| /* |
| * 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 |
| */ |
| |
| /** |
| * @file |
| * Crossover filter |
| * |
| * Split an audio stream into several bands. |
| */ |
| |
| #include "libavutil/attributes.h" |
| #include "libavutil/avstring.h" |
| #include "libavutil/channel_layout.h" |
| #include "libavutil/eval.h" |
| #include "libavutil/float_dsp.h" |
| #include "libavutil/internal.h" |
| #include "libavutil/opt.h" |
| |
| #include "audio.h" |
| #include "avfilter.h" |
| #include "formats.h" |
| #include "internal.h" |
| |
| #define MAX_SPLITS 16 |
| #define MAX_BANDS MAX_SPLITS + 1 |
| |
| #define B0 0 |
| #define B1 1 |
| #define B2 2 |
| #define A1 3 |
| #define A2 4 |
| |
| typedef struct BiquadCoeffs { |
| double cd[5]; |
| float cf[5]; |
| } BiquadCoeffs; |
| |
| typedef struct AudioCrossoverContext { |
| const AVClass *class; |
| |
| char *splits_str; |
| char *gains_str; |
| int order_opt; |
| float level_in; |
| |
| int order; |
| int filter_count; |
| int first_order; |
| int ap_filter_count; |
| int nb_splits; |
| float splits[MAX_SPLITS]; |
| |
| float gains[MAX_BANDS]; |
| |
| BiquadCoeffs lp[MAX_BANDS][20]; |
| BiquadCoeffs hp[MAX_BANDS][20]; |
| BiquadCoeffs ap[MAX_BANDS][20]; |
| |
| AVFrame *xover; |
| |
| AVFrame *input_frame; |
| AVFrame *frames[MAX_BANDS]; |
| |
| int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs); |
| |
| AVFloatDSPContext *fdsp; |
| } AudioCrossoverContext; |
| |
| #define OFFSET(x) offsetof(AudioCrossoverContext, x) |
| #define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM |
| |
| static const AVOption acrossover_options[] = { |
| { "split", "set split frequencies", OFFSET(splits_str), AV_OPT_TYPE_STRING, {.str="500"}, 0, 0, AF }, |
| { "order", "set filter order", OFFSET(order_opt), AV_OPT_TYPE_INT, {.i64=1}, 0, 9, AF, "m" }, |
| { "2nd", "2nd order (12 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "m" }, |
| { "4th", "4th order (24 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "m" }, |
| { "6th", "6th order (36 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "m" }, |
| { "8th", "8th order (48 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "m" }, |
| { "10th", "10th order (60 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF, "m" }, |
| { "12th", "12th order (72 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, AF, "m" }, |
| { "14th", "14th order (84 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, AF, "m" }, |
| { "16th", "16th order (96 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, AF, "m" }, |
| { "18th", "18th order (108 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, AF, "m" }, |
| { "20th", "20th order (120 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=9}, 0, 0, AF, "m" }, |
| { "level", "set input gain", OFFSET(level_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF }, |
| { "gain", "set output bands gain", OFFSET(gains_str), AV_OPT_TYPE_STRING, {.str="1.f"}, 0, 0, AF }, |
| { NULL } |
| }; |
| |
| AVFILTER_DEFINE_CLASS(acrossover); |
| |
| static int parse_gains(AVFilterContext *ctx) |
| { |
| AudioCrossoverContext *s = ctx->priv; |
