| /* |
| * Copyright (c) 2018 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/intreadwrite.h" |
| #include "libavutil/opt.h" |
| #include "libavutil/xga_font_data.h" |
| #include "audio.h" |
| #include "avfilter.h" |
| #include "internal.h" |
| |
| typedef struct ThreadData { |
| AVFrame *in, *out; |
| } ThreadData; |
| |
| typedef struct Pair { |
| int a, b; |
| } Pair; |
| |
| typedef struct BiquadContext { |
| double a[3]; |
| double b[3]; |
| double w1, w2; |
| } BiquadContext; |
| |
| typedef struct IIRChannel { |
| int nb_ab[2]; |
| double *ab[2]; |
| double g; |
| double *cache[2]; |
| double fir; |
| BiquadContext *biquads; |
| int clippings; |
| } IIRChannel; |
| |
| typedef struct AudioIIRContext { |
| const AVClass *class; |
| char *a_str, *b_str, *g_str; |
| double dry_gain, wet_gain; |
| double mix; |
| int normalize; |
| int format; |
| int process; |
| int precision; |
| int response; |
| int w, h; |
| int ir_channel; |
| AVRational rate; |
| |
| AVFrame *video; |
| |
| IIRChannel *iir; |
| int channels; |
| enum AVSampleFormat sample_format; |
| |
| int (*iir_channel)(AVFilterContext *ctx, void *arg, int ch, int nb_jobs); |
| } AudioIIRContext; |
| |
| static int query_formats(AVFilterContext *ctx) |
| { |
| AudioIIRContext *s = ctx->priv; |
| AVFilterFormats *formats; |
| AVFilterChannelLayouts *layouts; |
| enum AVSampleFormat sample_fmts[] = { |
| AV_SAMPLE_FMT_DBLP, |
| AV_SAMPLE_FMT_NONE |
| }; |
| static const enum AVPixelFormat pix_fmts[] = { |
| AV_PIX_FMT_RGB0, |
| AV_PIX_FMT_NONE |
| }; |
| int ret; |
| |
| if (s->response) { |
| AVFilterLink *videolink = ctx->outputs[1]; |
| |
| formats = ff_make_format_list(pix_fmts); |
| if ((ret = ff_formats_ref(formats, &videolink->incfg.formats)) < 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; |
| |
| sample_fmts[0] = s->sample_format; |
| 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); |
| } |
| |
| #define IIR_CH(name, type, min, max, need_clipping) \ |
| static int iir_ch_## name(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) \ |
| { \ |
| AudioIIRContext *s = ctx->priv; \ |
| const double ig = s->dry_gain; \ |
| const double og = s->wet_gain; \ |
| const double mix = s->mix; \ |
| ThreadData *td = arg; \ |
| AVFrame *in = td->in, *out = td->out; \ |
| const type *src = (const type *)in->extended_data[ch]; \ |
| double *oc = (double *)s->iir[ch].cache[0]; \ |
| double *ic = (double *)s->iir[ch].cache[1]; \ |
| const int nb_a = s->iir[ch].nb_ab[0]; \ |
| const int nb_b = s->iir[ch].nb_ab[1]; \ |
| const double *a = s->iir[ch].ab[0]; \ |
| const double *b = s->iir[ch].ab[1]; \ |
| const double g = s->iir[ch].g; \ |
| int *clippings = &s->iir[ch].clippings; \ |
| type *dst = (type *)out->extended_data[ch]; \ |
| int n; \ |
| \ |
| for (n = 0; n < in->nb_samples; n++) { \ |
| double sample = 0.; \ |
| int x; \ |
| \ |
| memmove(&ic[1], &ic[0], (nb_b - 1) * sizeof(*ic)); \ |
| memmove(&oc[1], &oc[0], (nb_a - 1) * sizeof(*oc)); \ |
| ic[0] = src[n] * ig; \ |
| for (x = 0; x < nb_b; x++) \ |
| sample += b[x] * ic[x]; \ |
| \ |
| for (x = 1; x < nb_a; x++) \ |
| sample -= a[x] * oc[x]; \ |
| \ |
| oc[0] = sample; \ |
| sample *= og * g; \ |
| sample = sample * mix + ic[0] * (1. - mix); \ |
| if (need_clipping && sample < min) { \ |
| (*clippings)++; \ |
| dst[n] = min; \ |
| } else if (need_clipping && sample > max) { \ |
| (*clippings)++; \ |
| dst[n] = max; \ |
| } else { \ |
| dst[n] = sample; \ |
| } \ |
| } \ |
| \ |
| return 0; \ |
| } |
| |
| IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1) |
| IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1) |
| IIR_CH(fltp, float, -1., 1., 0) |
| IIR_CH(dblp, double, -1., 1., 0) |
| |
| #define SERIAL_IIR_CH(name, type, min, max, need_clipping) \ |
| static int iir_ch_serial_## name(AVFilterContext *ctx, void *arg, \ |
| int ch, int nb_jobs) \ |
| { \ |
| AudioIIRContext *s = ctx->priv; \ |
| const double ig = s->dry_gain; \ |
| const double og = s->wet_gain; \ |
| const double mix = s->mix; \ |
| const double imix = 1. - mix; \ |
| ThreadData *td = arg; \ |
| AVFrame *in = td->in, *out = td->out; \ |
| const type *src = (const type *)in->extended_data[ch]; \ |
| type *dst = (type *)out->extended_data[ch]; \ |
| IIRChannel *iir = &s->iir[ch]; \ |
| const double g = iir->g; \ |
| int *clippings = &iir->clippings; \ |
| int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2; \ |
| int n, i; \ |
| \ |
| for (i = nb_biquads - 1; i >= 0; i--) { \ |
| const double a1 = -iir->biquads[i].a[1]; \ |
| const double a2 = -iir->biquads[i].a[2]; \ |
| const double b0 = iir->biquads[i].b[0]; \ |
| const double b1 = iir->biquads[i].b[1]; \ |
| const double b2 = iir->biquads[i].b[2]; \ |
| double w1 = iir->biquads[i].w1; \ |
| double w2 = iir->biquads[i].w2; \ |
| \ |
| for (n = 0; n < in->nb_samples; n++) { \ |
| double i0 = ig * (i ? dst[n] : src[n]); \ |
| double o0 = i0 * b0 + w1; \ |
| \ |
| w1 = b1 * i0 + w2 + a1 * o0; \ |
| w2 = b2 * i0 + a2 * o0; \ |
| o0 *= og * g; \ |
| \ |
| o0 = o0 * mix + imix * i0; \ |
| if (need_clipping && o0 < min) { \ |
| (*clippings)++; \ |
| dst[n] = min; \ |
| } else if (need_clipping && o0 > max) { \ |
| (*clippings)++; \ |
| dst[n] = max; \ |
| } else { \ |
| dst[n] = o0; \ |
| } \ |
| } \ |
| iir->biquads[i].w1 = w1; \ |
| iir->biquads[i].w2 = w2; \ |
| } \ |
| \ |
| return 0; \ |
| } |
| |
| SERIAL_IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1) |
| SERIAL_IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1) |
| SERIAL_IIR_CH(fltp, float, -1., 1., 0) |
| SERIAL_IIR_CH(dblp, double, -1., 1., 0) |
| |
| #define PARALLEL_IIR_CH(name, type, min, max, need_clipping) \ |
