| /* |
| * Copyright (c) 2020 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 "libavutil/cpu.h" |
| #include "libavutil/channel_layout.h" |
| #include "libavutil/ffmath.h" |
| #include "libavutil/eval.h" |
| #include "libavutil/mem.h" |
| #include "libavutil/opt.h" |
| #include "libavutil/tx.h" |
| #include "audio.h" |
| #include "avfilter.h" |
| #include "filters.h" |
| #include "formats.h" |
| #include "internal.h" |
| #include "window_func.h" |
| |
| typedef struct AudioFIRSourceContext { |
| const AVClass *class; |
| |
| char *freq_points_str; |
| char *magnitude_str; |
| char *phase_str; |
| int nb_taps; |
| int sample_rate; |
| int nb_samples; |
| int win_func; |
| int preset; |
| int interp; |
| int phaset; |
| |
| AVComplexFloat *complexf; |
| float *freq; |
| float *magnitude; |
| float *phase; |
| int freq_size; |
| int magnitude_size; |
| int phase_size; |
| int nb_freq; |
| int nb_magnitude; |
| int nb_phase; |
| |
| float *taps; |
| float *win; |
| int64_t pts; |
| |
| AVTXContext *tx_ctx, *itx_ctx; |
| av_tx_fn tx_fn, itx_fn; |
| } AudioFIRSourceContext; |
| |
| #define OFFSET(x) offsetof(AudioFIRSourceContext, x) |
| #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
| |
| static const AVOption afirsrc_options[] = { |
| { "taps", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=1025}, 9, UINT16_MAX, FLAGS }, |
| { "t", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=1025}, 9, UINT16_MAX, FLAGS }, |
| { "frequency", "set frequency points", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str="0 1"}, 0, 0, FLAGS }, |
| { "f", "set frequency points", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str="0 1"}, 0, 0, FLAGS }, |
| { "magnitude", "set magnitude values", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, FLAGS }, |
| { "m", "set magnitude values", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, FLAGS }, |
| { "phase", "set phase values", OFFSET(phase_str), AV_OPT_TYPE_STRING, {.str="0 0"}, 0, 0, FLAGS }, |
| { "p", "set phase values", OFFSET(phase_str), AV_OPT_TYPE_STRING, {.str="0 0"}, 0, 0, FLAGS }, |
| { "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS }, |
| { "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS }, |
| { "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS }, |
| { "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS }, |
| WIN_FUNC_OPTION("win_func", OFFSET(win_func), FLAGS, WFUNC_BLACKMAN), |
| WIN_FUNC_OPTION("w", OFFSET(win_func), FLAGS, WFUNC_BLACKMAN), |
| {NULL} |
| }; |
| |
| AVFILTER_DEFINE_CLASS(afirsrc); |
| |
| static av_cold int init(AVFilterContext *ctx) |
| { |
| AudioFIRSourceContext *s = ctx->priv; |
| |
| if (!(s->nb_taps & 1)) { |
| av_log(s, AV_LOG_WARNING, "Number of taps %d must be odd length.\n", s->nb_taps); |
| s->nb_taps |= 1; |
| } |
| |
| return 0; |
| } |
| |
| static av_cold void uninit(AVFilterContext *ctx) |
| { |
| AudioFIRSourceContext *s = ctx->priv; |
| |
| av_freep(&s->win); |
| av_freep(&s->taps); |
| av_freep(&s->freq); |
| av_freep(&s->magnitude); |
| av_freep(&s->phase); |
