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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/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 {
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;
int offset;
AVFrame *in;
AVFrame *cache;
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 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, 10, AFT },
{ "s", "set denoising strength", OFFSET(a), AV_OPT_TYPE_FLOAT, {.dbl=0.00001},0.00001, 10, 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, "mode" },
{ "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" },
{ "smooth", "set smooth factor", OFFSET(m), AV_OPT_TYPE_FLOAT, {.dbl=11.}, 1, 15, AFT },
{ "m", "set smooth factor", OFFSET(m), AV_OPT_TYPE_FLOAT, {.dbl=11.}, 1, 15, AFT },
{ NULL }
};
AVFILTER_DEFINE_CLASS(anlmdn);
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layouts = NULL;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE
};
int ret;
formats = ff_make_format_list(sample_fmts);
if (!formats)
return AVERROR(ENOMEM);
ret = ff_set_common_formats(ctx, formats);
if (ret < 0)
return ret;
layouts = ff_all_channel_counts();
if (!layouts)
return AVERROR(ENOMEM);
ret = ff_set_common_channel_layouts(ctx, layouts);
if (ret < 0)
return ret;
formats = ff_all_samplerates();
return ff_set_common_samplerates(ctx, formats);
}
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 += SQR(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] += -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)
{
AudioNLMeansContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
int newK, newS, newH, newN;
AVFrame *new_in, *new_cache;
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) {
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);
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;
}
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 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;
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++) {
const float distance = cache[j];
unsigned weight_lut_idx;
float w;
if (distance < 0.f) {
cache[j] = 0.f;
continue;
}
w = distance * sw;
if (w >= smooth)
continue;
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;
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 = NULL;
int available, wanted, ret;
if (s->pts == AV_NOPTS_VALUE)
s->pts = in->pts;
ret = av_audio_fifo_write(s->fifo, (void **)in->extended_data,
in->nb_samples);
av_frame_free(&in);
s->offset = 0;
available = av_audio_fifo_size(s->fifo);
wanted = (available / s->H) * s->H;
if (wanted >= s->H && available >= s->N) {
out = ff_get_audio_buffer(outlink, wanted);
if (!out)
return AVERROR(ENOMEM);
}
while (available >= s->N) {
ret = av_audio_fifo_peek(s->fifo, (void **)s->in->extended_data, s->N);
if (ret < 0)
break;
ctx->internal->execute(ctx, filter_channel, out, NULL, inlink->channels);
av_audio_fifo_drain(s->fifo, s->H);
s->offset += s->H;
available -= s->H;
}
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 += 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;
av_audio_fifo_free(s->fifo);
av_frame_free(&s->in);
av_frame_free(&s->cache);
}
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
{ NULL }
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
.request_frame = request_frame,
},
{ NULL }
};
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,
.priv_size = sizeof(AudioNLMeansContext),
.priv_class = &anlmdn_class,
.uninit = uninit,
.inputs = inputs,
.outputs = outputs,
.process_command = process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,
};
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