| static int iir_ch_parallel_## name(AVFilterContext *ctx, void *arg, \ |
| int ch, int nb_jobs) \ |
| { \ |
| AudioIIRContext *s = ctx->priv; \ |
| const double ig = s->dry_gain; \ |
| const double og = s->wet_gain; \ |
| const double mix = s->mix; \ |
| const double imix = 1. - mix; \ |
| ThreadData *td = arg; \ |
| AVFrame *in = td->in, *out = td->out; \ |
| const type *src = (const type *)in->extended_data[ch]; \ |
| type *dst = (type *)out->extended_data[ch]; \ |
| IIRChannel *iir = &s->iir[ch]; \ |
| const double g = iir->g; \ |
| const double fir = iir->fir; \ |
| int *clippings = &iir->clippings; \ |
| int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2; \ |
| int n, i; \ |
| \ |
| for (i = 0; i < nb_biquads; i++) { \ |
| const double a1 = -iir->biquads[i].a[1]; \ |
| const double a2 = -iir->biquads[i].a[2]; \ |
| const double b1 = iir->biquads[i].b[1]; \ |
| const double b2 = iir->biquads[i].b[2]; \ |
| double w1 = iir->biquads[i].w1; \ |
| double w2 = iir->biquads[i].w2; \ |
| \ |
| for (n = 0; n < in->nb_samples; n++) { \ |
| double i0 = ig * src[n]; \ |
| double o0 = w1; \ |
| \ |
| w1 = b1 * i0 + w2 + a1 * o0; \ |
| w2 = b2 * i0 + a2 * o0; \ |
| o0 *= og * g; \ |
| o0 += dst[n]; \ |
| \ |
| if (need_clipping && o0 < min) { \ |
| (*clippings)++; \ |
| dst[n] = min; \ |
| } else if (need_clipping && o0 > max) { \ |
| (*clippings)++; \ |
| dst[n] = max; \ |
| } else { \ |
| dst[n] = o0; \ |
| } \ |
| } \ |
| iir->biquads[i].w1 = w1; \ |
| iir->biquads[i].w2 = w2; \ |
| } \ |
| \ |
| for (n = 0; n < in->nb_samples; n++) { \ |
| dst[n] += fir * src[n]; \ |
| dst[n] = dst[n] * mix + imix * src[n]; \ |
| } \ |
| \ |
| return 0; \ |
| } |
| |
| PARALLEL_IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1) |
| PARALLEL_IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1) |
| PARALLEL_IIR_CH(fltp, float, -1., 1., 0) |
| PARALLEL_IIR_CH(dblp, double, -1., 1., 0) |
| |
| #define LATTICE_IIR_CH(name, type, min, max, need_clipping) \ |
| static int iir_ch_lattice_## name(AVFilterContext *ctx, void *arg, \ |
| int ch, int nb_jobs) \ |
| { \ |
| AudioIIRContext *s = ctx->priv; \ |
| const double ig = s->dry_gain; \ |
| const double og = s->wet_gain; \ |
| const double mix = s->mix; \ |
| ThreadData *td = arg; \ |
| AVFrame *in = td->in, *out = td->out; \ |
| const type *src = (const type *)in->extended_data[ch]; \ |
| double n0, n1, p0, *x = (double *)s->iir[ch].cache[0]; \ |
| const int nb_stages = s->iir[ch].nb_ab[1]; \ |
| const double *v = s->iir[ch].ab[0]; \ |
| const double *k = s->iir[ch].ab[1]; \ |
| const double g = s->iir[ch].g; \ |
| int *clippings = &s->iir[ch].clippings; \ |
| type *dst = (type *)out->extended_data[ch]; \ |
| int n; \ |
| \ |
| for (n = 0; n < in->nb_samples; n++) { \ |
| const double in = src[n] * ig; \ |
| double out = 0.; \ |
| \ |
| n1 = in; \ |
| for (int i = nb_stages - 1; i >= 0; i--) { \ |
| n0 = n1 - k[i] * x[i]; \ |
| p0 = n0 * k[i] + x[i]; \ |
| out += p0 * v[i+1]; \ |
| x[i] = p0; \ |
| n1 = n0; \ |
| } \ |
| \ |
| out += n1 * v[0]; \ |
| memmove(&x[1], &x[0], nb_stages * sizeof(*x)); \ |
| x[0] = n1; \ |
| out *= og * g; \ |
| out = out * mix + in * (1. - mix); \ |
| if (need_clipping && out < min) { \ |
| (*clippings)++; \ |
| dst[n] = min; \ |
| } else if (need_clipping && out > max) { \ |
| (*clippings)++; \ |
| dst[n] = max; \ |
| } else { \ |
| dst[n] = out; \ |
| } \ |
| } \ |
| \ |
| return 0; \ |
| } |
| |
| LATTICE_IIR_CH(s16p, int16_t, INT16_MIN, INT16_MAX, 1) |
| LATTICE_IIR_CH(s32p, int32_t, INT32_MIN, INT32_MAX, 1) |
| LATTICE_IIR_CH(fltp, float, -1., 1., 0) |
| LATTICE_IIR_CH(dblp, double, -1., 1., 0) |
| |
| static void count_coefficients(char *item_str, int *nb_items) |
| { |
| char *p; |
| |
| if (!item_str) |
| return; |
| |
| *nb_items = 1; |
| for (p = item_str; *p && *p != '|'; p++) { |
| if (*p == ' ') |
| (*nb_items)++; |
| } |
| } |
| |
| static int read_gains(AVFilterContext *ctx, char *item_str, int nb_items) |
| { |
| AudioIIRContext *s = ctx->priv; |
| char *p, *arg, *old_str, *prev_arg = NULL, *saveptr = NULL; |
| int i; |
| |
| p = old_str = av_strdup(item_str); |
| if (!p) |
| return AVERROR(ENOMEM); |
| for (i = 0; i < nb_items; i++) { |
| if (!(arg = av_strtok(p, "|", &saveptr))) |
| arg = prev_arg; |
| |
| if (!arg) { |
| av_freep(&old_str); |
| return AVERROR(EINVAL); |
| } |
| |
| p = NULL; |
| if (av_sscanf(arg, "%lf", &s->iir[i].g) != 1) { |
| av_log(ctx, AV_LOG_ERROR, "Invalid gains supplied: %s\n", arg); |
| av_freep(&old_str); |
| return AVERROR(EINVAL); |
| } |
| |
| prev_arg = arg; |
| } |
| |
| av_freep(&old_str); |
| |
| return 0; |
| } |
| |
| static int read_tf_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst) |
| { |
| char *p, *arg, *old_str, *saveptr = NULL; |
| int i; |
| |
| p = old_str = av_strdup(item_str); |
| if (!p) |
| return AVERROR(ENOMEM); |
| for (i = 0; i < nb_items; i++) { |
| if (!(arg = av_strtok(p, " ", &saveptr))) |
| break; |
| |
| p = NULL; |
| if (av_sscanf(arg, "%lf", &dst[i]) != 1) { |
| av_log(ctx, AV_LOG_ERROR, "Invalid coefficients supplied: %s\n", arg); |
| av_freep(&old_str); |
| return AVERROR(EINVAL); |
| } |
| } |
| |
| av_freep(&old_str); |
| |
| return 0; |
| } |
| |
| static int read_zp_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst, const char *format) |
| { |
| char *p, *arg, *old_str, *saveptr = NULL; |
| int i; |
| |
| p = old_str = av_strdup(item_str); |
| if (!p) |
| return AVERROR(ENOMEM); |
| for (i = 0; i < nb_items; i++) { |
| if (!(arg = av_strtok(p, " ", &saveptr))) |
| break; |
| |
| p = NULL; |
| if (av_sscanf(arg, format, &dst[i*2], &dst[i*2+1]) != 2) { |
