[FFmpeg-cvslog] avfilter/af_acrossover: add support for float sample format
Paul B Mahol
git at videolan.org
Fri Nov 27 19:58:32 EET 2020
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Fri Nov 27 18:32:56 2020 +0100| [cf98822b66ac298e165dfd845766303d95062001] | committer: Paul B Mahol
avfilter/af_acrossover: add support for float sample format
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=cf98822b66ac298e165dfd845766303d95062001
---
libavfilter/af_acrossover.c | 245 +++++++++++++++++++++++---------------------
1 file changed, 130 insertions(+), 115 deletions(-)
diff --git a/libavfilter/af_acrossover.c b/libavfilter/af_acrossover.c
index 550d4ddbaf..3b31031c43 100644
--- a/libavfilter/af_acrossover.c
+++ b/libavfilter/af_acrossover.c
@@ -67,6 +67,8 @@ typedef struct AudioCrossoverContext {
AVFrame *input_frame;
AVFrame *frames[MAX_BANDS];
+
+ int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
} AudioCrossoverContext;
#define OFFSET(x) offsetof(AudioCrossoverContext, x)
@@ -228,58 +230,12 @@ static void calc_q_factors(int order, double *q)
q[i] = 1. / (-2. * cos(M_PI * (2. * (i + 1) + n - 1.) / (2. * n)));
}
-static int config_input(AVFilterLink *inlink)
-{
- AVFilterContext *ctx = inlink->dst;
- AudioCrossoverContext *s = ctx->priv;
- int sample_rate = inlink->sample_rate;
- double q[16];
-
- s->xover = av_calloc(inlink->channels, sizeof(*s->xover));
- if (!s->xover)
- return AVERROR(ENOMEM);
-
- 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 ch = 0; ch < inlink->channels; ch++) {
- for (int band = 0; band <= s->nb_splits; band++) {
- if (s->first_order) {
- set_lp(&s->xover[ch].lp[band][0], s->splits[band], 0.5, sample_rate);
- set_hp(&s->xover[ch].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->xover[ch].lp[band][n], s->splits[band], q[idx], sample_rate);
- set_hp(&s->xover[ch].hp[band][n], s->splits[band], q[idx], sample_rate);
- }
-
- for (int x = 0; x <= s->nb_splits && s->first_order; x++)
- set_ap1(&s->xover[ch].ap[x][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);
-
- for (int x = 0; x <= s->nb_splits; x++)
- set_ap(&s->xover[ch].ap[x][band][n], s->splits[band], q[idx], sample_rate);
- }
- }
- }
-
- return 0;
-}
-
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats;
AVFilterChannelLayouts *layouts;
static const enum AVSampleFormat sample_fmts[] = {
- AV_SAMPLE_FMT_DBLP,
+ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
int ret;
@@ -304,90 +260,149 @@ static int query_formats(AVFilterContext *ctx)
return ff_set_common_samplerates(ctx, formats);
}
-static void biquad_process(BiquadContext *b,
- double *dst, const double *src,
- int nb_samples)
-{
- const double b0 = b->b0;
- const double b1 = b->b1;
- const double b2 = b->b2;
- const double a1 = b->a1;
- const double a2 = b->a2;
- double z1 = b->z1;
- double z2 = b->z2;
-
- for (int n = 0; n < nb_samples; n++) {
- const double in = src[n];
- double out;
-
- out = in * b0 + z1;
- z1 = b1 * in + z2 + a1 * out;
- z2 = b2 * in + a2 * out;
- dst[n] = out;
- }
+#define BIQUAD_PROCESS(name, type) \
+static void biquad_process_## name(BiquadContext *b, \
+ type *dst, const type *src, \
+ int nb_samples) \
+{ \
+ const type b0 = b->b0; \
+ const type b1 = b->b1; \
+ const type b2 = b->b2; \
+ const type a1 = b->a1; \
+ const type a2 = b->a2; \
+ type z1 = b->z1; \
+ type z2 = b->z2; \
+ \
+ for (int n = 0; n < nb_samples; n++) { \
+ 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->z1 = z1; \
+ b->z2 = z2; \
+}
- b->z1 = z1;
- b->z2 = z2;
+BIQUAD_PROCESS(fltp, float)
+BIQUAD_PROCESS(dblp, double)
+
+#define XOVER_PROCESS(name, type, one) \
+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; \
+ \
+ for (int ch = start; ch < end; ch++) { \
+ CrossoverChannel *xover = &s->xover[ch]; \
+ \
+ for (int band = 0; band < ctx->nb_outputs; band++) { \
+ for (int f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) { \
+ const type *src = band == 0 ? (const type *)in->extended_data[ch] : (const type *)frames[band]->extended_data[ch]; \
+ type *dst = (type *)frames[band + 1]->extended_data[ch]; \
+ const type *hsrc = f == 0 ? src : dst; \
+ BiquadContext *hp = &xover->hp[band][f]; \
+ \
+ biquad_process_## name(hp, dst, hsrc, nb_samples); \
+ } \
+ \
+ for (int f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) { \
+ const type *src = band == 0 ? (const type *)in->extended_data[ch] : (const type *)frames[band]->extended_data[ch]; \
+ type *dst = (type *)frames[band]->extended_data[ch]; \