| char *p, *arg, *saveptr = NULL; |
| int i, ret = 0; |
| |
| saveptr = NULL; |
| p = s->gains_str; |
| for (i = 0; i < MAX_BANDS; i++) { |
| float gain; |
| char c[3] = { 0 }; |
| |
| if (!(arg = av_strtok(p, " |", &saveptr))) |
| break; |
| |
| p = NULL; |
| |
| if (av_sscanf(arg, "%f%2s", &gain, c) < 1) { |
| av_log(ctx, AV_LOG_ERROR, "Invalid syntax for gain[%d].\n", i); |
| ret = AVERROR(EINVAL); |
| break; |
| } |
| |
| if (c[0] == 'd' && c[1] == 'B') |
| s->gains[i] = expf(gain * M_LN10 / 20.f); |
| else |
| s->gains[i] = gain; |
| } |
| |
| for (; i < MAX_BANDS; i++) |
| s->gains[i] = 1.f; |
| |
| return ret; |
| } |
| |
| static av_cold int init(AVFilterContext *ctx) |
| { |
| AudioCrossoverContext *s = ctx->priv; |
| char *p, *arg, *saveptr = NULL; |
| int i, ret = 0; |
| |
| s->fdsp = avpriv_float_dsp_alloc(0); |
| if (!s->fdsp) |
| return AVERROR(ENOMEM); |
| |
| p = s->splits_str; |
| for (i = 0; i < MAX_SPLITS; i++) { |
| float freq; |
| |
| if (!(arg = av_strtok(p, " |", &saveptr))) |
| break; |
| |
| p = NULL; |
| |
| if (av_sscanf(arg, "%f", &freq) != 1) { |
| av_log(ctx, AV_LOG_ERROR, "Invalid syntax for frequency[%d].\n", i); |
| return AVERROR(EINVAL); |
| } |
| if (freq <= 0) { |
| av_log(ctx, AV_LOG_ERROR, "Frequency %f must be positive number.\n", freq); |
| return AVERROR(EINVAL); |
| } |
| |
| if (i > 0 && freq <= s->splits[i-1]) { |
| av_log(ctx, AV_LOG_ERROR, "Frequency %f must be in increasing order.\n", freq); |
| return AVERROR(EINVAL); |
| } |
| |
| s->splits[i] = freq; |
| } |
| |
| s->nb_splits = i; |
| |
| ret = parse_gains(ctx); |
| if (ret < 0) |
| return ret; |
| |
| for (i = 0; i <= s->nb_splits; i++) { |
| AVFilterPad pad = { 0 }; |
| char *name; |
| |
| pad.type = AVMEDIA_TYPE_AUDIO; |
| name = av_asprintf("out%d", ctx->nb_outputs); |
| if (!name) |
| return AVERROR(ENOMEM); |
| pad.name = name; |
| |
| if ((ret = ff_insert_outpad(ctx, i, &pad)) < 0) { |
| av_freep(&pad.name); |
| return ret; |
| } |
| } |
| |
| return ret; |
| } |
| |
| static void set_lp(BiquadCoeffs *b, double fc, double q, double sr) |
| { |
| double omega = 2. * M_PI * fc / sr; |
| double cosine = cos(omega); |
| double alpha = sin(omega) / (2. * q); |
| |
| double b0 = (1. - cosine) / 2.; |
| double b1 = 1. - cosine; |
| double b2 = (1. - cosine) / 2.; |
| double a0 = 1. + alpha; |
| double a1 = -2. * cosine; |
| double a2 = 1. - alpha; |
| |
| b->cd[B0] = b0 / a0; |
| b->cd[B1] = b1 / a0; |
| b->cd[B2] = b2 / a0; |
| b->cd[A1] = -a1 / a0; |
| b->cd[A2] = -a2 / a0; |
| |
| b->cf[B0] = b->cd[B0]; |
| b->cf[B1] = b->cd[B1]; |
| b->cf[B2] = b->cd[B2]; |
| b->cf[A1] = b->cd[A1]; |
| b->cf[A2] = b->cd[A2]; |
| } |
| |
| static void set_hp(BiquadCoeffs *b, double fc, double q, double sr) |
| { |
| double omega = 2. * M_PI * fc / sr; |
| double cosine = cos(omega); |
| double alpha = sin(omega) / (2. * q); |
| |
| double b0 = (1. + cosine) / 2.; |
| double b1 = -1. - cosine; |
| double b2 = (1. + cosine) / 2.; |
| double a0 = 1. + alpha; |
| double a1 = -2. * cosine; |
| double a2 = 1. - alpha; |
| |
| b->cd[B0] = b0 / a0; |
| b->cd[B1] = b1 / a0; |
| b->cd[B2] = b2 / a0; |
