mirror of
https://github.com/FFmpeg/FFmpeg.git
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385 lines
12 KiB
C
385 lines
12 KiB
C
/*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/channel_layout.h"
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#include "libavutil/ffmath.h"
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#include "libavutil/opt.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "filters.h"
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#include "formats.h"
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typedef struct CrossfeedContext {
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const AVClass *class;
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double range;
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double strength;
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double slope;
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double level_in;
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double level_out;
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int block_samples;
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int block_size;
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double a0, a1, a2;
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double b0, b1, b2;
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double w1, w2;
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int64_t pts;
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int nb_samples;
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double *mid;
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double *side[3];
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} CrossfeedContext;
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats = NULL;
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AVFilterChannelLayouts *layout = NULL;
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int ret;
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if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 ||
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(ret = ff_set_common_formats (ctx , formats )) < 0 ||
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(ret = ff_add_channel_layout (&layout , &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO)) < 0 ||
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(ret = ff_set_common_channel_layouts (ctx , layout )) < 0 ||
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(ret = ff_set_common_all_samplerates (ctx )) < 0)
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return ret;
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return 0;
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}
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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CrossfeedContext *s = ctx->priv;
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double A = ff_exp10(s->strength * -30 / 40);
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double w0 = 2 * M_PI * (1. - s->range) * 2100 / inlink->sample_rate;
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double alpha;
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alpha = sin(w0) / 2 * sqrt((A + 1 / A) * (1 / s->slope - 1) + 2);
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s->a0 = (A + 1) + (A - 1) * cos(w0) + 2 * sqrt(A) * alpha;
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s->a1 = -2 * ((A - 1) + (A + 1) * cos(w0));
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s->a2 = (A + 1) + (A - 1) * cos(w0) - 2 * sqrt(A) * alpha;
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s->b0 = A * ((A + 1) - (A - 1) * cos(w0) + 2 * sqrt(A) * alpha);
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s->b1 = 2 * A * ((A - 1) - (A + 1) * cos(w0));
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s->b2 = A * ((A + 1) - (A - 1) * cos(w0) - 2 * sqrt(A) * alpha);
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s->a1 /= s->a0;
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s->a2 /= s->a0;
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s->b0 /= s->a0;
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s->b1 /= s->a0;
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s->b2 /= s->a0;
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if (s->block_samples == 0 && s->block_size > 0) {
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s->block_samples = s->block_size;
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s->mid = av_calloc(s->block_samples * 2, sizeof(*s->mid));
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for (int i = 0; i < 3; i++) {
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s->side[i] = av_calloc(s->block_samples * 2, sizeof(*s->side[0]));
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if (!s->side[i])
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return AVERROR(ENOMEM);
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}
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}
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return 0;
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}
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static void reverse_samples(double *dst, const double *src,
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int nb_samples)
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{
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for (int i = 0, j = nb_samples - 1; i < nb_samples; i++, j--)
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dst[i] = src[j];
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}
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static void filter_samples(double *dst, const double *src,
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int nb_samples,
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double b0, double b1, double b2,
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double a1, double a2,
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double *sw1, double *sw2)
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{
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double w1 = *sw1;
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double w2 = *sw2;
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for (int n = 0; n < nb_samples; n++) {
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double side = src[n];
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double oside = side * b0 + w1;
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w1 = b1 * side + w2 + a1 * oside;
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w2 = b2 * side + a2 * oside;
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dst[n] = oside;
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}
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*sw1 = w1;
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*sw2 = w2;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *in, int eof)
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{
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AVFilterContext *ctx = inlink->dst;
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AVFilterLink *outlink = ctx->outputs[0];
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CrossfeedContext *s = ctx->priv;
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const double *src = (const double *)in->data[0];
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const double level_in = s->level_in;
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const double level_out = s->level_out;
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const double b0 = s->b0;
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const double b1 = s->b1;
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const double b2 = s->b2;
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const double a1 = -s->a1;
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const double a2 = -s->a2;
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AVFrame *out;
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int drop = 0;
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double *dst;
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if (av_frame_is_writable(in) && s->block_samples == 0) {
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out = in;
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} else {
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out = ff_get_audio_buffer(outlink, s->block_samples > 0 ? s->block_samples : in->nb_samples);
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if (!out) {
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av_frame_free(&in);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(out, in);
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}
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dst = (double *)out->data[0];
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if (s->block_samples > 0 && s->pts == AV_NOPTS_VALUE)
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drop = 1;
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if (s->block_samples == 0) {
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double w1 = s->w1;
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double w2 = s->w2;
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for (int n = 0; n < out->nb_samples; n++, src += 2, dst += 2) {
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double mid = (src[0] + src[1]) * level_in * .5;
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double side = (src[0] - src[1]) * level_in * .5;
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double oside = side * b0 + w1;
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w1 = b1 * side + w2 + a1 * oside;
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w2 = b2 * side + a2 * oside;
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if (ctx->is_disabled) {
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dst[0] = src[0];
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dst[1] = src[1];
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} else {
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dst[0] = (mid + oside) * level_out;
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dst[1] = (mid - oside) * level_out;
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}
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}
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s->w1 = w1;
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s->w2 = w2;
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} else if (eof) {
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const double *src = (const double *)in->data[0];
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double *ssrc = s->side[1] + s->block_samples;
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double *msrc = s->mid;
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for (int n = 0; n < out->nb_samples; n++, src += 2, dst += 2) {
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if (ctx->is_disabled) {
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dst[0] = src[0];
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dst[1] = src[1];
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} else {
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dst[0] = (msrc[n] + ssrc[n]) * level_out;
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dst[1] = (msrc[n] - ssrc[n]) * level_out;
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}
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}
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} else {