| av_log(ctx, AV_LOG_ERROR, "Invalid coefficients supplied: %s\n", arg); |
| av_freep(&old_str); |
| return AVERROR(EINVAL); |
| } |
| } |
| |
| av_freep(&old_str); |
| |
| return 0; |
| } |
| |
| static const char *format[] = { "%lf", "%lf %lfi", "%lf %lfr", "%lf %lfd", "%lf %lfi" }; |
| |
| static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str, int ab) |
| { |
| AudioIIRContext *s = ctx->priv; |
| char *p, *arg, *old_str, *prev_arg = NULL, *saveptr = NULL; |
| int i, ret; |
| |
| p = old_str = av_strdup(item_str); |
| if (!p) |
| return AVERROR(ENOMEM); |
| for (i = 0; i < channels; i++) { |
| IIRChannel *iir = &s->iir[i]; |
| |
| if (!(arg = av_strtok(p, "|", &saveptr))) |
| arg = prev_arg; |
| |
| if (!arg) { |
| av_freep(&old_str); |
| return AVERROR(EINVAL); |
| } |
| |
| count_coefficients(arg, &iir->nb_ab[ab]); |
| |
| p = NULL; |
| iir->cache[ab] = av_calloc(iir->nb_ab[ab] + 1, sizeof(double)); |
| iir->ab[ab] = av_calloc(iir->nb_ab[ab] * (!!s->format + 1), sizeof(double)); |
| if (!iir->ab[ab] || !iir->cache[ab]) { |
| av_freep(&old_str); |
| return AVERROR(ENOMEM); |
| } |
| |
| if (s->format > 0) { |
| ret = read_zp_coefficients(ctx, arg, iir->nb_ab[ab], iir->ab[ab], format[s->format]); |
| } else { |
| ret = read_tf_coefficients(ctx, arg, iir->nb_ab[ab], iir->ab[ab]); |
| } |
| if (ret < 0) { |
| av_freep(&old_str); |
| return ret; |
| } |
| prev_arg = arg; |
| } |
| |
| av_freep(&old_str); |
| |
| return 0; |
| } |
| |
| static void cmul(double re, double im, double re2, double im2, double *RE, double *IM) |
| { |
| *RE = re * re2 - im * im2; |
| *IM = re * im2 + re2 * im; |
| } |
| |
| static int expand(AVFilterContext *ctx, double *pz, int n, double *coefs) |
| { |
| coefs[2 * n] = 1.0; |
| |
| for (int i = 1; i <= n; i++) { |
| for (int j = n - i; j < n; j++) { |
| double re, im; |
| |
| cmul(coefs[2 * (j + 1)], coefs[2 * (j + 1) + 1], |
| pz[2 * (i - 1)], pz[2 * (i - 1) + 1], &re, &im); |
| |
| coefs[2 * j] -= re; |
| coefs[2 * j + 1] -= im; |
| } |
| } |
| |
| for (int i = 0; i < n + 1; i++) { |
| if (fabs(coefs[2 * i + 1]) > FLT_EPSILON) { |
| av_log(ctx, AV_LOG_ERROR, "coefs: %f of z^%d is not real; poles/zeros are not complex conjugates.\n", |
| coefs[2 * i + 1], i); |
| return AVERROR(EINVAL); |
| } |
| } |
| |
| return 0; |
| } |
| |
| static void normalize_coeffs(AVFilterContext *ctx, int ch) |
| { |
| AudioIIRContext *s = ctx->priv; |
| IIRChannel *iir = &s->iir[ch]; |
| double sum_den = 0.; |
| |
| if (!s->normalize) |
| return; |
| |
| for (int i = 0; i < iir->nb_ab[1]; i++) { |
| sum_den += iir->ab[1][i]; |
| } |
| |
| if (sum_den > 1e-6) { |
| double factor, sum_num = 0.; |
| |
| for (int i = 0; i < iir->nb_ab[0]; i++) { |
| sum_num += iir->ab[0][i]; |
| } |
| |
| factor = sum_num / sum_den; |
| |
| for (int i = 0; i < iir->nb_ab[1]; i++) { |
| iir->ab[1][i] *= factor; |
| } |
| } |
| } |
| |
| static int convert_zp2tf(AVFilterContext *ctx, int channels) |
| { |
| AudioIIRContext *s = ctx->priv; |
| int ch, i, j, ret = 0; |
| |
| for (ch = 0; ch < channels; ch++) { |
| IIRChannel *iir = &s->iir[ch]; |
| double *topc, *botc; |
| |
| topc = av_calloc((iir->nb_ab[1] + 1) * 2, sizeof(*topc)); |
| botc = av_calloc((iir->nb_ab[0] + 1) * 2, sizeof(*botc)); |
| if (!topc || !botc) { |
| ret = AVERROR(ENOMEM); |
| goto fail; |
| } |
| |
| ret = expand(ctx, iir->ab[0], iir->nb_ab[0], botc); |
| if (ret < 0) { |
| goto fail; |
| } |
| |
| ret = expand(ctx, iir->ab[1], iir->nb_ab[1], topc); |
| if (ret < 0) { |
| goto fail; |
| } |
| |
| for (j = 0, i = iir->nb_ab[1]; i >= 0; j++, i--) { |
| iir->ab[1][j] = topc[2 * i]; |
| } |
| iir->nb_ab[1]++; |
| |
| for (j = 0, i = iir->nb_ab[0]; i >= 0; j++, i--) { |
| iir->ab[0][j] = botc[2 * i]; |
| } |
| iir->nb_ab[0]++; |
| |
| normalize_coeffs(ctx, ch); |
| |
| fail: |
| av_free(topc); |
| av_free(botc); |
| if (ret < 0) |
| break; |
| } |
| |
| return ret; |
| } |
| |
| static int decompose_zp2biquads(AVFilterContext *ctx, int channels) |
| { |
| AudioIIRContext *s = ctx->priv; |
| int ch, ret; |
| |
| for (ch = 0; ch < channels; ch++) { |
| IIRChannel *iir = &s->iir[ch]; |
| int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2; |
| int current_biquad = 0; |
| |
| iir->biquads = av_calloc(nb_biquads, sizeof(BiquadContext)); |
| if (!iir->biquads) |
| return AVERROR(ENOMEM); |
| |
| while (nb_biquads--) { |
| Pair outmost_pole = { -1, -1 }; |
| Pair nearest_zero = { -1, -1 }; |
| double zeros[4] = { 0 }; |
| double poles[4] = { 0 }; |
| double b[6] = { 0 }; |
| double a[6] = { 0 }; |
| double min_distance = DBL_MAX; |
| double max_mag = 0; |
| double factor; |
| int i; |
| |
| for (i = 0; i < iir->nb_ab[0]; i++) { |
| double mag; |
| |
| if (isnan(iir->ab[0][2 * i]) || isnan(iir->ab[0][2 * i + 1])) |
| continue; |
| mag = hypot(iir->ab[0][2 * i], iir->ab[0][2 * i + 1]); |
| |
| if (mag > max_mag) { |
| max_mag = mag; |
| outmost_pole.a = i; |
| } |
| } |
| |
| for (i = 0; i < iir->nb_ab[0]; i++) { |
| if (isnan(iir->ab[0][2 * i]) || isnan(iir->ab[0][2 * i + 1])) |
| continue; |
| |
| if (iir->ab[0][2 * i ] == iir->ab[0][2 * outmost_pole.a ] && |
| iir->ab[0][2 * i + 1] == -iir->ab[0][2 * outmost_pole.a + 1]) { |
| outmost_pole.b = i; |
| break; |
| } |
| } |
| |
| av_log(ctx, AV_LOG_VERBOSE, "outmost_pole is %d.%d\n", outmost_pole.a, outmost_pole.b); |
| |
| if (outmost_pole.a < 0 || outmost_pole.b < 0) |
| return AVERROR(EINVAL); |
| |
| for (i = 0; i < iir->nb_ab[1]; i++) { |
| double distance; |
| |
| if (isnan(iir->ab[1][2 * i]) || isnan(iir->ab[1][2 * i + 1])) |
| continue; |
| distance = hypot(iir->ab[0][2 * outmost_pole.a ] - iir->ab[1][2 * i ], |
