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https://github.com/nyanmisaka/ffmpeg-rockchip.git
synced 2025-10-09 18:50:56 +08:00
avfilter/af_adynamicequalizer: add precision option
This commit is contained in:
Paul B Mahol
@@ -43,242 +43,82 @@ typedef struct AudioDynamicEqualizerContext {
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int detection;
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int tftype;
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int dftype;
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int precision;
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int format;
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int (*filter_prepare)(AVFilterContext *ctx);
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int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
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double da_double[3], dm_double[3];
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float da_float[3], dm_float[3];
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double da[3], dm[3];
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AVFrame *state;
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} AudioDynamicEqualizerContext;
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static int config_input(AVFilterLink *inlink)
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterContext *ctx = inlink->dst;
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AudioDynamicEqualizerContext *s = ctx->priv;
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static const enum AVSampleFormat sample_fmts[3][3] = {
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{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
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{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE },
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{ AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
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};
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int ret;
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s->state = ff_get_audio_buffer(inlink, 8);
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if (!s->state)
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return AVERROR(ENOMEM);
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if ((ret = ff_set_common_all_channel_counts(ctx)) < 0)
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return ret;
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for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
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double *state = (double *)s->state->extended_data[ch];
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if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts[s->precision])) < 0)
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return ret;
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state[4] = 1.;
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}
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return 0;
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return ff_set_common_all_samplerates(ctx);
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}
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static double get_svf(double in, const double *m, const double *a, double *b)
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{
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const double v0 = in;
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const double v3 = v0 - b[1];
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const double v1 = a[0] * b[0] + a[1] * v3;
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const double v2 = b[1] + a[1] * b[0] + a[2] * v3;
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b[0] = 2. * v1 - b[0];
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b[1] = 2. * v2 - b[1];
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return m[0] * v0 + m[1] * v1 + m[2] * v2;
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}
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typedef struct ThreadData {
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AVFrame *in, *out;
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} ThreadData;
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static double get_coef(double x, double sr)
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{
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return exp(-1000. / (x * sr));
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}
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static int filter_prepare(AVFilterContext *ctx)
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typedef struct ThreadData {
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AVFrame *in, *out;
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} ThreadData;
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#define DEPTH 32
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#include "adynamicequalizer_template.c"
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#undef DEPTH
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#define DEPTH 64
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#include "adynamicequalizer_template.c"
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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AudioDynamicEqualizerContext *s = ctx->priv;
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const double sample_rate = ctx->inputs[0]->sample_rate;
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const double dfrequency = fmin(s->dfrequency, sample_rate * 0.5);
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const double dg = tan(M_PI * dfrequency / sample_rate);
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const double dqfactor = s->dqfactor;
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const int dftype = s->dftype;
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double *da = s->da;
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double *dm = s->dm;
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double k;
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s->attack_coef = get_coef(s->attack, sample_rate);
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s->release_coef = get_coef(s->release, sample_rate);
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s->format = inlink->format;
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s->state = ff_get_audio_buffer(inlink, 8);
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if (!s->state)
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return AVERROR(ENOMEM);
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switch (dftype) {
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case 0:
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k = 1. / dqfactor;
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switch (s->format) {
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case AV_SAMPLE_FMT_DBLP:
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for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
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double *state = (double *)s->state->extended_data[ch];
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da[0] = 1. / (1. + dg * (dg + k));
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da[1] = dg * da[0];
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da[2] = dg * da[1];
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dm[0] = 0.;
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dm[1] = k;
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dm[2] = 0.;
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break;
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case 1:
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k = 1. / dqfactor;
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da[0] = 1. / (1. + dg * (dg + k));
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da[1] = dg * da[0];
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da[2] = dg * da[1];
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dm[0] = 0.;
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dm[1] = 0.;
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dm[2] = 1.;
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break;
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case 2:
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k = 1. / dqfactor;
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da[0] = 1. / (1. + dg * (dg + k));
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da[1] = dg * da[0];
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da[2] = dg * da[1];
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dm[0] = 0.;
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dm[1] = -k;
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dm[2] = -1.;
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break;
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case 3:
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k = 1. / dqfactor;
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da[0] = 1. / (1. + dg * (dg + k));
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da[1] = dg * da[0];
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da[2] = dg * da[1];
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dm[0] = 0.;
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dm[1] = -k;
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dm[2] = -2.;
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break;
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}
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return 0;
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}
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static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
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{
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AudioDynamicEqualizerContext *s = ctx->priv;
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ThreadData *td = arg;
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AVFrame *in = td->in;
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AVFrame *out = td->out;
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const double sample_rate = in->sample_rate;
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const double makeup = s->makeup;
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const double ratio = s->ratio;
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const double range = s->range;
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const double tfrequency = fmin(s->tfrequency, sample_rate * 0.5);
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const double release = s->release_coef;
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const double irelease = 1. - release;
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const double attack = s->attack_coef;
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const double iattack = 1. - attack;
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const double tqfactor = s->tqfactor;
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const double fg = tan(M_PI * tfrequency / sample_rate);
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const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
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const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
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const int detection = s->detection;
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const int direction = s->direction;
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const int tftype = s->tftype;
