| /* |
| * 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/opt.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 a0, a1, a2; |
| double b0, b1, b2; |
| double i1, i2; |
| double o1, o2; |
| } BiquadContext; |
| |
| typedef struct IIRChannel { |
| int nb_ab[2]; |
| double *ab[2]; |
| double g; |
| double *cache[2]; |
| BiquadContext *biquads; |
| int clippings; |
| } IIRChannel; |
| |
| typedef struct AudioIIRContext { |
| const AVClass *class; |
| char *a_str, *b_str, *g_str; |
| double dry_gain, wet_gain; |
| int format; |
| int process; |
| int precision; |
| |
| 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 |
| }; |
| 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; |
| |
| 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; \ |
| ThreadData *td = arg; \ |
| AVFrame *in = td->in, *out = td->out; \ |
| const type *src = (const type *)in->extended_data[ch]; \ |
| double *ic = (double *)s->iir[ch].cache[0]; \ |
| double *oc = (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]; \ |
| 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; \ |
| 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; \ |
| 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]; \ |
| 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].a1; \ |
| const double a2 = -iir->biquads[i].a2; \ |
| const double b0 = iir->biquads[i].b0; \ |
| const double b1 = iir->biquads[i].b1; \ |
| const double b2 = iir->biquads[i].b2; \ |
| double i1 = iir->biquads[i].i1; \ |
| double i2 = iir->biquads[i].i2; \ |
| double o1 = iir->biquads[i].o1; \ |
| double o2 = iir->biquads[i].o2; \ |
| \ |
| for (n = 0; n < in->nb_samples; n++) { \ |
| double sample = ig * (i ? dst[n] : src[n]); \ |
| double o0 = sample * b0 + i1 * b1 + i2 * b2 + o1 * a1 + o2 * a2; \ |
| \ |
| i2 = i1; \ |
| i1 = src[n]; \ |
| o2 = o1; \ |
| o1 = o0; \ |
| o0 *= og; \ |
| \ |
| 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].i1 = i1; \ |
| iir->biquads[i].i2 = i2; \ |
| iir->biquads[i].o1 = o1; \ |
| iir->biquads[i].o2 = o2; \ |
| } \ |
| \ |
| 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) |
| |
| 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 (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 (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 (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" }; |
| |
| 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) { |
| 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 multiply(double wre, double wim, int npz, double *coeffs) |
| { |
| double nwre = -wre, nwim = -wim; |
| double cre, cim; |
| int i; |
| |
| for (i = npz; i >= 1; i--) { |
| cre = coeffs[2 * i + 0]; |
| cim = coeffs[2 * i + 1]; |
| |
| coeffs[2 * i + 0] = (nwre * cre - nwim * cim) + coeffs[2 * (i - 1) + 0]; |
| coeffs[2 * i + 1] = (nwre * cim + nwim * cre) + coeffs[2 * (i - 1) + 1]; |
| } |
| |
| cre = coeffs[0]; |
| cim = coeffs[1]; |
| coeffs[0] = nwre * cre - nwim * cim; |
| coeffs[1] = nwre * cim + nwim * cre; |
| } |
| |
| static int expand(AVFilterContext *ctx, double *pz, int nb, double *coeffs) |
| { |
| int i; |
| |
| coeffs[0] = 1.0; |
| coeffs[1] = 0.0; |
| |
| for (i = 0; i < nb; i++) { |
| coeffs[2 * (i + 1) ] = 0.0; |
| coeffs[2 * (i + 1) + 1] = 0.0; |
| } |
| |
| for (i = 0; i < nb; i++) |
| multiply(pz[2 * i], pz[2 * i + 1], nb, coeffs); |
| |
| for (i = 0; i < nb + 1; i++) { |
| if (fabs(coeffs[2 * i + 1]) > FLT_EPSILON) { |
| av_log(ctx, AV_LOG_ERROR, "coeff: %lf of z^%d is not real; poles/zeros are not complex conjugates.\n", |
| coeffs[2 * i + 1], i); |
| return AVERROR(EINVAL); |
| } |
| } |
| |
| return 0; |
| } |
| |
| 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[0] + 1) * 2, sizeof(*topc)); |
| botc = av_calloc((iir->nb_ab[1] + 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]++; |
| |
| 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; |
| 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[1]; 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].a0 = 1.0; |
| iir->biquads[current_biquad].a1 = a[2] / a[4]; |