| iir->ab[0][2 * outmost_pole.a + 1] - iir->ab[1][2 * i + 1]); |
| |
| if (distance < min_distance) { |
| min_distance = distance; |
| nearest_zero.a = i; |
| } |
| } |
| |
| for (i = 0; i < iir->nb_ab[1]; i++) { |
| if (isnan(iir->ab[1][2 * i]) || isnan(iir->ab[1][2 * i + 1])) |
| continue; |
| |
| if (iir->ab[1][2 * i ] == iir->ab[1][2 * nearest_zero.a ] && |
| iir->ab[1][2 * i + 1] == -iir->ab[1][2 * nearest_zero.a + 1]) { |
| nearest_zero.b = i; |
| break; |
| } |
| } |
| |
| av_log(ctx, AV_LOG_VERBOSE, "nearest_zero is %d.%d\n", nearest_zero.a, nearest_zero.b); |
| |
| if (nearest_zero.a < 0 || nearest_zero.b < 0) |
| return AVERROR(EINVAL); |
| |
| poles[0] = iir->ab[0][2 * outmost_pole.a ]; |
| poles[1] = iir->ab[0][2 * outmost_pole.a + 1]; |
| |
| zeros[0] = iir->ab[1][2 * nearest_zero.a ]; |
| zeros[1] = iir->ab[1][2 * nearest_zero.a + 1]; |
| |
| if (nearest_zero.a == nearest_zero.b && outmost_pole.a == outmost_pole.b) { |
| zeros[2] = 0; |
| zeros[3] = 0; |
| |
| poles[2] = 0; |
| poles[3] = 0; |
| } else { |
| poles[2] = iir->ab[0][2 * outmost_pole.b ]; |
| poles[3] = iir->ab[0][2 * outmost_pole.b + 1]; |
| |
| zeros[2] = iir->ab[1][2 * nearest_zero.b ]; |
| zeros[3] = iir->ab[1][2 * nearest_zero.b + 1]; |
| } |
| |
| ret = expand(ctx, zeros, 2, b); |
| if (ret < 0) |
| return ret; |
| |
| ret = expand(ctx, poles, 2, a); |
| if (ret < 0) |
| return ret; |
| |
| iir->ab[0][2 * outmost_pole.a] = iir->ab[0][2 * outmost_pole.a + 1] = NAN; |
| iir->ab[0][2 * outmost_pole.b] = iir->ab[0][2 * outmost_pole.b + 1] = NAN; |
| iir->ab[1][2 * nearest_zero.a] = iir->ab[1][2 * nearest_zero.a + 1] = NAN; |
| iir->ab[1][2 * nearest_zero.b] = iir->ab[1][2 * nearest_zero.b + 1] = NAN; |
| |
| iir->biquads[current_biquad].a[0] = 1.; |
| iir->biquads[current_biquad].a[1] = a[2] / a[4]; |
| iir->biquads[current_biquad].a[2] = a[0] / a[4]; |
| iir->biquads[current_biquad].b[0] = b[4] / a[4]; |
| iir->biquads[current_biquad].b[1] = b[2] / a[4]; |
| iir->biquads[current_biquad].b[2] = b[0] / a[4]; |
| |
| if (s->normalize && |
| fabs(iir->biquads[current_biquad].b[0] + |
| iir->biquads[current_biquad].b[1] + |
| iir->biquads[current_biquad].b[2]) > 1e-6) { |
| factor = (iir->biquads[current_biquad].a[0] + |
| iir->biquads[current_biquad].a[1] + |
| iir->biquads[current_biquad].a[2]) / |
| (iir->biquads[current_biquad].b[0] + |
| iir->biquads[current_biquad].b[1] + |
| iir->biquads[current_biquad].b[2]); |
| |
| av_log(ctx, AV_LOG_VERBOSE, "factor=%f\n", factor); |
| |
| iir->biquads[current_biquad].b[0] *= factor; |
| iir->biquads[current_biquad].b[1] *= factor; |
| iir->biquads[current_biquad].b[2] *= factor; |
| } |
| |
| iir->biquads[current_biquad].b[0] *= (current_biquad ? 1.0 : iir->g); |
| iir->biquads[current_biquad].b[1] *= (current_biquad ? 1.0 : iir->g); |
| iir->biquads[current_biquad].b[2] *= (current_biquad ? 1.0 : iir->g); |
| |
| av_log(ctx, AV_LOG_VERBOSE, "a=%f %f %f:b=%f %f %f\n", |
| iir->biquads[current_biquad].a[0], |
| iir->biquads[current_biquad].a[1], |
| iir->biquads[current_biquad].a[2], |
| iir->biquads[current_biquad].b[0], |
| iir->biquads[current_biquad].b[1], |
| iir->biquads[current_biquad].b[2]); |
| |
| current_biquad++; |
| } |
| } |
| |
| return 0; |
| } |
| |
| static void biquad_process(double *x, double *y, int length, |
| double b0, double b1, double b2, |
| double a1, double a2) |
| { |
| double w1 = 0., w2 = 0.; |
| |
| a1 = -a1; |
| a2 = -a2; |
| |
| for (int n = 0; n < length; n++) { |
| double out, in = x[n]; |
| |
| y[n] = out = in * b0 + w1; |
| w1 = b1 * in + w2 + a1 * out; |
| w2 = b2 * in + a2 * out; |
| } |
| } |
| |
| static void solve(double *matrix, double *vector, int n, double *y, double *x, double *lu) |
| { |
| double sum = 0.; |
| |
| for (int i = 0; i < n; i++) { |
| for (int j = i; j < n; j++) { |
| sum = 0.; |
| for (int k = 0; k < i; k++) |
| sum += lu[i * n + k] * lu[k * n + j]; |
| lu[i * n + j] = matrix[j * n + i] - sum; |
| } |
| for (int j = i + 1; j < n; j++) { |
| sum = 0.; |
| for (int k = 0; k < i; k++) |
| sum += lu[j * n + k] * lu[k * n + i]; |
| lu[j * n + i] = (1. / lu[i * n + i]) * (matrix[i * n + j] - sum); |
| } |
| } |
| |
| for (int i = 0; i < n; i++) { |
| sum = 0.; |
| for (int k = 0; k < i; k++) |
| sum += lu[i * n + k] * y[k]; |
| y[i] = vector[i] - sum; |
| } |
| |
| for (int i = n - 1; i >= 0; i--) { |
| sum = 0.; |
| for (int k = i + 1; k < n; k++) |
| sum += lu[i * n + k] * x[k]; |
| x[i] = (1 / lu[i * n + i]) * (y[i] - sum); |
| } |
| } |
| |
| static int convert_serial2parallel(AVFilterContext *ctx, int channels) |
| { |
| AudioIIRContext *s = ctx->priv; |
| int ret = 0; |
| |
| for (int ch = 0; ch < channels; ch++) { |
| IIRChannel *iir = &s->iir[ch]; |
| int nb_biquads = (FFMAX(iir->nb_ab[0], iir->nb_ab[1]) + 1) / 2; |
| int length = nb_biquads * 2 + 1; |
| double *impulse = av_calloc(length, sizeof(*impulse)); |
| double *y = av_calloc(length, sizeof(*y)); |
| double *resp = av_calloc(length, sizeof(*resp)); |
| double *M = av_calloc((length - 1) * 2 * nb_biquads, sizeof(*M)); |
| double *W = av_calloc((length - 1) * 2 * nb_biquads, sizeof(*W)); |
| |
| if (!impulse || !y || !resp || !M) { |
| av_free(impulse); |
| av_free(y); |
| av_free(resp); |
| av_free(M); |
| av_free(W); |
| return AVERROR(ENOMEM); |
| } |
| |
| impulse[0] = 1.; |
| |
| for (int n = 0; n < nb_biquads; n++) { |
| BiquadContext *biquad = &iir->biquads[n]; |
| |
| biquad_process(n ? y : impulse, y, length, |
| biquad->b[0], biquad->b[1], biquad->b[2], |
| biquad->a[1], biquad->a[2]); |
| } |
| |
| for (int n = 0; n < nb_biquads; n++) { |
