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ffmpeg/libavfilter/af_anlmdn.c

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/*
* Copyright (c) 2019 Paul B Mahol
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <float.h>
#include "libavutil/avassert.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "audio.h"
#include "filters.h"
#include "af_anlmdndsp.h"
#define WEIGHT_LUT_NBITS 20
#define WEIGHT_LUT_SIZE (1<<WEIGHT_LUT_NBITS)
typedef struct AudioNLMeansContext {
const AVClass *class;
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;
int N;
int H;
AVFrame *in;
AVFrame *cache;
AVFrame *window;
AudioNLMDNDSPContext dsp;
} AudioNLMeansContext;
enum OutModes {
IN_MODE,
OUT_MODE,
NOISE_MODE,
NB_MODES
};
#define OFFSET(x) offsetof(AudioNLMeansContext, x)
#define AFT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption anlmdn_options[] = {
{ "strength", "set denoising strength", OFFSET(a), AV_OPT_TYPE_FLOAT, {.dbl=0.00001},0.00001, 10000, AFT },
{ "s", "set denoising strength", OFFSET(a), AV_OPT_TYPE_FLOAT, {.dbl=0.00001},0.00001, 10000, AFT },
{ "patch", "set patch duration", OFFSET(pd), AV_OPT_TYPE_DURATION, {.i64=2000}, 1000, 100000, AFT },
{ "p", "set patch duration", OFFSET(pd), AV_OPT_TYPE_DURATION, {.i64=2000}, 1000, 100000, AFT },
{ "research", "set research duration", OFFSET(rd), AV_OPT_TYPE_DURATION, {.i64=6000}, 2000, 300000, AFT },
{ "r", "set research duration", OFFSET(rd), AV_OPT_TYPE_DURATION, {.i64=6000}, 2000, 300000, AFT },
{ "output", "set output mode", OFFSET(om), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_MODES-1, AFT, .unit = "mode" },
{ "o", "set output mode", OFFSET(om), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_MODES-1, AFT, .unit = "mode" },
{ "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AFT, .unit = "mode" },
{ "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AFT, .unit = "mode" },
{ "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE},0, 0, AFT, .unit = "mode" },
{ "smooth", "set smooth factor", OFFSET(m), AV_OPT_TYPE_FLOAT, {.dbl=11.}, 1, 1000, AFT },
{ "m", "set smooth factor", OFFSET(m), AV_OPT_TYPE_FLOAT, {.dbl=11.}, 1, 1000, AFT },
{ NULL }
};
AVFILTER_DEFINE_CLASS(anlmdn);
static inline float sqrdiff(float x, float y)
{
const float diff = x - y;
return diff * diff;
}
static float compute_distance_ssd_c(const float *f1, const float *f2, ptrdiff_t K)
{
float distance = 0.;
for (int k = -K; k <= K; k++)
distance += sqrdiff(f1[k], f2[k]);
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] += -sqrdiff(f[i - K - 1], f[j - K - 1]) + sqrdiff(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);
#endif
}
static int config_filter(AVFilterContext *ctx)
{
AudioNLMeansContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
int newK, newS, newH, newN;
newK = av_rescale(s->pd, outlink->sample_rate, AV_TIME_BASE);
newS = av_rescale(s->rd, outlink->sample_rate, AV_TIME_BASE);
newH = newK * 2 + 1;
newN = newH + (newK + newS) * 2;
av_log(ctx, AV_LOG_DEBUG, "K:%d S:%d H:%d N:%d\n", newK, newS, newH, newN);
if (!s->cache || s->cache->nb_samples < newS * 2) {
AVFrame *new_cache = ff_get_audio_buffer(outlink, newS * 2);
if (new_cache) {
if (s->cache)
av_samples_copy(new_cache->extended_data, s->cache->extended_data, 0, 0,
s->cache->nb_samples, new_cache->ch_layout.nb_channels, new_cache->format);
av_frame_free(&s->cache);
s->cache = new_cache;
} else {
return AVERROR(ENOMEM);
}
}
if (!s->cache)
return AVERROR(ENOMEM);
if (!s->window || s->window->nb_samples < newN) {
AVFrame *new_window = ff_get_audio_buffer(outlink, newN);
if (new_window) {
if (s->window)
av_samples_copy(new_window->extended_data, s->window->extended_data, 0, 0,
s->window->nb_samples, new_window->ch_layout.nb_channels, new_window->format);
