375 lines
10 KiB
C
375 lines
10 KiB
C
/*
|
|
* 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 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=0.00001},0.00001, 10, AFT },
|
|
{ "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, 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, AF },
|
|
{ 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_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->in)
|
|
return AVERROR(ENOMEM);
|
|
|
|
s->cache = ff_get_audio_buffer(outlink, s->S * 2);
|
|
if (!s->cache)
|
|
return AVERROR(ENOMEM);
|
|
|
|
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;
|
|
|
|
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);
|
|
}
|
|
|
|
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 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 }
|
|
};
|
|
|
|
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 = ff_filter_process_command,
|
|
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
|
|
AVFILTER_FLAG_SLICE_THREADS,
|
|
};
|