| BiquadContext *biquad = &iir->biquads[n]; |
| |
| biquad_process(impulse, resp, length - 1, |
| 1., 0., 0., biquad->a[1], biquad->a[2]); |
| |
| memcpy(M + n * 2 * (length - 1), resp, sizeof(*resp) * (length - 1)); |
| memcpy(M + n * 2 * (length - 1) + length, resp, sizeof(*resp) * (length - 2)); |
| memset(resp, 0, length * sizeof(*resp)); |
| } |
| |
| solve(M, &y[1], length - 1, &impulse[1], resp, W); |
| |
| iir->fir = y[0]; |
| |
| for (int n = 0; n < nb_biquads; n++) { |
| BiquadContext *biquad = &iir->biquads[n]; |
| |
| biquad->b[0] = 0.; |
| biquad->b[1] = resp[n * 2 + 0]; |
| biquad->b[2] = resp[n * 2 + 1]; |
| } |
| |
| av_free(impulse); |
| av_free(y); |
| av_free(resp); |
| av_free(M); |
| av_free(W); |
| |
| if (ret < 0) |
| return ret; |
| } |
| |
| return 0; |
| } |
| |
| static void convert_pr2zp(AVFilterContext *ctx, int channels) |
| { |
| AudioIIRContext *s = ctx->priv; |
| int ch; |
| |
| for (ch = 0; ch < channels; ch++) { |
| IIRChannel *iir = &s->iir[ch]; |
| int n; |
| |
| for (n = 0; n < iir->nb_ab[0]; n++) { |
| double r = iir->ab[0][2*n]; |
| double angle = iir->ab[0][2*n+1]; |
| |
| iir->ab[0][2*n] = r * cos(angle); |
| iir->ab[0][2*n+1] = r * sin(angle); |
| } |
| |
| for (n = 0; n < iir->nb_ab[1]; n++) { |
| double r = iir->ab[1][2*n]; |
| double angle = iir->ab[1][2*n+1]; |
| |
| iir->ab[1][2*n] = r * cos(angle); |
| iir->ab[1][2*n+1] = r * sin(angle); |
| } |
| } |
| } |
| |
| static void convert_sp2zp(AVFilterContext *ctx, int channels) |
| { |
| AudioIIRContext *s = ctx->priv; |
| int ch; |
| |
| for (ch = 0; ch < channels; ch++) { |
| IIRChannel *iir = &s->iir[ch]; |
| int n; |
| |
| for (n = 0; n < iir->nb_ab[0]; n++) { |
| double sr = iir->ab[0][2*n]; |
| double si = iir->ab[0][2*n+1]; |
| |
| iir->ab[0][2*n] = exp(sr) * cos(si); |
| iir->ab[0][2*n+1] = exp(sr) * sin(si); |
| } |
| |
| for (n = 0; n < iir->nb_ab[1]; n++) { |
| double sr = iir->ab[1][2*n]; |
| double si = iir->ab[1][2*n+1]; |
| |
| iir->ab[1][2*n] = exp(sr) * cos(si); |
| iir->ab[1][2*n+1] = exp(sr) * sin(si); |
| } |
| } |
| } |
| |
| static double fact(double i) |
| { |
| if (i <= 0.) |
| return 1.; |
| return i * fact(i - 1.); |
| } |
| |
| static double coef_sf2zf(double *a, int N, int n) |
| { |
| double z = 0.; |
| |
| for (int i = 0; i <= N; i++) { |
| double acc = 0.; |
| |
| for (int k = FFMAX(n - N + i, 0); k <= FFMIN(i, n); k++) { |
| acc += ((fact(i) * fact(N - i)) / |
| (fact(k) * fact(i - k) * fact(n - k) * fact(N - i - n + k))) * |
| ((k & 1) ? -1. : 1.);; |
| } |
| |
| z += a[i] * pow(2., i) * acc; |
| } |
| |
| return z; |
| } |
| |
| static void convert_sf2tf(AVFilterContext *ctx, int channels) |
| { |
| AudioIIRContext *s = ctx->priv; |
| int ch; |
| |
| for (ch = 0; ch < channels; ch++) { |
| IIRChannel *iir = &s->iir[ch]; |
| double *temp0 = av_calloc(iir->nb_ab[0], sizeof(*temp0)); |
| double *temp1 = av_calloc(iir->nb_ab[1], sizeof(*temp1)); |
| |
| if (!temp0 || !temp1) |
| goto next; |
| |
| memcpy(temp0, iir->ab[0], iir->nb_ab[0] * sizeof(*temp0)); |
| memcpy(temp1, iir->ab[1], iir->nb_ab[1] * sizeof(*temp1)); |
| |
| for (int n = 0; n < iir->nb_ab[0]; n++) |
| iir->ab[0][n] = coef_sf2zf(temp0, iir->nb_ab[0] - 1, n); |
| |
| for (int n = 0; n < iir->nb_ab[1]; n++) |
| iir->ab[1][n] = coef_sf2zf(temp1, iir->nb_ab[1] - 1, n); |
| |
| next: |
| av_free(temp0); |
| av_free(temp1); |
| } |
| } |
| |
| static void convert_pd2zp(AVFilterContext *ctx, int channels) |
| { |
| AudioIIRContext *s = ctx->priv; |
| int ch; |
| |
| for (ch = 0; ch < channels; ch++) { |
| IIRChannel *iir = &s->iir[ch]; |
| int n; |
| |
| for (n = 0; n < iir->nb_ab[0]; n++) { |
| double r = iir->ab[0][2*n]; |
| double angle = M_PI*iir->ab[0][2*n+1]/180.; |
| |
| iir->ab[0][2*n] = r * cos(angle); |
| iir->ab[0][2*n+1] = r * sin(angle); |
| } |
| |
| for (n = 0; n < iir->nb_ab[1]; n++) { |
| double r = iir->ab[1][2*n]; |
| double angle = M_PI*iir->ab[1][2*n+1]/180.; |
| |
| iir->ab[1][2*n] = r * cos(angle); |
| iir->ab[1][2*n+1] = r * sin(angle); |
| } |
| } |
| } |
| |
| static void check_stability(AVFilterContext *ctx, int channels) |
| { |
| AudioIIRContext *s = ctx->priv; |
| int ch; |
| |
| for (ch = 0; ch < channels; ch++) { |
| IIRChannel *iir = &s->iir[ch]; |
| |
| for (int n = 0; n < iir->nb_ab[0]; n++) { |
| double pr = hypot(iir->ab[0][2*n], iir->ab[0][2*n+1]); |
| |
| if (pr >= 1.) { |
| av_log(ctx, AV_LOG_WARNING, "pole %d at channel %d is unstable\n", n, ch); |
| break; |
| } |
| } |
| } |
| } |
| |
| static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color) |
| { |
| const uint8_t *font; |
| int font_height; |
| int i; |
| |
| font = avpriv_cga_font, font_height = 8; |
| |
| for (i = 0; txt[i]; i++) { |
| int char_y, mask; |
| |
| uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4; |
| for (char_y = 0; char_y < font_height; char_y++) { |
| for (mask = 0x80; mask; mask >>= 1) { |
| if (font[txt[i] * font_height + char_y] & mask) |
| AV_WL32(p, color); |
| p += 4; |
| } |
| p += pic->linesize[0] - 8 * 4; |
| } |
| } |
| } |
| |
| static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color) |
| { |
| int dx = FFABS(x1-x0); |
| int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1; |
| int err = (dx>dy ? dx : -dy) / 2, e2; |
| |
| for (;;) { |
| AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color); |
| |
| if (x0 == x1 && y0 == y1) |
| break; |
| |
| e2 = err; |
| |
| if (e2 >-dx) { |
| err -= dy; |
| x0--; |
| } |
| |
| if (e2 < dy) { |
| err += dx; |
| y0 += sy; |
| } |
| } |
| } |
| |
| static double distance(double x0, double x1, double y0, double y1) |
| { |
| return hypot(x0 - x1, y0 - y1); |
| } |
| |
