#include "audio.h"
#include "formats.h"
+#include "af_anlmdndsp.h"
+
+#define MAX_DIFF 11.f
+#define WEIGHT_LUT_NBITS 20
+#define WEIGHT_LUT_SIZE (1<<WEIGHT_LUT_NBITS)
+
#define SQR(x) ((x) * (x))
typedef struct AudioNLMeansContext {
float a;
int64_t pd;
int64_t rd;
+ int om;
+
+ float pdiff_lut_scale;
+ float weight_lut[WEIGHT_LUT_SIZE];
int K;
int S;
int64_t pts;
AVAudioFifo *fifo;
+ int eof_left;
- float (*compute_distance)(const float *f1, const float *f2, int K);
+ 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
static const AVOption anlmdn_options[] = {
- { "s", "set denoising strength", OFFSET(a), AV_OPT_TYPE_FLOAT, {.dbl=1}, 1, 9999, AF },
+ { "s", "set denoising strength", OFFSET(a), AV_OPT_TYPE_FLOAT, {.dbl=0.00001},0.00001, 10, AF },
{ "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, AF, "mode" },
+ { "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AF, "mode" },
+ { "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AF, "mode" },
+ { "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE},0, 0, AF, "mode" },
{ NULL }
};
return ff_set_common_samplerates(ctx, formats);
}
-static float compute_distance_ssd(const float *f1, const float *f2, int K)
+static float compute_distance_ssd_c(const float *f1, const float *f2, ptrdiff_t K)
{
float distance = 0.;
return distance;
}
+static void compute_cache_c(float *cache, const float *f,
+ ptrdiff_t S, ptrdiff_t K,
+ ptrdiff_t i, ptrdiff_t jj)
+{
+ int v = 0;
+
+ for (int j = jj; j < jj + S; j++, v++)
+ cache[v] += -SQR(f[i - K - 1] - f[j - K - 1]) + SQR(f[i + K] - f[j + K]);
+}
+
+void ff_anlmdn_init(AudioNLMDNDSPContext *dsp)
+{
+ dsp->compute_distance_ssd = compute_distance_ssd_c;
+ dsp->compute_cache = compute_cache_c;
+
+ if (ARCH_X86)
+ ff_anlmdn_init_x86(dsp);
+}
+
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioNLMeansContext *s = ctx->priv;
+ int ret;
s->K = av_rescale(s->pd, outlink->sample_rate, AV_TIME_BASE);
s->S = av_rescale(s->rd, outlink->sample_rate, AV_TIME_BASE);
+ s->eof_left = -1;
s->pts = AV_NOPTS_VALUE;
s->H = s->K * 2 + 1;
s->N = s->H + (s->K + s->S) * 2;
+ av_log(ctx, AV_LOG_DEBUG, "K:%d S:%d H:%d N:%d\n", s->K, s->S, s->H, s->N);
+
av_frame_free(&s->in);
av_frame_free(&s->cache);
s->in = ff_get_audio_buffer(outlink, s->N);
if (!s->fifo)
return AVERROR(ENOMEM);
- s->compute_distance = compute_distance_ssd;
+ ret = av_audio_fifo_write(s->fifo, (void **)s->in->extended_data, s->K + s->S);
+ if (ret < 0)
+ return ret;
+
+ s->pdiff_lut_scale = 1.f / MAX_DIFF * WEIGHT_LUT_SIZE;
+ for (int i = 0; i < WEIGHT_LUT_SIZE; i++) {
+ float w = -i / s->pdiff_lut_scale;
+
+ s->weight_lut[i] = expf(w);
+ }
+
+ ff_anlmdn_init(&s->dsp);
return 0;
}
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 = 32768.f / s->a;
+ const float sw = (65536.f / (4 * K + 2)) / sqrtf(s->a);
float *dst = (float *)out->extended_data[ch] + s->offset;
for (int i = S; i < s->H + S; i++) {
for (int j = i - S; j <= i + S; j++) {
if (i == j)
continue;
- cache[v++] = s->compute_distance(f + i, f + j, K);
+ cache[v++] = s->dsp.compute_distance_ssd(f + i, f + j, K);
}
} else {
- for (int j = i - S; j < i; j++, v++)
- cache[v] = cache[v] - SQR(f[i - K - 1] - f[j - K - 1]) + SQR(f[i + K] - f[j + K]);
-
- for (int j = i + 1; j <= i + S; j++, v++)
- cache[v] = cache[v] - SQR(f[i - K - 1] - f[j - K - 1]) + SQR(f[i + K] - f[j + K]);
+ s->dsp.compute_cache(cache, f, S, K, i, i - S);
+ s->dsp.compute_cache(cache + S, f, S, K, i, i + 1);
}
- for (int j = 0; j < v; j++) {
+ for (int j = 0; j < 2 * S && !ctx->is_disabled; j++) {
const float distance = cache[j];
+ unsigned weight_lut_idx;
float w;
- av_assert0(distance >= 0.f);
- w = -distance * sw;
- if (w < -11.f)
+ av_assert2(distance >= 0.f);
+ w = distance * sw;
+ if (w >= MAX_DIFF)
continue;
- w = expf(w);
+ weight_lut_idx = w * s->pdiff_lut_scale;
+ av_assert2(weight_lut_idx < WEIGHT_LUT_SIZE);
+ w = s->weight_lut[weight_lut_idx];
P += w * f[i - S + j + (j >= S)];
Q += w;
}
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;
if (out) {
out->pts = s->pts;
out->nb_samples = s->offset;
+ if (s->eof_left >= 0) {
+ out->nb_samples = FFMIN(s->eof_left, s->offset);
+ s->eof_left -= out->nb_samples;
+ }
s->pts += s->offset;
return ff_filter_frame(outlink, out);
return ret;
}
+static int request_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AudioNLMeansContext *s = ctx->priv;
+ int ret;
+
+ ret = ff_request_frame(ctx->inputs[0]);
+
+ if (ret == AVERROR_EOF && s->eof_left != 0) {
+ AVFrame *in;
+
+ if (s->eof_left < 0)
+ s->eof_left = av_audio_fifo_size(s->fifo) - (s->S + s->K);
+ if (s->eof_left < 0)
+ return AVERROR_EOF;
+ in = ff_get_audio_buffer(outlink, s->H);
+ if (!in)
+ return AVERROR(ENOMEM);
+
+ return filter_frame(ctx->inputs[0], in);
+ }
+
+ return ret;
+}
+
static av_cold void uninit(AVFilterContext *ctx)
{
AudioNLMeansContext *s = ctx->priv;
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
+ .request_frame = request_frame,
},
{ NULL }
};
.uninit = uninit,
.inputs = inputs,
.outputs = outputs,
- .flags = AVFILTER_FLAG_SLICE_THREADS,
+ .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
+ AVFILTER_FLAG_SLICE_THREADS,
};
projects / ffmpeg / blobdiff