| av_freep(&s->complexf); |
| av_tx_uninit(&s->tx_ctx); |
| av_tx_uninit(&s->itx_ctx); |
| } |
| |
| static av_cold int query_formats(AVFilterContext *ctx) |
| { |
| AudioFIRSourceContext *s = ctx->priv; |
| static const AVChannelLayout chlayouts[] = { AV_CHANNEL_LAYOUT_MONO, { 0 } }; |
| int sample_rates[] = { s->sample_rate, -1 }; |
| static const enum AVSampleFormat sample_fmts[] = { |
| AV_SAMPLE_FMT_FLT, |
| AV_SAMPLE_FMT_NONE |
| }; |
| int ret = ff_set_common_formats_from_list(ctx, sample_fmts); |
| if (ret < 0) |
| return ret; |
| |
| ret = ff_set_common_channel_layouts_from_list(ctx, chlayouts); |
| if (ret < 0) |
| return ret; |
| |
| return ff_set_common_samplerates_from_list(ctx, sample_rates); |
| } |
| |
| static int parse_string(char *str, float **items, int *nb_items, int *items_size) |
| { |
| float *new_items; |
| char *tail; |
| |
| new_items = av_fast_realloc(NULL, items_size, sizeof(float)); |
| if (!new_items) |
| return AVERROR(ENOMEM); |
| *items = new_items; |
| |
| tail = str; |
| if (!tail) |
| return AVERROR(EINVAL); |
| |
| do { |
| (*items)[(*nb_items)++] = av_strtod(tail, &tail); |
| new_items = av_fast_realloc(*items, items_size, (*nb_items + 2) * sizeof(float)); |
| if (!new_items) |
| return AVERROR(ENOMEM); |
| *items = new_items; |
| if (tail && *tail) |
| tail++; |
| } while (tail && *tail); |
| |
| return 0; |
| } |
| |
| static void lininterp(AVComplexFloat *complexf, |
| const float *freq, |
| const float *magnitude, |
| const float *phase, |
| int m, int minterp) |
| { |
| for (int i = 0; i < minterp; i++) { |
| for (int j = 1; j < m; j++) { |
| const float x = i / (float)minterp; |
| |
| if (x <= freq[j]) { |
| const float mg = (x - freq[j-1]) / (freq[j] - freq[j-1]) * (magnitude[j] - magnitude[j-1]) + magnitude[j-1]; |
| const float ph = (x - freq[j-1]) / (freq[j] - freq[j-1]) * (phase[j] - phase[j-1]) + phase[j-1]; |
| |
| complexf[i].re = mg * cosf(ph); |
| complexf[i].im = mg * sinf(ph); |
| break; |
| } |
| } |
| } |
| } |
| |
| static av_cold int config_output(AVFilterLink *outlink) |
| { |
| AVFilterContext *ctx = outlink->src; |
| AudioFIRSourceContext *s = ctx->priv; |
| float overlap, scale = 1.f, compensation; |
| int fft_size, middle, ret; |
| |
| s->nb_freq = s->nb_magnitude = s->nb_phase = 0; |
| |
| ret = parse_string(s->freq_points_str, &s->freq, &s->nb_freq, &s->freq_size); |
| if (ret < 0) |
| return ret; |
| |
| ret = parse_string(s->magnitude_str, &s->magnitude, &s->nb_magnitude, &s->magnitude_size); |
| if (ret < 0) |
| return ret; |
| |
| ret = parse_string(s->phase_str, &s->phase, &s->nb_phase, &s->phase_size); |
| if (ret < 0) |
| return ret; |
| |
| if (s->nb_freq != s->nb_magnitude && s->nb_freq != s->nb_phase && s->nb_freq >= 2) { |
| av_log(ctx, AV_LOG_ERROR, "Number of frequencies, magnitudes and phases must be same and >= 2.\n"); |
| return AVERROR(EINVAL); |
| } |
| |
| for (int i = 0; i < s->nb_freq; i++) { |
| if (i == 0 && s->freq[i] != 0.f) { |
| av_log(ctx, AV_LOG_ERROR, "First frequency must be 0.\n"); |
| return AVERROR(EINVAL); |