| static void get_response(int channel, int format, double w, |
| const double *b, const double *a, |
| int nb_b, int nb_a, double *magnitude, double *phase) |
| { |
| double realz, realp; |
| double imagz, imagp; |
| double real, imag; |
| double div; |
| |
| if (format == 0) { |
| realz = 0., realp = 0.; |
| imagz = 0., imagp = 0.; |
| for (int x = 0; x < nb_a; x++) { |
| realz += cos(-x * w) * a[x]; |
| imagz += sin(-x * w) * a[x]; |
| } |
| |
| for (int x = 0; x < nb_b; x++) { |
| realp += cos(-x * w) * b[x]; |
| imagp += sin(-x * w) * b[x]; |
| } |
| |
| div = realp * realp + imagp * imagp; |
| real = (realz * realp + imagz * imagp) / div; |
| imag = (imagz * realp - imagp * realz) / div; |
| |
| *magnitude = hypot(real, imag); |
| *phase = atan2(imag, real); |
| } else { |
| double p = 1., z = 1.; |
| double acc = 0.; |
| |
| for (int x = 0; x < nb_a; x++) { |
| z *= distance(cos(w), a[2 * x], sin(w), a[2 * x + 1]); |
| acc += atan2(sin(w) - a[2 * x + 1], cos(w) - a[2 * x]); |
| } |
| |
| for (int x = 0; x < nb_b; x++) { |
| p *= distance(cos(w), b[2 * x], sin(w), b[2 * x + 1]); |
| acc -= atan2(sin(w) - b[2 * x + 1], cos(w) - b[2 * x]); |
| } |
| |
| *magnitude = z / p; |
| *phase = acc; |
| } |
| } |
| |
| static void draw_response(AVFilterContext *ctx, AVFrame *out, int sample_rate) |
| { |
| AudioIIRContext *s = ctx->priv; |
| double *mag, *phase, *temp, *delay, min = DBL_MAX, max = -DBL_MAX; |
| double min_delay = DBL_MAX, max_delay = -DBL_MAX, min_phase, max_phase; |
| int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1; |
| char text[32]; |
| int ch, i; |
| |
| memset(out->data[0], 0, s->h * out->linesize[0]); |
| |
| phase = av_malloc_array(s->w, sizeof(*phase)); |
| temp = av_malloc_array(s->w, sizeof(*temp)); |
| mag = av_malloc_array(s->w, sizeof(*mag)); |
| delay = av_malloc_array(s->w, sizeof(*delay)); |
| if (!mag || !phase || !delay || !temp) |
| goto end; |
| |
| ch = av_clip(s->ir_channel, 0, s->channels - 1); |
| for (i = 0; i < s->w; i++) { |
| const double *b = s->iir[ch].ab[0]; |
| const double *a = s->iir[ch].ab[1]; |
| const int nb_b = s->iir[ch].nb_ab[0]; |
| const int nb_a = s->iir[ch].nb_ab[1]; |
| double w = i * M_PI / (s->w - 1); |
| double m, p; |
| |
| get_response(ch, s->format, w, b, a, nb_b, nb_a, &m, &p); |
| |
| mag[i] = s->iir[ch].g * m; |
| phase[i] = p; |
| min = fmin(min, mag[i]); |
| max = fmax(max, mag[i]); |
| } |
| |
| temp[0] = 0.; |
| for (i = 0; i < s->w - 1; i++) { |
| double d = phase[i] - phase[i + 1]; |
| temp[i + 1] = ceil(fabs(d) / (2. * M_PI)) * 2. * M_PI * ((d > M_PI) - (d < -M_PI)); |
| } |
| |
| min_phase = phase[0]; |
| max_phase = phase[0]; |
| for (i = 1; i < s->w; i++) { |
| temp[i] += temp[i - 1]; |
| phase[i] += temp[i]; |
| min_phase = fmin(min_phase, phase[i]); |
| max_phase = fmax(max_phase, phase[i]); |
| } |
| |
| for (i = 0; i < s->w - 1; i++) { |
| double div = s->w / (double)sample_rate; |
| |
| delay[i + 1] = -(phase[i] - phase[i + 1]) / div; |
| min_delay = fmin(min_delay, delay[i + 1]); |
| max_delay = fmax(max_delay, delay[i + 1]); |
| } |
| delay[0] = delay[1]; |
| |
| for (i = 0; i < s->w; i++) { |
| int ymag = mag[i] / max * (s->h - 1); |
| int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1); |
| int yphase = (phase[i] - min_phase) / (max_phase - min_phase) * (s->h - 1); |
| |
| ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1); |
| yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1); |
| ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1); |
| |
| if (prev_ymag < 0) |
| prev_ymag = ymag; |
| if (prev_yphase < 0) |
| prev_yphase = yphase; |
| if (prev_ydelay < 0) |
| prev_ydelay = ydelay; |
| |
| draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF); |
| draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00); |
| draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF); |
| |
| prev_ymag = ymag; |
| prev_yphase = yphase; |
| prev_ydelay = ydelay; |
| } |
| |
| if (s->w > 400 && s->h > 100) { |
| drawtext(out, 2, 2, "Max Magnitude:", 0xDDDDDDDD); |
| snprintf(text, sizeof(text), "%.2f", max); |
| drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD); |
| |
| drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD); |
| snprintf(text, sizeof(text), "%.2f", min); |
| drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD); |
| |
| drawtext(out, 2, 22, "Max Phase:", 0xDDDDDDDD); |
| snprintf(text, sizeof(text), "%.2f", max_phase); |
| drawtext(out, 15 * 8 + 2, 22, text, 0xDDDDDDDD); |
| |
| drawtext(out, 2, 32, "Min Phase:", 0xDDDDDDDD); |
| snprintf(text, sizeof(text), "%.2f", min_phase); |
| drawtext(out, 15 * 8 + 2, 32, text, 0xDDDDDDDD); |
| |
| drawtext(out, 2, 42, "Max Delay:", 0xDDDDDDDD); |
| snprintf(text, sizeof(text), "%.2f", max_delay); |
| drawtext(out, 11 * 8 + 2, 42, text, 0xDDDDDDDD); |
| |
| drawtext(out, 2, 52, "Min Delay:", 0xDDDDDDDD); |
| snprintf(text, sizeof(text), "%.2f", min_delay); |
| drawtext(out, 11 * 8 + 2, 52, text, 0xDDDDDDDD); |
| } |
| |
| end: |
| av_free(delay); |
| av_free(temp); |
| av_free(phase); |
| av_free(mag); |
| } |
| |
| static int config_output(AVFilterLink *outlink) |
| { |
| AVFilterContext *ctx = outlink->src; |
| AudioIIRContext *s = ctx->priv; |
| AVFilterLink *inlink = ctx->inputs[0]; |
| int ch, ret, i; |
| |
| s->channels = inlink->channels; |
| s->iir = av_calloc(s->channels, sizeof(*s->iir)); |
| if (!s->iir) |
| return AVERROR(ENOMEM); |
| |
| ret = read_gains(ctx, s->g_str, inlink->channels); |
| if (ret < 0) |
| return ret; |
| |
| ret = read_channels(ctx, inlink->channels, s->a_str, 0); |
| if (ret < 0) |
| return ret; |
| |
