宅霸联机平台(宅霸游戏联机平台下载) V3.6.6官方版

Line	Branch	Exec	Source
1			/*
2			* Copyright (c) 2019 Paul B Mahol
3			*
4			* This file is part of FFmpeg.
5			*
6			* FFmpeg is free software; you can redistribute it and/or
7			* modify it under the terms of the GNU Lesser General Public
8			* License as published by the Free Software Foundation; either
9			* version 2.1 of the License, or (at your option) any later version.
10			*
11			* FFmpeg is distributed in the hope that it will be useful,
12			* but WITHOUT ANY WARRANTY; without even the implied warranty of
13			* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14			* Lesser General Public License for more details.
15			*
16			* You should have received a copy of the GNU Lesser General Public
17			* License along with FFmpeg; if not, write to the Free Software
18			* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19			*/
20
21			#include <float.h>
22
23			#include "libavutil/avassert.h"
24			#include "libavutil/opt.h"
25			#include "avfilter.h"
26			#include "audio.h"
27			#include "filters.h"
28
29			#include "af_anlmdndsp.h"
30
31			#define WEIGHT_LUT_NBITS 20
32			#define WEIGHT_LUT_SIZE (1<<WEIGHT_LUT_NBITS)
33
34			typedef struct AudioNLMeansContext {
35			const AVClass *class;
36
37			float a;
38			int64_t pd;
39			int64_t rd;
40			float m;
41			int om;
42
43			float pdiff_lut_scale;
44			float weight_lut[WEIGHT_LUT_SIZE];
45
46			int K;
47			int S;
48			int N;
49			int H;
50
51			AVFrame *in;
52			AVFrame *cache;
53			AVFrame *window;
54
55			AudioNLMDNDSPContext dsp;
56			} AudioNLMeansContext;
57
58			enum OutModes {
59			IN_MODE,
60			OUT_MODE,
61			NOISE_MODE,
62			NB_MODES
63			};
64
65			#define OFFSET(x) offsetof(AudioNLMeansContext, x)
66			#define AFT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
67
68			static const AVOption anlmdn_options[] = {
69			{ "strength", "set denoising strength", OFFSET(a), AV_OPT_TYPE_FLOAT, {.dbl=0.00001},0.00001, 10000, AFT },
70			{ "s", "set denoising strength", OFFSET(a), AV_OPT_TYPE_FLOAT, {.dbl=0.00001},0.00001, 10000, AFT },
71			{ "patch", "set patch duration", OFFSET(pd), AV_OPT_TYPE_DURATION, {.i64=2000}, 1000, 100000, AFT },
72			{ "p", "set patch duration", OFFSET(pd), AV_OPT_TYPE_DURATION, {.i64=2000}, 1000, 100000, AFT },
73			{ "research", "set research duration", OFFSET(rd), AV_OPT_TYPE_DURATION, {.i64=6000}, 2000, 300000, AFT },
74			{ "r", "set research duration", OFFSET(rd), AV_OPT_TYPE_DURATION, {.i64=6000}, 2000, 300000, AFT },
75			{ "output", "set output mode", OFFSET(om), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_MODES-1, AFT, .unit = "mode" },
76			{ "o", "set output mode", OFFSET(om), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_MODES-1, AFT, .unit = "mode" },
77			{ "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AFT, .unit = "mode" },
78			{ "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AFT, .unit = "mode" },
79			{ "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE},0, 0, AFT, .unit = "mode" },
80			{ "smooth", "set smooth factor", OFFSET(m), AV_OPT_TYPE_FLOAT, {.dbl=11.}, 1, 1000, AFT },
81			{ "m", "set smooth factor", OFFSET(m), AV_OPT_TYPE_FLOAT, {.dbl=11.}, 1, 1000, AFT },
82			{ NULL }
83			};
84
85			AVFILTER_DEFINE_CLASS(anlmdn);
86
87		✗	static inline float sqrdiff(float x, float y)
88			{
89		✗	const float diff = x - y;
90
91		✗	return diff * diff;
92			}
93
94		✗	static float compute_distance_ssd_c(const float *f1, const float *f2, ptrdiff_t K)
95			{
96		✗	float distance = 0.;
97
98		✗	for (int k = -K; k <= K; k++)
99		✗	distance += sqrdiff(f1[k], f2[k]);
100
101		✗	return distance;
102			}
103
104		✗	static void compute_cache_c(float *cache, const float *f,
105			ptrdiff_t S, ptrdiff_t K,
106			ptrdiff_t i, ptrdiff_t jj)
107			{
108		✗	int v = 0;
109
110		✗	for (int j = jj; j < jj + S; j++, v++)