| } |
| |
| if (i == s->nb_freq - 1 && s->freq[i] != 1.f) { |
| av_log(ctx, AV_LOG_ERROR, "Last frequency must be 1.\n"); |
| return AVERROR(EINVAL); |
| } |
| |
| if (i && s->freq[i] < s->freq[i-1]) { |
| av_log(ctx, AV_LOG_ERROR, "Frequencies must be in increasing order.\n"); |
| return AVERROR(EINVAL); |
| } |
| } |
| |
| fft_size = 1 << (av_log2(s->nb_taps) + 1); |
| s->complexf = av_calloc(fft_size * 2, sizeof(*s->complexf)); |
| if (!s->complexf) |
| return AVERROR(ENOMEM); |
| |
| ret = av_tx_init(&s->tx_ctx, &s->tx_fn, AV_TX_FLOAT_FFT, 1, fft_size, &scale, 0); |
| if (ret < 0) |
| return ret; |
| |
| s->taps = av_calloc(s->nb_taps, sizeof(*s->taps)); |
| if (!s->taps) |
| return AVERROR(ENOMEM); |
| |
| s->win = av_calloc(s->nb_taps, sizeof(*s->win)); |
| if (!s->win) |
| return AVERROR(ENOMEM); |
| |
| generate_window_func(s->win, s->nb_taps, s->win_func, &overlap); |
| |
| lininterp(s->complexf, s->freq, s->magnitude, s->phase, s->nb_freq, fft_size / 2); |
| |
| s->tx_fn(s->tx_ctx, s->complexf + fft_size, s->complexf, sizeof(*s->complexf)); |
| |
| compensation = 2.f / fft_size; |
| middle = s->nb_taps / 2; |
| |
| for (int i = 0; i <= middle; i++) { |
| s->taps[ i] = s->complexf[fft_size + middle - i].re * compensation * s->win[i]; |
| s->taps[middle + i] = s->complexf[fft_size + i].re * compensation * s->win[middle + i]; |
| } |
| |
| s->pts = 0; |
| |
| return 0; |
| } |
| |
| static int activate(AVFilterContext *ctx) |
| { |
| AVFilterLink *outlink = ctx->outputs[0]; |
| AudioFIRSourceContext *s = ctx->priv; |
| AVFrame *frame; |
| int nb_samples; |
| |
| if (!ff_outlink_frame_wanted(outlink)) |
| return FFERROR_NOT_READY; |
| |
| nb_samples = FFMIN(s->nb_samples, s->nb_taps - s->pts); |
| if (nb_samples <= 0) { |
| ff_outlink_set_status(outlink, AVERROR_EOF, s->pts); |
| return 0; |
| } |
| |
| if (!(frame = ff_get_audio_buffer(outlink, nb_samples))) |
| return AVERROR(ENOMEM); |
| |
| memcpy(frame->data[0], s->taps + s->pts, nb_samples * sizeof(float)); |
| |
| frame->pts = s->pts; |
| s->pts += nb_samples; |
| return ff_filter_frame(outlink, frame); |
| } |
| |
| static const AVFilterPad afirsrc_outputs[] = { |
| { |
| .name = "default", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .config_props = config_output, |
| }, |
| }; |
| |
| const AVFilter ff_asrc_afirsrc = { |
| .name = "afirsrc", |
| .description = NULL_IF_CONFIG_SMALL("Generate a FIR coefficients audio stream."), |
| .init = init, |
| .uninit = uninit, |
| .activate = activate, |
| .priv_size = sizeof(AudioFIRSourceContext), |
| .inputs = NULL, |
| FILTER_OUTPUTS(afirsrc_outputs), |
| FILTER_QUERY_FUNC(query_formats), |
| .priv_class = &afirsrc_class, |
| }; |
| |
| #define DEFAULT_BANDS "25 40 63 100 160 250 400 630 1000 1600 2500 4000 6300 10000 16000 24000" |
| |
| typedef struct EqPreset { |
| char name[16]; |
| float gains[16]; |
| } EqPreset; |
| |
| static const EqPreset eq_presets[] = { |
| { "flat", { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 } }, |