| ret = read_channels(ctx, inlink->channels, s->b_str, 1); |
| if (ret < 0) |
| return ret; |
| |
| if (s->format == -1) { |
| convert_sf2tf(ctx, inlink->channels); |
| s->format = 0; |
| } else if (s->format == 2) { |
| convert_pr2zp(ctx, inlink->channels); |
| } else if (s->format == 3) { |
| convert_pd2zp(ctx, inlink->channels); |
| } else if (s->format == 4) { |
| convert_sp2zp(ctx, inlink->channels); |
| } |
| if (s->format > 0) { |
| check_stability(ctx, inlink->channels); |
| } |
| |
| av_frame_free(&s->video); |
| if (s->response) { |
| s->video = ff_get_video_buffer(ctx->outputs[1], s->w, s->h); |
| if (!s->video) |
| return AVERROR(ENOMEM); |
| |
| draw_response(ctx, s->video, inlink->sample_rate); |
| } |
| |
| if (s->format == 0) |
| av_log(ctx, AV_LOG_WARNING, "transfer function coefficients format is not recommended for too high number of zeros/poles.\n"); |
| |
| if (s->format > 0 && s->process == 0) { |
| av_log(ctx, AV_LOG_WARNING, "Direct processsing is not recommended for zp coefficients format.\n"); |
| |
| ret = convert_zp2tf(ctx, inlink->channels); |
| if (ret < 0) |
| return ret; |
| } else if (s->format == -2 && s->process > 0) { |
| av_log(ctx, AV_LOG_ERROR, "Only direct processing is implemented for lattice-ladder function.\n"); |
| return AVERROR_PATCHWELCOME; |
| } else if (s->format <= 0 && s->process == 1) { |
| av_log(ctx, AV_LOG_ERROR, "Serial processing is not implemented for transfer function.\n"); |
| return AVERROR_PATCHWELCOME; |
| } else if (s->format <= 0 && s->process == 2) { |
| av_log(ctx, AV_LOG_ERROR, "Parallel processing is not implemented for transfer function.\n"); |
| return AVERROR_PATCHWELCOME; |
| } else if (s->format > 0 && s->process == 1) { |
| ret = decompose_zp2biquads(ctx, inlink->channels); |
| if (ret < 0) |
| return ret; |
| } else if (s->format > 0 && s->process == 2) { |
| if (s->precision > 1) |
| av_log(ctx, AV_LOG_WARNING, "Parallel processing is not recommended for fixed-point precisions.\n"); |
| ret = decompose_zp2biquads(ctx, inlink->channels); |
| if (ret < 0) |
| return ret; |
| ret = convert_serial2parallel(ctx, inlink->channels); |
| if (ret < 0) |
| return ret; |
| } |
| |
| for (ch = 0; s->format == -2 && ch < inlink->channels; ch++) { |
| IIRChannel *iir = &s->iir[ch]; |
| |
| if (iir->nb_ab[0] != iir->nb_ab[1] + 1) { |
| av_log(ctx, AV_LOG_ERROR, "Number of ladder coefficients must be one more than number of reflection coefficients.\n"); |
| return AVERROR(EINVAL); |
| } |
| } |
| |
| for (ch = 0; s->format == 0 && ch < inlink->channels; ch++) { |
| IIRChannel *iir = &s->iir[ch]; |
| |
| for (i = 1; i < iir->nb_ab[0]; i++) { |
| iir->ab[0][i] /= iir->ab[0][0]; |
| } |
| |
| iir->ab[0][0] = 1.0; |
| for (i = 0; i < iir->nb_ab[1]; i++) { |
| iir->ab[1][i] *= iir->g; |
| } |
| |
| normalize_coeffs(ctx, ch); |
| } |
| |
| switch (inlink->format) { |
| case AV_SAMPLE_FMT_DBLP: s->iir_channel = s->process == 2 ? iir_ch_parallel_dblp : s->process == 1 ? iir_ch_serial_dblp : iir_ch_dblp; break; |
| case AV_SAMPLE_FMT_FLTP: s->iir_channel = s->process == 2 ? iir_ch_parallel_fltp : s->process == 1 ? iir_ch_serial_fltp : iir_ch_fltp; break; |
| case AV_SAMPLE_FMT_S32P: s->iir_channel = s->process == 2 ? iir_ch_parallel_s32p : s->process == 1 ? iir_ch_serial_s32p : iir_ch_s32p; break; |
| case AV_SAMPLE_FMT_S16P: s->iir_channel = s->process == 2 ? iir_ch_parallel_s16p : s->process == 1 ? iir_ch_serial_s16p : iir_ch_s16p; break; |
| } |
| |
| if (s->format == -2) { |
| switch (inlink->format) { |
| case AV_SAMPLE_FMT_DBLP: s->iir_channel = iir_ch_lattice_dblp; break; |
| case AV_SAMPLE_FMT_FLTP: s->iir_channel = iir_ch_lattice_fltp; break; |
| case AV_SAMPLE_FMT_S32P: s->iir_channel = iir_ch_lattice_s32p; break; |
| case AV_SAMPLE_FMT_S16P: s->iir_channel = iir_ch_lattice_s16p; break; |
| } |
| } |
| |
| return 0; |
| } |
| |
| static int filter_frame(AVFilterLink *inlink, AVFrame *in) |
| { |
| AVFilterContext *ctx = inlink->dst; |
| AudioIIRContext *s = ctx->priv; |
| AVFilterLink *outlink = ctx->outputs[0]; |
| ThreadData td; |
| AVFrame *out; |
| int ch, ret; |
| |
| if (av_frame_is_writable(in) && s->process != 2) { |
| 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); |
| } |
| |
| td.in = in; |
| td.out = out; |
| ctx->internal->execute(ctx, s->iir_channel, &td, NULL, outlink->channels); |
| |
| for (ch = 0; ch < outlink->channels; ch++) { |
| if (s->iir[ch].clippings > 0) |
| av_log(ctx, AV_LOG_WARNING, "Channel %d clipping %d times. Please reduce gain.\n", |
| ch, s->iir[ch].clippings); |
| s->iir[ch].clippings = 0; |
| } |
| |
| if (in != out) |
| av_frame_free(&in); |
| |
| if (s->response) { |
| AVFilterLink *outlink = ctx->outputs[1]; |
| int64_t old_pts = s->video->pts; |
| int64_t new_pts = av_rescale_q(out->pts, ctx->inputs[0]->time_base, outlink->time_base); |
| |
| if (new_pts > old_pts) { |
| AVFrame *clone; |
| |
| s->video->pts = new_pts; |
| clone = av_frame_clone(s->video); |
| if (!clone) |
| return AVERROR(ENOMEM); |
| ret = ff_filter_frame(outlink, clone); |
| if (ret < 0) |
| return ret; |
| } |
| } |
| |
| return ff_filter_frame(outlink, out); |
| } |
| |
| static int config_video(AVFilterLink *outlink) |
| { |
| AVFilterContext *ctx = outlink->src; |
| AudioIIRContext *s = ctx->priv; |
| |
| outlink->sample_aspect_ratio = (AVRational){1,1}; |
| outlink->w = s->w; |
| outlink->h = s->h; |
| outlink->frame_rate = s->rate; |
| outlink->time_base = av_inv_q(outlink->frame_rate); |
| |
| return 0; |
| } |
| |
| static av_cold int init(AVFilterContext *ctx) |
| { |
| AudioIIRContext *s = ctx->priv; |
| AVFilterPad pad, vpad; |
| int ret; |
| |
