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authorPaul B Mahol <onemda@gmail.com>2018-05-14 10:07:44 +0200
committerPaul B Mahol <onemda@gmail.com>2018-05-16 12:20:38 +0200
commit4e816b549142d11f716b8d23e4f68b622298f28f (patch)
tree3f988e7f6979b1a10c9686270478de55f8c0cb64 /libavfilter/af_aderivative.c
parent64f59a21b39b380eaadc6d5f5dc721965fb785e4 (diff)
avfilter: add aderivative and aintegral filter
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Diffstat (limited to 'libavfilter/af_aderivative.c')
-rw-r--r--libavfilter/af_aderivative.c207
1 files changed, 207 insertions, 0 deletions
diff --git a/libavfilter/af_aderivative.c b/libavfilter/af_aderivative.c
new file mode 100644
index 0000000000..a591515cbf
--- /dev/null
+++ b/libavfilter/af_aderivative.c
@@ -0,0 +1,207 @@
+/*
+ * 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 "audio.h"
+#include "avfilter.h"
+#include "internal.h"
+
+typedef struct ADerivativeContext {
+ const AVClass *class;
+ AVFrame *prev;
+ void (*filter)(void **dst, void **prv, const void **src,
+ int nb_samples, int channels);
+} ADerivativeContext;
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats = NULL;
+ AVFilterChannelLayouts *layouts = NULL;
+ static const enum AVSampleFormat derivative_sample_fmts[] = {
+ AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_DBLP,
+ AV_SAMPLE_FMT_NONE
+ };
+ static const enum AVSampleFormat integral_sample_fmts[] = {
+ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
+ AV_SAMPLE_FMT_NONE
+ };
+ int ret;
+
+ formats = ff_make_format_list(strcmp(ctx->filter->name, "aintegral") ?
+ derivative_sample_fmts : integral_sample_fmts);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_formats(ctx, formats);
+ if (ret < 0)
+ return 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_all_samplerates();
+ return ff_set_common_samplerates(ctx, formats);
+}
+
+#define DERIVATIVE(name, type) \
+static void aderivative_## name ##p(void **d, void **p, const void **s, \
+ int nb_samples, int channels) \
+{ \
+ int n, c; \
+ \
+ for (c = 0; c < channels; c++) { \
+ const type *src = s[c]; \
+ type *dst = d[c]; \
+ type *prv = p[c]; \
+ \
+ for (n = 0; n < nb_samples; n++) { \
+ const type current = src[n]; \
+ \
+ dst[n] = current - prv[0]; \
+ prv[0] = current; \
+ } \
+ } \
+}
+
+DERIVATIVE(flt, float)
+DERIVATIVE(dbl, double)
+DERIVATIVE(s16, int16_t)
+DERIVATIVE(s32, int32_t)
+
+#define INTEGRAL(name, type) \
+static void aintegral_## name ##p(void **d, void **p, const void **s, \
+ int nb_samples, int channels) \
+{ \
+ int n, c; \
+ \
+ for (c = 0; c < channels; c++) { \
+ const type *src = s[c]; \
+ type *dst = d[c]; \
+ type *prv = p[c]; \
+ \
+ for (n = 0; n < nb_samples; n++) { \
+ const type current = src[n]; \
+ \
+ dst[n] = current + prv[0]; \
+ prv[0] = dst[n]; \
+ } \
+ } \
+}
+
+INTEGRAL(flt, float)
+INTEGRAL(dbl, double)
+
+static int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ ADerivativeContext *s = ctx->priv;
+
+ switch (inlink->format) {
+ case AV_SAMPLE_FMT_FLTP: s->filter = aderivative_fltp; break;
+ case AV_SAMPLE_FMT_DBLP: s->filter = aderivative_dblp; break;
+ case AV_SAMPLE_FMT_S32P: s->filter = aderivative_s32p; break;
+ case AV_SAMPLE_FMT_S16P: s->filter = aderivative_s16p; break;
+ }
+
+ if (strcmp(ctx->filter->name, "aintegral"))
+ return 0;
+
+ switch (inlink->format) {
+ case AV_SAMPLE_FMT_FLTP: s->filter = aintegral_fltp; break;
+ case AV_SAMPLE_FMT_DBLP: s->filter = aintegral_dblp; break;
+ }
+
+ return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ ADerivativeContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+ AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
+
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+ av_frame_copy_props(out, in);
+
+ if (!s->prev) {
+ s->prev = ff_get_audio_buffer(inlink, 1);
+ if (!s->prev) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+ }
+
+ s->filter((void **)out->extended_data, (void **)s->prev->extended_data, (const void **)in->extended_data,
+ in->nb_samples, in->channels);
+
+ av_frame_free(&in);
+ return ff_filter_frame(outlink, out);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ ADerivativeContext *s = ctx->priv;
+
+ av_frame_free(&s->prev);
+}
+
+static const AVFilterPad aderivative_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ .config_props = config_input,
+ },
+ { NULL }
+};
+
+static const AVFilterPad aderivative_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_aderivative = {
+ .name = "aderivative",
+ .description = NULL_IF_CONFIG_SMALL("Compute derivative of input audio."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(ADerivativeContext),
+ .uninit = uninit,
+ .inputs = aderivative_inputs,
+ .outputs = aderivative_outputs,
+};
+
+AVFilter ff_af_aintegral = {
+ .name = "aintegral",
+ .description = NULL_IF_CONFIG_SMALL("Compute integral of input audio."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(ADerivativeContext),
+ .uninit = uninit,
+ .inputs = aderivative_inputs,
+ .outputs = aderivative_outputs,
+};