| iir->biquads[current_biquad].a2 = a[0] / a[4]; |
| iir->biquads[current_biquad].b0 = b[4] / a[4] * (current_biquad ? 1.0 : iir->g); |
| iir->biquads[current_biquad].b1 = b[2] / a[4] * (current_biquad ? 1.0 : iir->g); |
| iir->biquads[current_biquad].b2 = b[0] / a[4] * (current_biquad ? 1.0 : iir->g); |
| |
| av_log(ctx, AV_LOG_VERBOSE, "a=%lf %lf %lf:b=%lf %lf %lf\n", |
| iir->biquads[current_biquad].a0, |
| iir->biquads[current_biquad].a1, |
| iir->biquads[current_biquad].a2, |
| iir->biquads[current_biquad].b0, |
| iir->biquads[current_biquad].b1, |
| iir->biquads[current_biquad].b2); |
| |
| current_biquad++; |
| } |
| } |
| |
| 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_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 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 == 2) { |
| convert_pr2zp(ctx, inlink->channels); |
| } else if (s->format == 3) { |
| convert_pd2zp(ctx, inlink->channels); |
| } |
| |
| if (s->format == 0) |
| av_log(ctx, AV_LOG_WARNING, "tf 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 == 0 && s->process == 1) { |
| av_log(ctx, AV_LOG_ERROR, "Serial cascading is not implemented for transfer function.\n"); |
| return AVERROR_PATCHWELCOME; |
| } else if (s->format > 0 && s->process == 1) { |
| if (inlink->format == AV_SAMPLE_FMT_S16P) |
| av_log(ctx, AV_LOG_WARNING, "Serial cascading is not recommended for i16 precision.\n"); |
| |
| ret = decompose_zp2biquads(ctx, inlink->channels); |
| if (ret < 0) |
| return ret; |
| } |
| |
| for (ch = 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]; |
| } |
| |
| for (i = 0; i < iir->nb_ab[1]; i++) { |
| iir->ab[1][i] *= iir->g / iir->ab[0][0]; |
| } |
| } |
| |
| switch (inlink->format) { |
| case AV_SAMPLE_FMT_DBLP: s->iir_channel = s->process == 1 ? iir_ch_serial_dblp : iir_ch_dblp; break; |
| case AV_SAMPLE_FMT_FLTP: s->iir_channel = s->process == 1 ? iir_ch_serial_fltp : iir_ch_fltp; break; |
| case AV_SAMPLE_FMT_S32P: s->iir_channel = s->process == 1 ? iir_ch_serial_s32p : iir_ch_s32p; break; |
| case AV_SAMPLE_FMT_S16P: s->iir_channel = s->process == 1 ? iir_ch_serial_s16p : iir_ch_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; |
| |
| 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); |
| } |
| |
| 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); |
| |
| return ff_filter_frame(outlink, out); |
| } |
| |
| static av_cold int init(AVFilterContext *ctx) |
| { |
| AudioIIRContext *s = ctx->priv; |
| |
| 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; |
| } |
| |
| 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); |
| } |
| |
| static const AVFilterPad inputs[] = { |
| { |
| .name = "default", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .filter_frame = filter_frame, |
| }, |
| { NULL } |
| }; |
| |
| static const AVFilterPad outputs[] = { |
| { |
| .name = "default", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .config_props = config_output, |
| }, |
| { NULL } |
| }; |
| |
| #define OFFSET(x) offsetof(AudioIIRContext, x) |
| #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
| |
| static const AVOption aiir_options[] = { |
| { "z", "set B/numerator/zeros coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 0, 0, AF }, |
| { "p", "set A/denominator/poles coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1+0i 1-0i"}, 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 }, |
| { "f", "set coefficients format", OFFSET(format), AV_OPT_TYPE_INT, {.i64=1}, 0, 3, AF, "format" }, |
| { "tf", "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" }, |
| { "r", "set kind of processing", OFFSET(process), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "process" }, |
| { "d", "direct", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "process" }, |
| { "s", "serial cascading", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "process" }, |
| { "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" }, |
| { 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, |
| .outputs = outputs, |
| .flags = AVFILTER_FLAG_SLICE_THREADS, |
| }; |