| if (!s->a_str || !s->b_str || !s->g_str) { |
| av_log(ctx, AV_LOG_ERROR, "Valid coefficients are mandatory.\n"); |
| return AVERROR(EINVAL); |
| } |
| |
| switch (s->precision) { |
| case 0: s->sample_format = AV_SAMPLE_FMT_DBLP; break; |
| case 1: s->sample_format = AV_SAMPLE_FMT_FLTP; break; |
| case 2: s->sample_format = AV_SAMPLE_FMT_S32P; break; |
| case 3: s->sample_format = AV_SAMPLE_FMT_S16P; break; |
| default: return AVERROR_BUG; |
| } |
| |
| pad = (AVFilterPad){ |
| .name = "default", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .config_props = config_output, |
| }; |
| |
| ret = ff_insert_outpad(ctx, 0, &pad); |
| if (ret < 0) |
| return ret; |
| |
| if (s->response) { |
| vpad = (AVFilterPad){ |
| .name = "filter_response", |
| .type = AVMEDIA_TYPE_VIDEO, |
| .config_props = config_video, |
| }; |
| |
| ret = ff_insert_outpad(ctx, 1, &vpad); |
| if (ret < 0) |
| return ret; |
| } |
| |
| return 0; |
| } |
| |
| static av_cold void uninit(AVFilterContext *ctx) |
| { |
| AudioIIRContext *s = ctx->priv; |
| int ch; |
| |
| if (s->iir) { |
| for (ch = 0; ch < s->channels; ch++) { |
| IIRChannel *iir = &s->iir[ch]; |
| av_freep(&iir->ab[0]); |
| av_freep(&iir->ab[1]); |
| av_freep(&iir->cache[0]); |
| av_freep(&iir->cache[1]); |
| av_freep(&iir->biquads); |
| } |
| } |
| av_freep(&s->iir); |
| |
| av_frame_free(&s->video); |
| } |
| |
| static const AVFilterPad inputs[] = { |
| { |
| .name = "default", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .filter_frame = filter_frame, |
| }, |
| { NULL } |
| }; |
| |
| #define OFFSET(x) offsetof(AudioIIRContext, x) |
| #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
| #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
| |
| static const AVOption aiir_options[] = { |
| { "zeros", "set B/numerator/zeros/reflection coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF }, |
| { "z", "set B/numerator/zeros/reflection coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF }, |
| { "poles", "set A/denominator/poles/ladder coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF }, |
| { "p", "set A/denominator/poles/ladder coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF }, |
| { "gains", "set channels gains", OFFSET(g_str), AV_OPT_TYPE_STRING, {.str="1|1"}, 0, 0, AF }, |
| { "k", "set channels gains", OFFSET(g_str), AV_OPT_TYPE_STRING, {.str="1|1"}, 0, 0, AF }, |
| { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF }, |
| { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF }, |
| { "format", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, -2, 4, AF, "format" }, |
| { "f", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, -2, 4, AF, "format" }, |
| { "ll", "lattice-ladder function", 0, AV_OPT_TYPE_CONST, {.i64=-2}, 0, 0, AF, "format" }, |
| { "sf", "analog transfer function", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "format" }, |
| { "tf", "digital transfer function", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "format" }, |
| { "zp", "Z-plane zeros/poles", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "format" }, |
| { "pr", "Z-plane zeros/poles (polar radians)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "format" }, |
| { "pd", "Z-plane zeros/poles (polar degrees)", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "format" }, |
| { "sp", "S-plane zeros/poles", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF, "format" }, |
| { "process", "set kind of processing", OFFSET(process), AV_OPT_TYPE_INT, {.i64=1}, 0, 2, AF, "process" }, |
| { "r", "set kind of processing", OFFSET(process), AV_OPT_TYPE_INT, {.i64=1}, 0, 2, AF, "process" }, |
| { "d", "direct", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "process" }, |
| { "s", "serial", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "process" }, |
| { "p", "parallel", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "process" }, |
| { "precision", "set filtering precision", OFFSET(precision),AV_OPT_TYPE_INT, {.i64=0}, 0, 3, AF, "precision" }, |
| { "e", "set precision", OFFSET(precision),AV_OPT_TYPE_INT, {.i64=0}, 0, 3, AF, "precision" }, |
| { "dbl", "double-precision floating-point", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "precision" }, |
| { "flt", "single-precision floating-point", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision" }, |
| { "i32", "32-bit integers", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision" }, |
| { "i16", "16-bit integers", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision" }, |
| { "normalize", "normalize coefficients", OFFSET(normalize),AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF }, |
| { "n", "normalize coefficients", OFFSET(normalize),AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF }, |
| { "mix", "set mix", OFFSET(mix), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF }, |
| { "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF }, |
| { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF }, |
| { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF }, |
| { "rate", "set video rate", OFFSET(rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF }, |
| { NULL }, |
| }; |
| |
| AVFILTER_DEFINE_CLASS(aiir); |
| |
| AVFilter ff_af_aiir = { |
| .name = "aiir", |
| .description = NULL_IF_CONFIG_SMALL("Apply Infinite Impulse Response filter with supplied coefficients."), |
| .priv_size = sizeof(AudioIIRContext), |
| .priv_class = &aiir_class, |
| .init = init, |
| .uninit = uninit, |
| .query_formats = query_formats, |
| .inputs = inputs, |
| .flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS | |
| AVFILTER_FLAG_SLICE_THREADS, |
| }; |