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const int mode = s->mode;
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const double *da = s->da;
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const double *dm = s->dm;
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for (int ch = start; ch < end; ch++) {
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const double *src = (const double *)in->extended_data[ch];
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double *dst = (double *)out->extended_data[ch];
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double *state = (double *)s->state->extended_data[ch];
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const double threshold = detection == 0 ? state[5] : s->threshold;
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if (detection < 0)
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state[5] = threshold;
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for (int n = 0; n < out->nb_samples; n++) {
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double detect, gain, v, listen;
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double fa[3], fm[3];
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double k, g;
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detect = listen = get_svf(src[n], dm, da, state);
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detect = fabs(detect);
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if (detection > 0)
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state[5] = fmax(state[5], detect);
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if (direction == 0) {
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if (detect < threshold) {
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if (mode == 0)
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detect = 1. / av_clipd(1. + makeup + (threshold - detect) * ratio, 1., range);
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else
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detect = av_clipd(1. + makeup + (threshold - detect) * ratio, 1., range);
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} else {
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detect = 1.;
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}
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} else {
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if (detect > threshold) {
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if (mode == 0)
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detect = 1. / av_clipd(1. + makeup + (detect - threshold) * ratio, 1., range);
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else
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detect = av_clipd(1. + makeup + (detect - threshold) * ratio, 1., range);
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} else {
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detect = 1.;
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}
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}
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if (direction == 0) {
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if (detect > state[4]) {
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detect = iattack * detect + attack * state[4];
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} else {
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detect = irelease * detect + release * state[4];
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}
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} else {
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if (detect < state[4]) {
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detect = iattack * detect + attack * state[4];
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} else {
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detect = irelease * detect + release * state[4];
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}
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}
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if (state[4] != detect || n == 0) {
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state[4] = gain = detect;
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switch (tftype) {
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case 0:
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k = 1. / (tqfactor * gain);
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fa[0] = 1. / (1. + fg * (fg + k));
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fa[1] = fg * fa[0];
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fa[2] = fg * fa[1];
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fm[0] = 1.;
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fm[1] = k * (gain * gain - 1.);
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fm[2] = 0.;
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break;
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case 1:
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k = 1. / tqfactor;
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g = fg / sqrt(gain);
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fa[0] = 1. / (1. + g * (g + k));
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fa[1] = g * fa[0];
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fa[2] = g * fa[1];
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fm[0] = 1.;
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fm[1] = k * (gain - 1.);
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fm[2] = gain * gain - 1.;
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break;
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case 2:
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k = 1. / tqfactor;
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g = fg / sqrt(gain);
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fa[0] = 1. / (1. + g * (g + k));
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fa[1] = g * fa[0];
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fa[2] = g * fa[1];
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fm[0] = gain * gain;
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fm[1] = k * (1. - gain) * gain;
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fm[2] = 1. - gain * gain;
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break;
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}
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}
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v = get_svf(src[n], fm, fa, &state[2]);
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v = mode == -1 ? listen : v;
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dst[n] = ctx->is_disabled ? src[n] : v;
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state[4] = 1.;
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}
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s->filter_prepare = filter_prepare_double;
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s->filter_channels = filter_channels_double;
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break;
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case AV_SAMPLE_FMT_FLTP:
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for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
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float *state = (float *)s->state->extended_data[ch];
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state[4] = 1.;
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}
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s->filter_prepare = filter_prepare_float;
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s->filter_channels = filter_channels_float;
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break;
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}
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return 0;
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@@ -288,6 +128,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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{
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AVFilterContext *ctx = inlink->dst;
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AVFilterLink *outlink = ctx->outputs[0];
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AudioDynamicEqualizerContext *s = ctx->priv;
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ThreadData td;
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AVFrame *out;
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@@ -304,8 +145,8 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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td.in = in;
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td.out = out;
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filter_prepare(ctx);
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ff_filter_execute(ctx, filter_channels, &td, NULL,
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s->filter_prepare(ctx);
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ff_filter_execute(ctx, s->filter_channels, &td, NULL,
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FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
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if (out != in)
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@@ -321,6 +162,7 @@ static av_cold void uninit(AVFilterContext *ctx)
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}
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#define OFFSET(x) offsetof(AudioDynamicEqualizerContext, x)
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#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
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static const AVOption adynamicequalizer_options[] = {
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@@ -354,6 +196,10 @@ static const AVOption adynamicequalizer_options[] = {
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{ "disabled", 0, 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, FLAGS, "auto" },
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{ "off", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "auto" },
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{ "on", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "auto" },
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{ "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, AF, "precision" },
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{ "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "precision" },
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{ "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision" },
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{ "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision" },
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{ NULL }
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};
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@@ -383,7 +229,7 @@ const AVFilter ff_af_adynamicequalizer = {
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.uninit = uninit,
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FILTER_INPUTS(inputs),
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FILTER_OUTPUTS(outputs),
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FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
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FILTER_QUERY_FUNC(query_formats),
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.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
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AVFILTER_FLAG_SLICE_THREADS,
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.process_command = ff_filter_process_command,
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