| { "acoustic", { 5.0, 4.5, 4.0, 3.5, 1.5, 1.0, 1.5, 1.5, 2.0, 3.0, 3.5, 4.0, 3.7, 3.0, 3.0 } }, |
| { "bass", { 10.0, 8.8, 8.5, 6.5, 2.5, 1.5, 0, 0, 0, 0, 0, 0, 0, 0, 0 } }, |
| { "beats", { -5.5, -5.0, -4.5, -4.2, -3.5, -3.0, -1.9, 0, 0, 0, 0, 0, 0, 0, 0 } }, |
| { "classic", { -0.3, 0.3, -3.5, -9.0, -1.0, 0.0, 1.8, 2.1, 0.0, 0.0, 0.0, 4.4, 9.0, 9.0, 9.0 } }, |
| { "clear", { 3.5, 5.5, 6.5, 9.5, 8.0, 6.5, 3.5, 2.5, 1.3, 5.0, 7.0, 9.0, 10.0, 11.0, 9.0 } }, |
| { "deep bass", { 12.0, 8.0, 0.0, -6.7, -12.0, -9.0, -3.5, -3.5, -6.1, 0.0, -3.0, -5.0, 0.0, 1.2, 3.0 } }, |
| { "dubstep", { 12.0, 10.0, 0.5, -1.0, -3.0, -5.0, -5.0, -4.8, -4.5, -2.5, -1.0, 0.0, -2.5, -2.5, 0.0 } }, |
| { "electronic", { 4.0, 4.0, 3.5, 1.0, 0.0, -0.5, -2.0, 0.0, 2.0, 0.0, 0.0, 1.0, 3.0, 4.0, 4.5 } }, |
| { "hardstyle", { 6.1, 7.0, 12.0, 6.1, -5.0, -12.0, -2.5, 3.0, 6.5, 0.0, -2.2, -4.5, -6.1, -9.2, -10.0 } }, |
| { "hip-hop", { 4.5, 4.3, 4.0, 2.5, 1.5, 3.0, -1.0, -1.5, -1.5, 1.5, 0.0, -1.0, 0.0, 1.5, 3.0 } }, |
| { "jazz", { 0.0, 0.0, 0.0, 2.0, 4.0, 5.9, -5.9, -4.5, -2.5, 2.5, 1.0, -0.8, -0.8, -0.8, -0.8 } }, |
| { "metal", { 10.5, 10.5, 7.5, 0.0, 2.0, 5.5, 0.0, 0.0, 0.0, 6.1, 0.0, 0.0, 6.1, 10.0, 12.0 } }, |
| { "movie", { 3.0, 3.0, 6.1, 8.5, 9.0, 7.0, 6.1, 6.1, 5.0, 8.0, 3.5, 3.5, 8.0, 10.0, 8.0 } }, |
| { "pop", { 0.0, 0.0, 0.0, 0.0, 0.0, 1.3, 2.0, 2.5, 5.0, -1.5, -2.0, -3.0, -3.0, -3.0, -3.0 } }, |
| { "r&b", { 3.0, 3.0, 7.0, 6.1, 4.5, 1.5, -1.5, -2.0, -1.5, 2.0, 2.5, 3.0, 3.5, 3.8, 4.0 } }, |
| { "rock", { 0.0, 0.0, 0.0, 3.0, 3.0, -10.0, -4.0, -1.0, 0.8, 3.0, 3.0, 3.0, 3.0, 3.0, 3.0 } }, |
| { "vocal booster", { -1.5, -2.0, -3.0, -3.0, -0.5, 1.5, 3.5, 3.5, 3.5, 3.0, 2.0, 1.5, 0.0, 0.0, -1.5 } }, |
| }; |
| |
| static const AVOption afireqsrc_options[] = { |
| { "preset","set equalizer preset", OFFSET(preset), AV_OPT_TYPE_INT, {.i64=0}, -1, FF_ARRAY_ELEMS(eq_presets)-1, FLAGS, .unit = "preset" }, |
| { "p", "set equalizer preset", OFFSET(preset), AV_OPT_TYPE_INT, {.i64=0}, -1, FF_ARRAY_ELEMS(eq_presets)-1, FLAGS, .unit = "preset" }, |
| { "custom", NULL, 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, FLAGS, .unit = "preset" }, |
| { eq_presets[ 0].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 0}, 0, 0, FLAGS, .unit = "preset" }, |
| { eq_presets[ 1].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 1}, 0, 0, FLAGS, .unit = "preset" }, |
| { eq_presets[ 2].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 2}, 0, 0, FLAGS, .unit = "preset" }, |
| { eq_presets[ 3].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 3}, 0, 0, FLAGS, .unit = "preset" }, |
| { eq_presets[ 4].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 4}, 0, 0, FLAGS, .unit = "preset" }, |
| { eq_presets[ 5].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 5}, 0, 0, FLAGS, .unit = "preset" }, |
| { eq_presets[ 6].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 6}, 0, 0, FLAGS, .unit = "preset" }, |
| { eq_presets[ 7].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 7}, 0, 0, FLAGS, .unit = "preset" }, |
