| /* |
| * 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/avstring.h" |
| #include "libavutil/opt.h" |
| #include "avfilter.h" |
| #include "audio.h" |
| #include "formats.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, "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, 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 inputs[] = { |
| { |
| .name = "default", |
| .type = AVMEDIA_TYPE_AUDIO, |
| }, |
| }; |
| |
| 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(inputs), |
| FILTER_OUTPUTS(outputs), |
| FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_FLTP), |
| .process_command = process_command, |
| .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL | |
| AVFILTER_FLAG_SLICE_THREADS, |
| }; |