pwrq = 2.0 / (p->imr + 1.0); + + w0 = 2 * M_PI * freq / srate; + alpha = sin(w0) / (2. * 0.707); + a0 = 1 + alpha; + a1 = -2 * cos(w0); + a2 = 1 - alpha; + b0 = (1 + cos(w0)) / 2; + b1 = -(1 + cos(w0)); + b2 = (1 + cos(w0)) / 2; + + p->hp[0] =-a1 / a0; + p->hp[1] =-a2 / a0; + p->hp[2] = b0 / a0; + p->hp[3] = b1 / a0; + p->hp[4] = b2 / a0; + + w0 = 2 * M_PI * ceil / srate; + alpha = sin(w0) / (2. * 0.707); + a0 = 1 + alpha; + a1 = -2 * cos(w0); + a2 = 1 - alpha; + b0 = (1 - cos(w0)) / 2; + b1 = 1 - cos(w0); + b2 = (1 - cos(w0)) / 2; + + p->lp[0] =-a1 / a0; + p->lp[1] =-a2 / a0; + p->lp[2] = b0 / a0; + p->lp[3] = b1 / a0; + p->lp[4] = b2 / a0; +} + +static double bprocess(double in, const double *const c, + double *w1, double *w2) +{ + double out = c[2] * in + *w1; + + *w1 = c[3] * in + *w2 + c[0] * out; + *w2 = c[4] * in + c[1] * out; + + return out; +} + +static double distortion_process(AExciterContext *s, ChannelParams *p, double in) +{ + double proc = in, med; + + proc = bprocess(proc, p->hp, &p->hw[0][0], &p->hw[0][1]); + proc = bprocess(proc, p->hp, &p->hw[1][0], &p->hw[1][1]); + + if (proc >= 0.0) { + med = (D(p->ap + proc * (p->kpa - proc)) + p->kpb) * p->pwrq; + } else { + med = (D(p->an - proc * (p->kna + proc)) + p->knb) * p->pwrq * -1.0; + } + + proc = p->srct * (med - p->prev_med + p->prev_out); + p->prev_med = M(med); + p->prev_out = M(proc); + + proc = bprocess(proc, p->hp, &p->hw[2][0], &p->hw[2][1]); + proc = bprocess(proc, p->hp, &p->hw[3][0], &p->hw[3][1]); + + if (s->ceil >= 10000.) { + proc = bprocess(proc, p->lp, &p->lw[0][0], &p->lw[0][1]); + proc = bprocess(proc, p->lp, &p->lw[1][0], &p->lw[1][1]); + } + + return proc; +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *in) +{ + AVFilterContext *ctx = inlink->dst; + AExciterContext *s = ctx->priv; + AVFilterLink *outlink = ctx->outputs[0]; + AVFrame *out; + const double *src = (const double *)in->data[0]; + const double level_in = s->level_in; + const double level_out = s->level_out; + const double amount = s->amount; + const double listen = 1.0 - s->listen; + double *dst; + + if (av_frame_is_writable(in)) { + out = in; + } else { + out = ff_get_audio_buffer(inlink, in->nb_samples); + if (!out) { + av_frame_free(&in); + return AVERROR(ENOMEM); + } + av_frame_copy_props(out, in); + } + + dst = (double *)out->data[0]; + for (int n = 0; n < in->nb_samples; n++) { + for (int c = 0; c < inlink->channels; c++) { + double sample = src[c] * level_in; + + sample = distortion_process(s, &s->cp[c], sample); + sample = sample * amount + listen * src[c]; + + sample *= level_out; + if (ctx->is_disabled) + dst[c] = src[c]; + else + dst[c] = sample; + } + + src += inlink->channels; + dst += inlink->channels; + } + + if (in != out) + av_frame_free(&in); + + return ff_filter_frame(outlink, out); +} + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterFormats *formats; + AVFilterChannelLayouts *layouts; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_DBL, + AV_SAMPLE_FMT_NONE + }; + int 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_make_format_list(sample_fmts); + if (!formats) + return AVERROR(ENOMEM); + ret = ff_set_common_formats(ctx, formats); + if (ret < 0) + return ret; + + formats = ff_all_samplerates(); + if (!formats) + return AVERROR(ENOMEM); + return ff_set_common_samplerates(ctx, formats); +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + AExciterContext *s = ctx->priv; + + av_freep(&s->cp); +} + +static int config_input(AVFilterLink *inlink) +{ + AVFilterContext *ctx = inlink->dst; + AExciterContext *s = ctx->priv; + + if (!s->cp) + s->cp = av_calloc(inlink->channels, sizeof(*s->cp)); + if (!s->cp) + return AVERROR(ENOMEM); + + for (int i = 0; i < inlink->channels; i++) + set_params(&s->cp[i], s->blend, s->drive, inlink->sample_rate, + s->freq, s->ceil); + + return 0; +} + +static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, + char *res, int res_len, int flags) +{ + AVFilterLink *inlink = ctx->inputs[0]; + int ret; + + ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags); + if (ret < 0) + return ret; + + return config_input(inlink); +} + +static const AVFilterPad avfilter_af_aexciter_inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = config_input, + .filter_frame = filter_frame, + }, + { NULL } +}; + +static const AVFilterPad avfilter_af_aexciter_outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + }, + { NULL } +}; + +AVFilter ff_af_aexciter = { + .name = "aexciter", + .description = NULL_IF_CONFIG_SMALL("Enhance high frequency part of audio."), + .priv_size = sizeof(AExciterContext), + .priv_class = &aexciter_class, + .uninit = uninit, + .query_formats = query_formats, + .inputs = avfilter_af_aexciter_inputs, + .outputs = avfilter_af_aexciter_outputs, + .process_command = process_command, + .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 73d859ce5e..a00008b44a 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -39,6 +39,7 @@ extern AVFilter ff_af_aderivative; extern AVFilter ff_af_aecho; extern AVFilter ff_af_aemphasis; extern AVFilter ff_af_aeval; +extern AVFilter ff_af_aexciter; extern AVFilter ff_af_afade; extern AVFilter ff_af_afftdn; extern AVFilter ff_af_afftfilt; diff --git a/libavfilter/version.h b/libavfilter/version.h index 43924194ae..d74eeb3f7b 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 103 +#define LIBAVFILTER_VERSION_MINOR 104 #define LIBAVFILTER_VERSION_MICRO 100 |