| { eq_presets[ 8].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 8}, 0, 0, FLAGS, .unit = "preset" }, |
| { eq_presets[ 9].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 9}, 0, 0, FLAGS, .unit = "preset" }, |
| { eq_presets[10].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=10}, 0, 0, FLAGS, .unit = "preset" }, |
| { eq_presets[11].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=11}, 0, 0, FLAGS, .unit = "preset" }, |
| { eq_presets[12].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=12}, 0, 0, FLAGS, .unit = "preset" }, |
| { eq_presets[13].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=13}, 0, 0, FLAGS, .unit = "preset" }, |
| { eq_presets[14].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=14}, 0, 0, FLAGS, .unit = "preset" }, |
| { eq_presets[15].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=15}, 0, 0, FLAGS, .unit = "preset" }, |
| { eq_presets[16].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=16}, 0, 0, FLAGS, .unit = "preset" }, |
| { eq_presets[17].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=17}, 0, 0, FLAGS, .unit = "preset" }, |
| { "gains", "set gain values per band", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0"}, 0, 0, FLAGS }, |
| { "g", "set gain values per band", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0"}, 0, 0, FLAGS }, |
| { "bands", "set central frequency values per band", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str=DEFAULT_BANDS}, 0, 0, FLAGS }, |
| { "b", "set central frequency values per band", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str=DEFAULT_BANDS}, 0, 0, FLAGS }, |
| { "taps", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=4096}, 16, UINT16_MAX, FLAGS }, |
| { "t", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=4096}, 16, UINT16_MAX, FLAGS }, |
| { "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS }, |
| { "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS }, |
| { "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS }, |
| { "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS }, |
| { "interp","set the interpolation", OFFSET(interp), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS, .unit = "interp" }, |
| { "i", "set the interpolation", OFFSET(interp), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS, .unit = "interp" }, |
| { "linear", NULL, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, .unit = "interp" }, |
| { "cubic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, .unit = "interp" }, |
| { "phase","set the phase", OFFSET(phaset), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, FLAGS, .unit = "phase" }, |
| { "h", "set the phase", OFFSET(phaset), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, FLAGS, .unit = "phase" }, |
| { "linear", "linear phase", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, .unit = "phase" }, |
| { "min", "minimum phase", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, .unit = "phase" }, |
| {NULL} |
| }; |
| |
| AVFILTER_DEFINE_CLASS(afireqsrc); |
| |
| static void eq_interp(AVComplexFloat *complexf, |