| b->cd[A1] = -a1 / a0; |
| b->cd[A2] = -a2 / a0; |
| |
| b->cf[B0] = b->cd[B0]; |
| b->cf[B1] = b->cd[B1]; |
| b->cf[B2] = b->cd[B2]; |
| b->cf[A1] = b->cd[A1]; |
| b->cf[A2] = b->cd[A2]; |
| } |
| |
| static void set_ap(BiquadCoeffs *b, double fc, double q, double sr) |
| { |
| double omega = 2. * M_PI * fc / sr; |
| double cosine = cos(omega); |
| double alpha = sin(omega) / (2. * q); |
| |
| double a0 = 1. + alpha; |
| double a1 = -2. * cosine; |
| double a2 = 1. - alpha; |
| double b0 = a2; |
| double b1 = a1; |
| double b2 = a0; |
| |
| b->cd[B0] = b0 / a0; |
| b->cd[B1] = b1 / a0; |
| b->cd[B2] = b2 / a0; |
| b->cd[A1] = -a1 / a0; |
| b->cd[A2] = -a2 / a0; |
| |
| b->cf[B0] = b->cd[B0]; |
| b->cf[B1] = b->cd[B1]; |
| b->cf[B2] = b->cd[B2]; |
| b->cf[A1] = b->cd[A1]; |
| b->cf[A2] = b->cd[A2]; |
| } |
| |
| static void set_ap1(BiquadCoeffs *b, double fc, double sr) |
| { |
| double omega = 2. * M_PI * fc / sr; |
| |
| b->cd[A1] = exp(-omega); |
| b->cd[A2] = 0.; |
| b->cd[B0] = -b->cd[A1]; |
| b->cd[B1] = 1.; |
| b->cd[B2] = 0.; |
| |
| b->cf[B0] = b->cd[B0]; |
| b->cf[B1] = b->cd[B1]; |
| b->cf[B2] = b->cd[B2]; |
| b->cf[A1] = b->cd[A1]; |
| b->cf[A2] = b->cd[A2]; |
| } |
| |
| static void calc_q_factors(int order, double *q) |
| { |
| double n = order / 2.; |
| |
| for (int i = 0; i < n / 2; i++) |
| q[i] = 1. / (-2. * cos(M_PI * (2. * (i + 1) + n - 1.) / (2. * n))); |
| } |
| |
| static int query_formats(AVFilterContext *ctx) |
| { |
| AVFilterFormats *formats; |
| AVFilterChannelLayouts *layouts; |
| static const enum AVSampleFormat sample_fmts[] = { |
| AV_SAMPLE_FMT_FLTP, 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; |
| |
| 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 BIQUAD_PROCESS(name, type) \ |
| static void biquad_process_## name(const type *const c, \ |
| type *b, \ |
| type *dst, const type *src, \ |
| int nb_samples) \ |
| { \ |
| const type b0 = c[B0]; \ |
| const type b1 = c[B1]; \ |
| const type b2 = c[B2]; \ |
| const type a1 = c[A1]; \ |
| const type a2 = c[A2]; \ |
| type z1 = b[0]; \ |
| type z2 = b[1]; \ |
| \ |
| for (int n = 0; n + 1 < nb_samples; n++) { \ |
| type in = src[n]; \ |
| type out; \ |
| \ |
| out = in * b0 + z1; \ |
| z1 = b1 * in + z2 + a1 * out; \ |
| z2 = b2 * in + a2 * out; \ |
| dst[n] = out; \ |
| \ |
| n++; \ |
| in = src[n]; \ |
| out = in * b0 + z1; \ |
| z1 = b1 * in + z2 + a1 * out; \ |
| z2 = b2 * in + a2 * out; \ |
| dst[n] = out; \ |
| } \ |
| \ |
| if (nb_samples & 1) { \ |
| const int n = nb_samples - 1; \ |
| const type in = src[n]; \ |
| type out; \ |
| \ |
| out = in * b0 + z1; \ |
| z1 = b1 * in + z2 + a1 * out; \ |
| z2 = b2 * in + a2 * out; \ |
| dst[n] = out; \ |
| } \ |
| \ |
| b[0] = z1; \ |
| b[1] = z2; \ |
| } |
| |
| BIQUAD_PROCESS(fltp, float) |
| BIQUAD_PROCESS(dblp, double) |
| |
| #define XOVER_PROCESS(name, type, one, ff) \ |
| static int filter_channels_## name(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) \ |
| { \ |
| AudioCrossoverContext *s = ctx->priv; \ |
| AVFrame *in = s->input_frame; \ |
| AVFrame **frames = s->frames; \ |
