#include "audio.h"
#include "formats.h"
-#define NUMTAPS 64
+#define NUMTAPS 32
-static const int8_t filt[NUMTAPS] = {
+static const int8_t filt[NUMTAPS * 2] = {
/* 30° 330° */
4, -6, /* 32 tap stereo FIR filter. */
4, -11, /* One side filters as if the */
4, 0};
typedef struct EarwaxContext {
- int16_t taps[NUMTAPS * 2];
+ int16_t filter[2][NUMTAPS];
+ int16_t taps[4][NUMTAPS * 2];
+
+ AVFrame *frame[2];
} EarwaxContext;
static int query_formats(AVFilterContext *ctx)
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layout = NULL;
- if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_S16 )) < 0 ||
+ if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_S16P )) < 0 ||
(ret = ff_set_common_formats (ctx , formats )) < 0 ||
(ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO )) < 0 ||
(ret = ff_set_common_channel_layouts (ctx , layout )) < 0 ||
}
//FIXME: replace with DSPContext.scalarproduct_int16
-static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin, int16_t *out)
+static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin,
+ const int16_t *filt, int16_t *out)
{
int32_t sample;
int16_t j;
sample = 0;
for (j = 0; j < NUMTAPS; j++)
sample += in[j] * filt[j];
- *out = av_clip_int16(sample >> 6);
+ *out = av_clip_int16(sample >> 7);
out++;
in++;
}
return out;
}
-static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
+static int config_input(AVFilterLink *inlink)
{
- AVFilterLink *outlink = inlink->dst->outputs[0];
- int16_t *taps, *endin, *in, *out;
- AVFrame *outsamples = ff_get_audio_buffer(outlink, insamples->nb_samples);
- int len;
+ EarwaxContext *s = inlink->dst->priv;
- if (!outsamples) {
- av_frame_free(&insamples);
- return AVERROR(ENOMEM);
+ for (int i = 0; i < NUMTAPS; i++) {
+ s->filter[0][i] = filt[i * 2];
+ s->filter[1][i] = filt[i * 2 + 1];
}
- av_frame_copy_props(outsamples, insamples);
- taps = ((EarwaxContext *)inlink->dst->priv)->taps;
- out = (int16_t *)outsamples->data[0];
- in = (int16_t *)insamples ->data[0];
+ return 0;
+}
+
+static void convolve(AVFilterContext *ctx, AVFrame *in,
+ int input_ch, int output_ch,
+ int filter_ch, int tap_ch)
+{
+ EarwaxContext *s = ctx->priv;
+ int16_t *taps, *endin, *dst, *src;
+ int len;
+
+ taps = s->taps[tap_ch];
+ dst = (int16_t *)s->frame[input_ch]->data[output_ch];
+ src = (int16_t *)in->data[input_ch];
- len = FFMIN(NUMTAPS, 2*insamples->nb_samples);
+ len = FFMIN(NUMTAPS, in->nb_samples);
// copy part of new input and process with saved input
- memcpy(taps+NUMTAPS, in, len * sizeof(*taps));
- out = scalarproduct(taps, taps + len, out);
+ memcpy(taps+NUMTAPS, src, len * sizeof(*taps));
+ dst = scalarproduct(taps, taps + len, s->filter[filter_ch], dst);
// process current input
- if (2*insamples->nb_samples >= NUMTAPS ){
- endin = in + insamples->nb_samples * 2 - NUMTAPS;
- scalarproduct(in, endin, out);
+ if (in->nb_samples >= NUMTAPS) {
+ endin = src + in->nb_samples - NUMTAPS;
+ scalarproduct(src, endin, s->filter[filter_ch], dst);
// save part of input for next round
memcpy(taps, endin, NUMTAPS * sizeof(*taps));
- } else
- memmove(taps, taps + 2*insamples->nb_samples, NUMTAPS * sizeof(*taps));
+ } else {
+ memmove(taps, taps + in->nb_samples, NUMTAPS * sizeof(*taps));
+ }
+}
+
+static void mix(AVFilterContext *ctx, AVFrame *out,
+ int output_ch, int f0, int f1, int i0, int i1)
+{
+ EarwaxContext *s = ctx->priv;
+ const int16_t *srcl = (const int16_t *)s->frame[f0]->data[i0];
+ const int16_t *srcr = (const int16_t *)s->frame[f1]->data[i1];
+ int16_t *dst = (int16_t *)out->data[output_ch];
+
+ for (int n = 0; n < out->nb_samples; n++)
+ dst[n] = av_clip_int16(srcl[n] + srcr[n]);
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ EarwaxContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+ AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
+
+ for (int ch = 0; ch < 2; ch++) {
+ if (!s->frame[ch] || s->frame[ch]->nb_samples < in->nb_samples) {
+ av_frame_free(&s->frame[ch]);
+ s->frame[ch] = ff_get_audio_buffer(outlink, in->nb_samples);
+ if (!s->frame[ch]) {
+ av_frame_free(&in);
+ av_frame_free(&out);
+ return AVERROR(ENOMEM);
+ }
+ }
+ }
+
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+ av_frame_copy_props(out, in);
+
+ convolve(ctx, in, 0, 0, 0, 0);
+ convolve(ctx, in, 0, 1, 1, 1);
+ convolve(ctx, in, 1, 0, 0, 2);
+ convolve(ctx, in, 1, 1, 1, 3);
+
+ mix(ctx, out, 0, 0, 1, 1, 0);
+ mix(ctx, out, 1, 0, 1, 0, 1);
+
+ av_frame_free(&in);
+ return ff_filter_frame(outlink, out);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ EarwaxContext *s = ctx->priv;
- av_frame_free(&insamples);
- return ff_filter_frame(outlink, outsamples);
+ av_frame_free(&s->frame[0]);
+ av_frame_free(&s->frame[1]);
}
static const AVFilterPad earwax_inputs[] = {
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
+ .config_props = config_input,
},
{ NULL }
};
{ NULL }
};
-AVFilter ff_af_earwax = {
+const AVFilter ff_af_earwax = {
.name = "earwax",
.description = NULL_IF_CONFIG_SMALL("Widen the stereo image."),
.query_formats = query_formats,
.priv_size = sizeof(EarwaxContext),
+ .uninit = uninit,
.inputs = earwax_inputs,
.outputs = earwax_outputs,
};
projects / ffmpeg / blobdiff