| const float *freq, |
| const float *magnitude, |
| int m, int interp, int minterp, |
| const float factor) |
| { |
| for (int i = 0; i < minterp; i++) { |
| for (int j = 0; j < m; j++) { |
| const float x = factor * i; |
| |
| if (x <= freq[j+1]) { |
| float g; |
| |
| if (interp == 0) { |
| const float d = freq[j+1] - freq[j]; |
| const float d0 = x - freq[j]; |
| const float d1 = freq[j+1] - x; |
| const float g0 = magnitude[j]; |
| const float g1 = magnitude[j+1]; |
| |
| if (d0 && d1) { |
| g = (d0 * g1 + d1 * g0) / d; |
| } else if (d0) { |
| g = g1; |
| } else { |
| g = g0; |
| } |
| } else { |
| if (x <= freq[j]) { |
| g = magnitude[j]; |
| } else { |
| float x1, x2, x3; |
| float a, b, c, d; |
| float m0, m1, m2, msum; |
| const float unit = freq[j+1] - freq[j]; |
| |
| m0 = j != 0 ? unit * (magnitude[j] - magnitude[j-1]) / (freq[j] - freq[j-1]) : 0; |
| m1 = magnitude[j+1] - magnitude[j]; |
| m2 = j != minterp - 1 ? unit * (magnitude[j+2] - magnitude[j+1]) / (freq[j+2] - freq[j+1]) : 0; |
| |
| msum = fabsf(m0) + fabsf(m1); |
| m0 = msum > 0.f ? (fabsf(m0) * m1 + fabsf(m1) * m0) / msum : 0.f; |
| msum = fabsf(m1) + fabsf(m2); |
| m1 = msum > 0.f ? (fabsf(m1) * m2 + fabsf(m2) * m1) / msum : 0.f; |
| |
| d = magnitude[j]; |
| c = m0; |
| b = 3.f * magnitude[j+1] - m1 - 2.f * c - 3.f * d; |
| a = magnitude[j+1] - b - c - d; |
| |
| x1 = (x - freq[j]) / unit; |
| x2 = x1 * x1; |
| x3 = x2 * x1; |
| |
| g = a * x3 + b * x2 + c * x1 + d; |
| } |
| } |
| |
| complexf[i].re = g; |
| complexf[i].im = 0; |
| complexf[minterp * 2 - i - 1].re = g; |
| complexf[minterp * 2 - i - 1].im = 0; |
| |
| break; |
| } |
| } |
| } |
| } |
| |
| static av_cold int config_eq_output(AVFilterLink *outlink) |
| { |
| AVFilterContext *ctx = outlink->src; |
| AudioFIRSourceContext *s = ctx->priv; |
| int fft_size, middle, asize, ret; |
| float scale, factor; |
| |
| s->nb_freq = s->nb_magnitude = 0; |
| if (s->preset < 0) { |
| ret = parse_string(s->freq_points_str, &s->freq, &s->nb_freq, &s->freq_size); |
| if (ret < 0) |
| return ret; |
| |
| ret = parse_string(s->magnitude_str, &s->magnitude, &s->nb_magnitude, &s->magnitude_size); |
| if (ret < 0) |
| return ret; |
| } else { |
| char *freq_str; |
| |
| s->nb_magnitude = FF_ARRAY_ELEMS(eq_presets[s->preset].gains); |
| |
| freq_str = av_strdup(DEFAULT_BANDS); |
| if (!freq_str) |
| return AVERROR(ENOMEM); |
| |
| ret = parse_string(freq_str, &s->freq, &s->nb_freq, &s->freq_size); |
| av_free(freq_str); |
| if (ret < 0) |
| return ret; |
| |
| s->magnitude = av_calloc(s->nb_magnitude + 1, sizeof(*s->magnitude)); |
| if (!s->magnitude) |
| return AVERROR(ENOMEM); |
| memcpy(s->magnitude, eq_presets[s->preset].gains, sizeof(*s->magnitude) * s->nb_magnitude); |
| } |
| |
| if (s->nb_freq != s->nb_magnitude || s->nb_freq < 2) { |
| av_log(ctx, AV_LOG_ERROR, "Number of bands and gains must be same and >= 2.\n"); |
| return AVERROR(EINVAL); |
| } |
| |
| s->freq[s->nb_freq] = outlink->sample_rate * 0.5f; |
| s->magnitude[s->nb_freq] = s->magnitude[s->nb_freq-1]; |
| |