+ const type *lsrc = f == 0 ? src : dst; \
+ BiquadContext *lp = &xover->lp[band][f]; \
+ \
+ biquad_process_## name(lp, dst, lsrc, nb_samples); \
+ } \
+ \
+ for (int aband = band + 1; aband < ctx->nb_outputs; aband++) { \
+ if (s->first_order) { \
+ const type *src = (const type *)frames[band]->extended_data[ch]; \
+ type *dst = (type *)frames[band]->extended_data[ch]; \
+ BiquadContext *ap = &xover->ap[band][aband][0]; \
+ \
+ biquad_process_## name(ap, dst, src, nb_samples); \
+ } \
+ \
+ for (int f = s->first_order; f < s->ap_filter_count; f++) { \
+ const type *src = (const type *)frames[band]->extended_data[ch]; \
+ type *dst = (type *)frames[band]->extended_data[ch]; \
+ BiquadContext *ap = &xover->ap[band][aband][f]; \
+ \
+ biquad_process_## name(ap, dst, src, nb_samples); \
+ } \
+ } \
+ } \
+ \
+ for (int band = 0; band < ctx->nb_outputs && s->first_order; band++) { \
+ if (band & 1) { \
+ type *dst = (type *)frames[band]->extended_data[ch]; \
+ \
+ for (int n = 0; n < nb_samples; n++) \
+ dst[n] *= -one; \
+ } \
+ } \
+ } \
+ \
+ return 0; \
}
-static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
+XOVER_PROCESS(fltp, float, 1.f)
+XOVER_PROCESS(dblp, double, 1.0)
+
+static int config_input(AVFilterLink *inlink)
{
+ AVFilterContext *ctx = inlink->dst;
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;
+ int sample_rate = inlink->sample_rate;
+ double q[16];
- for (int ch = start; ch < end; ch++) {
- CrossoverChannel *xover = &s->xover[ch];
+ s->xover = av_calloc(inlink->channels, sizeof(*s->xover));
+ if (!s->xover)
+ return AVERROR(ENOMEM);
- for (int band = 0; band < ctx->nb_outputs; band++) {
- for (int f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) {
- const double *src = band == 0 ? (const double *)in->extended_data[ch] : (const double *)frames[band]->extended_data[ch];
- double *dst = (double *)frames[band + 1]->extended_data[ch];
- const double *hsrc = f == 0 ? src : dst;
- BiquadContext *hp = &xover->hp[band][f];
+ 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);
- biquad_process(hp, dst, hsrc, nb_samples);
+ for (int ch = 0; ch < inlink->channels; ch++) {
+ for (int band = 0; band <= s->nb_splits; band++) {
+ if (s->first_order) {
+ set_lp(&s->xover[ch].lp[band][0], s->splits[band], 0.5, sample_rate);
+ set_hp(&s->xover[ch].hp[band][0], s->splits[band], 0.5, sample_rate);
}
- for (int f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) {
- const double *src = band == 0 ? (const double *)in->extended_data[ch] : (const double *)frames[band]->extended_data[ch];
- double *dst = (double *)frames[band]->extended_data[ch];
- const double *lsrc = f == 0 ? src : dst;
- BiquadContext *lp = &xover->lp[band][f];
+ 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;
- biquad_process(lp, dst, lsrc, nb_samples);
+ set_lp(&s->xover[ch].lp[band][n], s->splits[band], q[idx], sample_rate);
+ set_hp(&s->xover[ch].hp[band][n], s->splits[band], q[idx], sample_rate);
}
- for (int aband = band + 1; aband < ctx->nb_outputs; aband++) {
- if (s->first_order) {
- const double *src = (const double *)frames[band]->extended_data[ch];
- double *dst = (double *)frames[band]->extended_data[ch];
- BiquadContext *ap = &xover->ap[band][aband][0];
-
- biquad_process(ap, dst, src, nb_samples);
- }
+ for (int x = 0; x <= s->nb_splits && s->first_order; x++)
+ set_ap1(&s->xover[ch].ap[x][band][0], s->splits[band], sample_rate);
- for (int f = s->first_order; f < s->ap_filter_count; f++) {
- const double *src = (const double *)frames[band]->extended_data[ch];
- double *dst = (double *)frames[band]->extended_data[ch];
- BiquadContext *ap = &xover->ap[band][aband][f];
+ 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);
- biquad_process(ap, dst, src, nb_samples);
- }
+ for (int x = 0; x <= s->nb_splits; x++)
+ set_ap(&s->xover[ch].ap[x][band][n], s->splits[band], q[idx], sample_rate);
}
}
+ }
- for (int band = 0; band < ctx->nb_outputs && s->first_order; band++) {
- if (band & 1) {
- double *dst = (double *)frames[band]->extended_data[ch];
-
- for (int n = 0; n < nb_samples; n++)
- dst[n] *= -1.;
- }
- }
+ 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;
}
return 0;
@@ -415,8 +430,8 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
goto fail;
s->input_frame = in;
- ctx->internal->execute(ctx, filter_channels, NULL, NULL, FFMIN(inlink->channels,
- ff_filter_get_nb_threads(ctx)));
+ 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]);
More information about the ffmpeg-cvslog
mailing list