#include "libavutil/avassert.h"
#include "libavutil/audio_fifo.h"
+#include "libavutil/avstring.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "audio.h"
#include "formats.h"
+#include "af_anlmdndsp.h"
+
+#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;
+ float m;
+ 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
+#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=1}, 1, 9999, 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 },
+ { "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, AFT },
+ { "r", "set research duration", OFFSET(rd), AV_OPT_TYPE_DURATION, {.i64=6000}, 2000, 300000, AFT },
+ { "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, AFT },
{ 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 int config_output(AVFilterLink *outlink)
+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_filter(AVFilterContext *ctx)
{
- AVFilterContext *ctx = outlink->src;
AudioNLMeansContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+ int newK, newS, newH, newN;
+ AVFrame *new_in, *new_cache;
- s->K = av_rescale(s->pd, outlink->sample_rate, AV_TIME_BASE);
- s->S = av_rescale(s->rd, outlink->sample_rate, AV_TIME_BASE);
+ newK = av_rescale(s->pd, outlink->sample_rate, AV_TIME_BASE);
+ newS = av_rescale(s->rd, outlink->sample_rate, AV_TIME_BASE);
- s->pts = AV_NOPTS_VALUE;
- s->H = s->K * 2 + 1;
- s->N = s->H + (s->K + s->S) * 2;
+ newH = newK * 2 + 1;
+ newN = newH + (newK + newS) * 2;
- av_frame_free(&s->in);
- av_frame_free(&s->cache);
- s->in = ff_get_audio_buffer(outlink, s->N);
- if (!s->in)
- return AVERROR(ENOMEM);
+ av_log(ctx, AV_LOG_DEBUG, "K:%d S:%d H:%d N:%d\n", newK, newS, newH, newN);
- s->cache = ff_get_audio_buffer(outlink, s->S * 2);
+ if (!s->cache || s->cache->nb_samples < newS * 2) {
+ new_cache = ff_get_audio_buffer(outlink, newS * 2);
+ if (new_cache) {
+ av_frame_free(&s->cache);
+ s->cache = new_cache;
+ } else {
+ return AVERROR(ENOMEM);
+ }
+ }
if (!s->cache)
return AVERROR(ENOMEM);
+ 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;
+
+ s->weight_lut[i] = expf(w);
+ }
+
+ if (!s->in || s->in->nb_samples < newN) {
+ new_in = ff_get_audio_buffer(outlink, newN);
+ if (new_in) {
+ av_frame_free(&s->in);
+ s->in = new_in;
+ } else {
+ return AVERROR(ENOMEM);
+ }
+ }
+ if (!s->in)
+ return AVERROR(ENOMEM);
+
+ s->K = newK;
+ s->S = newS;
+ s->H = newH;
+ s->N = newN;
+
+ return 0;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AudioNLMeansContext *s = ctx->priv;
+ int ret;
+
+ s->eof_left = -1;
+ s->pts = AV_NOPTS_VALUE;
+
+ ret = config_filter(ctx);
+ if (ret < 0)
+ return ret;
+
s->fifo = av_audio_fifo_alloc(outlink->format, outlink->channels, 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;
+
+ 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;
+ const float smooth = s->m;
for (int i = S; i < s->H + S; i++) {
float P = 0.f, Q = 0.f;
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)
+ if (distance < 0.f) {
+ cache[j] = 0.f;
+ continue;
+ }
+ w = distance * sw;
+ if (w >= smooth)
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;
- s->pts += s->offset;
+ if (s->eof_left >= 0) {
+ out->nb_samples = FFMIN(s->eof_left, s->offset);
+ s->eof_left -= out->nb_samples;
+ }
+ s->pts += av_rescale_q(s->offset, (AVRational){1, outlink->sample_rate}, outlink->time_base);
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 int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
+ char *res, int res_len, int flags)
+{
+ int ret;
+
+ ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
+ if (ret < 0)
+ return ret;
+
+ ret = config_filter(ctx);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
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 }
};
-AVFilter ff_af_anlmdn = {
+const AVFilter ff_af_anlmdn = {
.name = "anlmdn",
.description = NULL_IF_CONFIG_SMALL("Reduce broadband noise from stream using Non-Local Means."),
.query_formats = query_formats,
.uninit = uninit,
.inputs = inputs,
.outputs = outputs,
- .flags = AVFILTER_FLAG_SLICE_THREADS,
+ .process_command = process_command,
+ .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
+ AVFILTER_FLAG_SLICE_THREADS,
};
projects / ffmpeg / blobdiff