float a;
int64_t pd;
int64_t rd;
+ int om;
float pdiff_lut_scale;
float weight_lut[WEIGHT_LUT_SIZE];
int64_t pts;
AVAudioFifo *fifo;
+ int eof_left;
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 }
};
{
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;
if (!s->fifo)
return AVERROR(ENOMEM);
+ 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;
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++) {
s->dsp.compute_cache(cache + S, f, S, K, i, i + 1);
}
- for (int j = 0; j < 2 * S; j++) {
+ for (int j = 0; j < 2 * S && !ctx->is_disabled; j++) {
const float distance = cache[j];
unsigned weight_lut_idx;
float 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,
};