#include "libavutil/avassert.h"
#include "libavutil/audio_fifo.h"
+#include "libavutil/avstring.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "audio.h"
#include "af_anlmdndsp.h"
-#define MAX_DIFF 11.f
#define WEIGHT_LUT_NBITS 20
#define WEIGHT_LUT_SIZE (1<<WEIGHT_LUT_NBITS)
float a;
int64_t pd;
int64_t rd;
+ float m;
+ int om;
float pdiff_lut_scale;
float weight_lut[WEIGHT_LUT_SIZE];
AudioNLMDNDSPContext dsp;
} AudioNLMeansContext;
+enum OutModes {
+ IN_MODE,
+ OUT_MODE,
+ NOISE_MODE,
+ NB_MODES
+};
+
#define OFFSET(x) offsetof(AudioNLMeansContext, x)
#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+#define AFT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption anlmdn_options[] = {
- { "s", "set denoising strength", OFFSET(a), AV_OPT_TYPE_FLOAT, {.dbl=0.00001},0.00001, 10, AF },
+ { "s", "set denoising strength", OFFSET(a), AV_OPT_TYPE_FLOAT, {.dbl=0.00001},0.00001, 10, AFT },
{ "p", "set patch duration", OFFSET(pd), AV_OPT_TYPE_DURATION, {.i64=2000}, 1000, 100000, AF },
{ "r", "set research duration", OFFSET(rd), AV_OPT_TYPE_DURATION, {.i64=6000}, 2000, 300000, AF },
+ { "o", "set output mode", OFFSET(om), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_MODES-1, AFT, "mode" },
+ { "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AFT, "mode" },
+ { "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AFT, "mode" },
+ { "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE},0, 0, AFT, "mode" },
+ { "m", "set smooth factor", OFFSET(m), AV_OPT_TYPE_FLOAT, {.dbl=11.}, 1, 15, AF },
{ NULL }
};
if (ret < 0)
return ret;
- s->pdiff_lut_scale = 1.f / MAX_DIFF * WEIGHT_LUT_SIZE;
+ s->pdiff_lut_scale = 1.f / s->m * WEIGHT_LUT_SIZE;
for (int i = 0; i < WEIGHT_LUT_SIZE; i++) {
float w = -i / s->pdiff_lut_scale;
AVFrame *out = arg;
const int S = s->S;
const int K = s->K;
+ const int om = s->om;
const float *f = (const float *)(s->in->extended_data[ch]) + K;
float *cache = (float *)s->cache->extended_data[ch];
const float sw = (65536.f / (4 * K + 2)) / sqrtf(s->a);
float *dst = (float *)out->extended_data[ch] + s->offset;
+ const float smooth = s->m;
for (int i = S; i < s->H + S; i++) {
float P = 0.f, Q = 0.f;
unsigned weight_lut_idx;
float w;
- av_assert2(distance >= 0.f);
+ if (distance < 0.f) {
+ cache[j] = 0.f;
+ continue;
+ }
w = distance * sw;
- if (w >= MAX_DIFF)
+ if (w >= smooth)
continue;
weight_lut_idx = w * s->pdiff_lut_scale;
av_assert2(weight_lut_idx < WEIGHT_LUT_SIZE);
P += f[i];
Q += 1;
- dst[i - S] = P / Q;
+ switch (om) {
+ case IN_MODE: dst[i - S] = f[i]; break;
+ case OUT_MODE: dst[i - S] = P / Q; break;
+ case NOISE_MODE: dst[i - S] = f[i] - (P / Q); break;
+ }
}
return 0;
out->nb_samples = FFMIN(s->eof_left, s->offset);
s->eof_left -= out->nb_samples;
}
- s->pts += s->offset;
+ s->pts += av_rescale_q(s->offset, (AVRational){1, outlink->sample_rate}, outlink->time_base);
return ff_filter_frame(outlink, out);
}
if (s->eof_left < 0)
s->eof_left = av_audio_fifo_size(s->fifo) - (s->S + s->K);
- if (s->eof_left < 0)
+ if (s->eof_left <= 0)
return AVERROR_EOF;
in = ff_get_audio_buffer(outlink, s->H);
if (!in)
.uninit = uninit,
.inputs = inputs,
.outputs = outputs,
+ .process_command = ff_filter_process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,
};