| const int start = (in->channels * jobnr) / nb_jobs; \ |
| const int end = (in->channels * (jobnr+1)) / nb_jobs; \ |
| const int nb_samples = in->nb_samples; \ |
| const int nb_outs = ctx->nb_outputs; \ |
| const int first_order = s->first_order; \ |
| \ |
| for (int ch = start; ch < end; ch++) { \ |
| const type *src = (const type *)in->extended_data[ch]; \ |
| type *xover = (type *)s->xover->extended_data[ch]; \ |
| \ |
| s->fdsp->vector_## ff ##mul_scalar((type *)frames[0]->extended_data[ch], src, \ |
| s->level_in, FFALIGN(nb_samples, sizeof(type))); \ |
| \ |
| for (int band = 0; band < nb_outs; band++) { \ |
| for (int f = 0; band + 1 < nb_outs && f < s->filter_count; f++) { \ |
| const type *prv = (const type *)frames[band]->extended_data[ch]; \ |
| type *dst = (type *)frames[band + 1]->extended_data[ch]; \ |
| const type *hsrc = f == 0 ? prv : dst; \ |
| type *hp = xover + nb_outs * 20 + band * 20 + f * 2; \ |
| const type *const hpc = (type *)&s->hp[band][f].c ## ff; \ |
| \ |
| biquad_process_## name(hpc, hp, dst, hsrc, nb_samples); \ |
| } \ |
| \ |
| for (int f = 0; band + 1 < nb_outs && f < s->filter_count; f++) { \ |
| type *dst = (type *)frames[band]->extended_data[ch]; \ |
| const type *lsrc = dst; \ |
| type *lp = xover + band * 20 + f * 2; \ |
| const type *const lpc = (type *)&s->lp[band][f].c ## ff; \ |
| \ |
| biquad_process_## name(lpc, lp, dst, lsrc, nb_samples); \ |
| } \ |
| \ |
| for (int aband = band + 1; aband + 1 < nb_outs; aband++) { \ |
| if (first_order) { \ |
| const type *asrc = (const type *)frames[band]->extended_data[ch]; \ |
| type *dst = (type *)frames[band]->extended_data[ch]; \ |
| type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20; \ |
| const type *const apc = (type *)&s->ap[aband][0].c ## ff; \ |
| \ |
| biquad_process_## name(apc, ap, dst, asrc, nb_samples); \ |
| } \ |
| \ |
| for (int f = first_order; f < s->ap_filter_count; f++) { \ |
| const type *asrc = (const type *)frames[band]->extended_data[ch]; \ |
| type *dst = (type *)frames[band]->extended_data[ch]; \ |
| type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20 + f * 2;\ |
| const type *const apc = (type *)&s->ap[aband][f].c ## ff; \ |
| \ |
| biquad_process_## name(apc, ap, dst, asrc, nb_samples); \ |
| } \ |
| } \ |
| } \ |
| \ |
| for (int band = 0; band < nb_outs; band++) { \ |
| const type gain = s->gains[band] * ((band & 1 && first_order) ? -one : one); \ |
| type *dst = (type *)frames[band]->extended_data[ch]; \ |
| \ |
| s->fdsp->vector_## ff ##mul_scalar(dst, dst, gain, \ |
| FFALIGN(nb_samples, sizeof(type))); \ |
| } \ |
| } \ |
| \ |
| return 0; \ |
| } |
| |
| XOVER_PROCESS(fltp, float, 1.f, f) |
| XOVER_PROCESS(dblp, double, 1.0, d) |
| |
| static int config_input(AVFilterLink *inlink) |
| { |
| AVFilterContext *ctx = inlink->dst; |
| AudioCrossoverContext *s = ctx->priv; |
| int sample_rate = inlink->sample_rate; |
| double q[16]; |
| |
| s->order = (s->order_opt + 1) * 2; |
| s->filter_count = s->order / 2; |
| s->first_order = s->filter_count & 1; |
| s->ap_filter_count = s->filter_count / 2 + s->first_order; |
| calc_q_factors(s->order, q); |
| |