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double *mdst = s->mid + s->block_samples;
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double *sdst = s->side[0] + s->block_samples;
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double *ssrc = s->side[0];
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double *msrc = s->mid;
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double w1 = s->w1;
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double w2 = s->w2;
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for (int n = 0; n < out->nb_samples; n++, src += 2) {
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mdst[n] = (src[0] + src[1]) * level_in * .5;
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sdst[n] = (src[0] - src[1]) * level_in * .5;
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}
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sdst = s->side[1];
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filter_samples(sdst, ssrc, s->block_samples,
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b0, b1, b2, a1, a2,
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&w1, &w2);
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s->w1 = w1;
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s->w2 = w2;
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ssrc = s->side[0] + s->block_samples;
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sdst = s->side[1] + s->block_samples;
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filter_samples(sdst, ssrc, s->block_samples,
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b0, b1, b2, a1, a2,
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&w1, &w2);
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reverse_samples(s->side[2], s->side[1], s->block_samples * 2);
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w1 = w2 = 0.;
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filter_samples(s->side[2], s->side[2], s->block_samples * 2,
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b0, b1, b2, a1, a2,
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&w1, &w2);
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reverse_samples(s->side[1], s->side[2], s->block_samples * 2);
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src = (const double *)in->data[0];
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ssrc = s->side[1];
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for (int n = 0; n < out->nb_samples; n++, src += 2, dst += 2) {
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if (ctx->is_disabled) {
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dst[0] = src[0];
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dst[1] = src[1];
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} else {
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dst[0] = (msrc[n] + ssrc[n]) * level_out;
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dst[1] = (msrc[n] - ssrc[n]) * level_out;
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}
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}
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memmove(s->mid, s->mid + s->block_samples,
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s->block_samples * sizeof(*s->mid));
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memmove(s->side[0], s->side[0] + s->block_samples,
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s->block_samples * sizeof(*s->side[0]));
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}
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if (s->block_samples > 0) {
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int nb_samples = in->nb_samples;
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int64_t pts = in->pts;
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out->pts = s->pts;
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out->nb_samples = s->nb_samples;
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s->pts = pts;
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s->nb_samples = nb_samples;
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}
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if (out != in)
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av_frame_free(&in);
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if (!drop) {
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return ff_filter_frame(outlink, out);
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} else {
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av_frame_free(&out);
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ff_filter_set_ready(ctx, 10);
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return 0;
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}
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}
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static int activate(AVFilterContext *ctx)
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{
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AVFilterLink *inlink = ctx->inputs[0];
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AVFilterLink *outlink = ctx->outputs[0];
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CrossfeedContext *s = ctx->priv;
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AVFrame *in = NULL;
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int64_t pts;
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int status;
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int ret;
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FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
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if (s->block_samples > 0) {
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ret = ff_inlink_consume_samples(inlink, s->block_samples, s->block_samples, &in);
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} else {
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ret = ff_inlink_consume_frame(inlink, &in);
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}
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if (ret < 0)
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return ret;
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if (ret > 0)
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return filter_frame(inlink, in, 0);
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if (s->block_samples > 0 && ff_inlink_queued_samples(inlink) >= s->block_samples) {
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ff_filter_set_ready(ctx, 10);
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return 0;
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}
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if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
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if (s->block_samples > 0) {
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AVFrame *in = ff_get_audio_buffer(outlink, s->block_samples);
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if (!in)
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return AVERROR(ENOMEM);
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ret = filter_frame(inlink, in, 1);
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}
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ff_outlink_set_status(outlink, status, pts);
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return ret;
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}
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FF_FILTER_FORWARD_WANTED(outlink, inlink);
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return FFERROR_NOT_READY;
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}
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static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
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char *res, int res_len, int flags)
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{
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int ret;
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ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
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if (ret < 0)
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return ret;
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return config_input(ctx->inputs[0]);
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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CrossfeedContext *s = ctx->priv;
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av_freep(&s->mid);
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for (int i = 0; i < 3; i++)
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av_freep(&s->side[i]);
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}
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#define OFFSET(x) offsetof(CrossfeedContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
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#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption crossfeed_options[] = {
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{ "strength", "set crossfeed strength", OFFSET(strength), AV_OPT_TYPE_DOUBLE, {.dbl=.2}, 0, 1, FLAGS },
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{ "range", "set soundstage wideness", OFFSET(range), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, 0, 1, FLAGS },
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{ "slope", "set curve slope", OFFSET(slope), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .01, 1, FLAGS },
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{ "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=.9}, 0, 1, FLAGS },
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{ "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1.}, 0, 1, FLAGS },
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{ "block_size", "set the block size", OFFSET(block_size),AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(crossfeed);
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static const AVFilterPad inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_input,
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},
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};
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static const AVFilterPad outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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};
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const AVFilter ff_af_crossfeed = {
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.name = "crossfeed",
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.description = NULL_IF_CONFIG_SMALL("Apply headphone crossfeed filter."),
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.priv_size = sizeof(CrossfeedContext),
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.priv_class = &crossfeed_class,
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.activate = activate,
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.uninit = uninit,
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FILTER_INPUTS(inputs),
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FILTER_OUTPUTS(outputs),
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FILTER_QUERY_FUNC(query_formats),
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.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
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.process_command = process_command,
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};
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