| fft_size = s->nb_taps * 2; |
| factor = FFMIN(outlink->sample_rate * 0.5f, s->freq[s->nb_freq - 1]) / (float)fft_size; |
| asize = FFALIGN(fft_size, av_cpu_max_align()); |
| s->complexf = av_calloc(asize * 2, sizeof(*s->complexf)); |
| if (!s->complexf) |
| return AVERROR(ENOMEM); |
| |
| scale = 1.f; |
| ret = av_tx_init(&s->itx_ctx, &s->itx_fn, AV_TX_FLOAT_FFT, 1, fft_size, &scale, 0); |
| if (ret < 0) |
| return ret; |
| |
| s->taps = av_calloc(s->nb_taps, sizeof(*s->taps)); |
| if (!s->taps) |
| return AVERROR(ENOMEM); |
| |
| eq_interp(s->complexf, s->freq, s->magnitude, s->nb_freq, s->interp, s->nb_taps, factor); |
| |
| for (int i = 0; i < fft_size; i++) |
| s->complexf[i].re = ff_exp10f(s->complexf[i].re / 20.f); |
| |
| if (s->phaset) { |
| const float threshold = powf(10.f, -100.f / 20.f); |
| const float logt = logf(threshold); |
| |
| scale = 1.f; |
| ret = av_tx_init(&s->tx_ctx, &s->tx_fn, AV_TX_FLOAT_FFT, 0, fft_size, &scale, 0); |
| if (ret < 0) |
| return ret; |
| |
| for (int i = 0; i < fft_size; i++) |
| s->complexf[i].re = s->complexf[i].re < threshold ? logt : logf(s->complexf[i].re); |
| |
| s->itx_fn(s->itx_ctx, s->complexf + asize, s->complexf, sizeof(float)); |
| for (int i = 0; i < fft_size; i++) { |
| s->complexf[i + asize].re /= fft_size; |
| s->complexf[i + asize].im /= fft_size; |
| } |
| |
| for (int i = 1; i < s->nb_taps; i++) { |
| s->complexf[asize + i].re += s->complexf[asize + fft_size - i].re; |
| s->complexf[asize + i].im -= s->complexf[asize + fft_size - i].im; |
| s->complexf[asize + fft_size - i].re = 0.f; |
| s->complexf[asize + fft_size - i].im = 0.f; |
| } |
| s->complexf[asize + s->nb_taps - 1].im *= -1.f; |
| |
| s->tx_fn(s->tx_ctx, s->complexf, s->complexf + asize, sizeof(float)); |
| |
| for (int i = 0; i < fft_size; i++) { |
| float eR = expf(s->complexf[i].re); |
| |
| s->complexf[i].re = eR * cosf(s->complexf[i].im); |
| s->complexf[i].im = eR * sinf(s->complexf[i].im); |
| } |
| |
| s->itx_fn(s->itx_ctx, s->complexf + asize, s->complexf, sizeof(float)); |
| |
| for (int i = 0; i < s->nb_taps; i++) |
| s->taps[i] = s->complexf[i + asize].re / fft_size; |
| } else { |
| s->itx_fn(s->itx_ctx, s->complexf + asize, s->complexf, sizeof(float)); |
| |
| middle = s->nb_taps / 2; |
| for (int i = 0; i < middle; i++) { |
| s->taps[middle - i] = s->complexf[i + asize].re / fft_size; |
| s->taps[middle + i] = s->complexf[i + asize].re / fft_size; |
| } |
| } |
| |
| s->pts = 0; |
| |
| return 0; |
| } |
| |
| static const AVFilterPad afireqsrc_outputs[] = { |
| { |
| .name = "default", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .config_props = config_eq_output, |
| }, |
| }; |
| |
| const AVFilter ff_asrc_afireqsrc = { |
| .name = "afireqsrc", |
| .description = NULL_IF_CONFIG_SMALL("Generate a FIR equalizer coefficients audio stream."), |
| .uninit = uninit, |
| .activate = activate, |
| .priv_size = sizeof(AudioFIRSourceContext), |
| .inputs = NULL, |
| FILTER_OUTPUTS(afireqsrc_outputs), |
| FILTER_QUERY_FUNC(query_formats), |
| .priv_class = &afireqsrc_class, |
| }; |