| for (int band = 0; band <= s->nb_splits; band++) { |
| if (s->first_order) { |
| set_lp(&s->lp[band][0], s->splits[band], 0.5, sample_rate); |
| set_hp(&s->hp[band][0], s->splits[band], 0.5, sample_rate); |
| } |
| |
| for (int n = s->first_order; n < s->filter_count; n++) { |
| const int idx = s->filter_count / 2 - ((n + s->first_order) / 2 - s->first_order) - 1; |
| |
| set_lp(&s->lp[band][n], s->splits[band], q[idx], sample_rate); |
| set_hp(&s->hp[band][n], s->splits[band], q[idx], sample_rate); |
| } |
| |
| if (s->first_order) |
| set_ap1(&s->ap[band][0], s->splits[band], sample_rate); |
| |
| for (int n = s->first_order; n < s->ap_filter_count; n++) { |
| const int idx = (s->filter_count / 2 - ((n * 2 + s->first_order) / 2 - s->first_order) - 1); |
| |
| set_ap(&s->ap[band][n], s->splits[band], q[idx], sample_rate); |
| } |
| } |
| |
| switch (inlink->format) { |
| case AV_SAMPLE_FMT_FLTP: s->filter_channels = filter_channels_fltp; break; |
| case AV_SAMPLE_FMT_DBLP: s->filter_channels = filter_channels_dblp; break; |
| } |
| |
| s->xover = ff_get_audio_buffer(inlink, 2 * (ctx->nb_outputs * 10 + ctx->nb_outputs * 10 + |
| ctx->nb_outputs * ctx->nb_outputs * 10)); |
| if (!s->xover) |
| return AVERROR(ENOMEM); |
| |
| return 0; |
| } |
| |
| static int filter_frame(AVFilterLink *inlink, AVFrame *in) |
| { |
| AVFilterContext *ctx = inlink->dst; |
| AudioCrossoverContext *s = ctx->priv; |
| AVFrame **frames = s->frames; |
| int i, ret = 0; |
| |
| for (i = 0; i < ctx->nb_outputs; i++) { |
| frames[i] = ff_get_audio_buffer(ctx->outputs[i], in->nb_samples); |
| |
| if (!frames[i]) { |
| ret = AVERROR(ENOMEM); |
| break; |
| } |
| |
| frames[i]->pts = in->pts; |
| } |
| |
| if (ret < 0) |
| goto fail; |
| |
| s->input_frame = in; |
| ctx->internal->execute(ctx, s->filter_channels, NULL, NULL, FFMIN(inlink->channels, |
| ff_filter_get_nb_threads(ctx))); |
| |
| for (i = 0; i < ctx->nb_outputs; i++) { |
| ret = ff_filter_frame(ctx->outputs[i], frames[i]); |
| frames[i] = NULL; |
| if (ret < 0) |
| break; |
| } |
| |
| fail: |
| for (i = 0; i < ctx->nb_outputs; i++) |
| av_frame_free(&frames[i]); |
| av_frame_free(&in); |
| s->input_frame = NULL; |
| |
| return ret; |
| } |
| |
| static av_cold void uninit(AVFilterContext *ctx) |
| { |
| AudioCrossoverContext *s = ctx->priv; |
| int i; |
| |
| av_freep(&s->fdsp); |
| av_frame_free(&s->xover); |
| |
| for (i = 0; i < ctx->nb_outputs; i++) |
| av_freep(&ctx->output_pads[i].name); |
| } |
| |
| static const AVFilterPad inputs[] = { |
| { |
| .name = "default", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .filter_frame = filter_frame, |
| .config_props = config_input, |
| }, |
| { NULL } |
| }; |
| |
| AVFilter ff_af_acrossover = { |
| .name = "acrossover", |
| .description = NULL_IF_CONFIG_SMALL("Split audio into per-bands streams."), |
| .priv_size = sizeof(AudioCrossoverContext), |
| .priv_class = &acrossover_class, |
| .init = init, |
| .uninit = uninit, |
| .query_formats = query_formats, |
| .inputs = inputs, |
| .outputs = NULL, |
| .flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS | |
| AVFILTER_FLAG_SLICE_THREADS, |
| }; |