av_frame_free(&s->window);
s->window = new_window;
} else {
return AVERROR(ENOMEM);
}
}
if (!s->window)
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);
}
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;
ret = config_filter(ctx);
if (ret < 0)
return ret;
ff_anlmdn_init(&s->dsp);
return 0;
}
static int filter_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
{
AudioNLMeansContext *s = ctx->priv;
AVFrame *out = arg;
const int S = s->S;
const int K = s->K;
const int N = s->N;
const int H = s->H;
const int om = s->om;
const float *f = (const float *)(s->window->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];
const float *const weight_lut = s->weight_lut;
const float pdiff_lut_scale = s->pdiff_lut_scale;
const float smooth = fminf(s->m, WEIGHT_LUT_SIZE / pdiff_lut_scale);
const int offset = N - H;
float *src = (float *)s->window->extended_data[ch];
const AVFrame *const in = s->in;
memmove(src, &src[H], offset * sizeof(float));
memcpy(&src[offset], in->extended_data[ch], in->nb_samples * sizeof(float));
memset(&src[offset + in->nb_samples], 0, (H - in->nb_samples) * sizeof(float));
for (int i = S; i < H + S; i++) {
float P = 0.f, Q = 0.f;
int v = 0;
if (i == S) {
for (int j = i - S; j <= i + S; j++) {
if (i == j)
continue;
cache[v++] = s->dsp.compute_distance_ssd(f + i, f + j, K);
}
} else {
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 < 2 * S && !ctx->is_disabled; j++) {
float distance = cache[j];
unsigned weight_lut_idx;
float w;
if (distance < 0.f)
cache[j] = distance = 0.f;
w = distance * sw;
if (w >= smooth)
continue;
weight_lut_idx = w * pdiff_lut_scale;
av_assert2(weight_lut_idx < WEIGHT_LUT_SIZE);
w = weight_lut[weight_lut_idx];
P += w * f[i - S + j + (j >= S)];
Q += w;
}
P += f[i];
Q += 1.f;
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;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioNLMeansContext *s = ctx->priv;
AVFrame *out;
if (av_frame_is_writable(in)) {
out = in;
} else {
out = ff_get_audio_buffer(outlink, in->nb_samples);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
out->pts = in->pts;
}
s->in = in;
ff_filter_execute(ctx, filter_channel, out, NULL, inlink->ch_layout.nb_channels);
if (out != in)
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static int activate(AVFilterContext *ctx)
{
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
AudioNLMeansContext *s = ctx->priv;
AVFrame *in = NULL;
int ret = 0, status;
int64_t pts;
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
ret = ff_inlink_consume_samples(inlink, s->H, s->H, &in);
if (ret < 0)
return ret;
if (ret > 0) {
return filter_frame(inlink, in);
} else if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
ff_outlink_set_status(outlink, status, pts);
return 0;
} else {
if (ff_inlink_queued_samples(inlink) >= s->H) {
ff_filter_set_ready(ctx, 10);
} else if (ff_outlink_frame_wanted(outlink)) {
ff_inlink_request_frame(inlink);
}
return 0;
}
}
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;
return config_filter(ctx);
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioNLMeansContext *s = ctx->priv;
av_frame_free(&s->cache);
av_frame_free(&s->window);
}
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
};
const AVFilter ff_af_anlmdn = {
.name = "anlmdn",
.description = NULL_IF_CONFIG_SMALL("Reduce broadband noise from stream using Non-Local Means."),
.priv_size = sizeof(AudioNLMeansContext),
.priv_class = &anlmdn_class,
.activate = activate,
.uninit = uninit,
FILTER_INPUTS(ff_audio_default_filterpad),
FILTER_OUTPUTS(outputs),
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_FLTP),
.process_command = process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,
};
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