111		✗	cache[v] += -sqrdiff(f[i - K - 1], f[j - K - 1]) + sqrdiff(f[i + K], f[j + K]);
112		✗	}
113
114		✗	void ff_anlmdn_init(AudioNLMDNDSPContext *dsp)
115			{
116		✗	dsp->compute_distance_ssd = compute_distance_ssd_c;
117		✗	dsp->compute_cache = compute_cache_c;
118
119			#if ARCH_X86
120		✗	ff_anlmdn_init_x86(dsp);
121			#endif
122		✗	}
123
124		✗	static int config_filter(AVFilterContext *ctx)
125			{
126		✗	AudioNLMeansContext *s = ctx->priv;
127		✗	AVFilterLink *outlink = ctx->outputs[0];
128			int newK, newS, newH, newN;
129
130		✗	newK = av_rescale(s->pd, outlink->sample_rate, AV_TIME_BASE);
131		✗	newS = av_rescale(s->rd, outlink->sample_rate, AV_TIME_BASE);
132
133		✗	newH = newK * 2 + 1;
134		✗	newN = newH + (newK + newS) * 2;
135
136		✗	av_log(ctx, AV_LOG_DEBUG, "K:%d S:%d H:%d N:%d\n", newK, newS, newH, newN);
137
138		✗	if (!s->cache || s->cache->nb_samples < newS * 2) {
139		✗	AVFrame *new_cache = ff_get_audio_buffer(outlink, newS * 2);
140		✗	if (new_cache) {
141		✗	if (s->cache)
142		✗	av_samples_copy(new_cache->extended_data, s->cache->extended_data, 0, 0,
143		✗	s->cache->nb_samples, new_cache->ch_layout.nb_channels, new_cache->format);
144		✗	av_frame_free(&s->cache);
145		✗	s->cache = new_cache;
146			} else {
147		✗	return AVERROR(ENOMEM);
148			}
149			}
150		✗	if (!s->cache)
151		✗	return AVERROR(ENOMEM);
152
153		✗	if (!s->window || s->window->nb_samples < newN) {
154		✗	AVFrame *new_window = ff_get_audio_buffer(outlink, newN);
155		✗	if (new_window) {
156		✗	if (s->window)
157		✗	av_samples_copy(new_window->extended_data, s->window->extended_data, 0, 0,
158		✗	s->window->nb_samples, new_window->ch_layout.nb_channels, new_window->format);
159		✗	av_frame_free(&s->window);
160		✗	s->window = new_window;
161			} else {
162		✗	return AVERROR(ENOMEM);
163			}
164			}
165		✗	if (!s->window)
166		✗	return AVERROR(ENOMEM);
167
168		✗	s->pdiff_lut_scale = 1.f / s->m * WEIGHT_LUT_SIZE;
169		✗	for (int i = 0; i < WEIGHT_LUT_SIZE; i++) {
170		✗	float w = -i / s->pdiff_lut_scale;
171
172		✗	s->weight_lut[i] = expf(w);
173			}
174
175		✗	s->K = newK;
176		✗	s->S = newS;
177		✗	s->H = newH;
178		✗	s->N = newN;
179
180		✗	return 0;
181			}
182
183		✗	static int config_output(AVFilterLink *outlink)
184			{
185		✗	AVFilterContext *ctx = outlink->src;
186		✗	AudioNLMeansContext *s = ctx->priv;
187			int ret;
188
189		✗	ret = config_filter(ctx);
190		✗	if (ret < 0)
191		✗	return ret;
192
193		✗	ff_anlmdn_init(&s->dsp);
194
195		✗	return 0;
196			}
197
198		✗	static int filter_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
199			{
200		✗	AudioNLMeansContext *s = ctx->priv;
201		✗	AVFrame *out = arg;
202		✗	const int S = s->S;
203		✗	const int K = s->K;
204		✗	const int N = s->N;
205		✗	const int H = s->H;
206		✗	const int om = s->om;
207		✗	const float *f = (const float *)(s->window->extended_data[ch]) + K;
208		✗	float *cache = (float *)s->cache->extended_data[ch];
209		✗	const float sw = (65536.f / (4 * K + 2)) / sqrtf(s->a);
210		✗	float *dst = (float *)out->extended_data[ch];
211		✗	const float *const weight_lut = s->weight_lut;
212		✗	const float pdiff_lut_scale = s->pdiff_lut_scale;
213		✗	const float smooth = fminf(s->m, WEIGHT_LUT_SIZE / pdiff_lut_scale);
214		✗	const int offset = N - H;
215		✗	float *src = (float *)s->window->extended_data[ch];
216		✗	const AVFrame *const in = s->in;
217
218		✗	memmove(src, &src[H], offset * sizeof(float));
219		✗	memcpy(&src[offset], in->extended_data[ch], in->nb_samples * sizeof(float));
220		✗	memset(&src[offset + in->nb_samples], 0, (H - in->nb_samples) * sizeof(float));
221
222		✗	for (int i = S; i < H + S; i++) {
223		✗	float P = 0.f, Q = 0.f;
224		✗	int v = 0;
225
226		✗	if (i == S) {
227		✗	for (int j = i - S; j <= i + S; j++) {
228		✗	if (i == j)
229		✗	continue;
230		✗	cache[v++] = s->dsp.compute_distance_ssd(f + i, f + j, K);
231			}
232			} else {
233		✗	s->dsp.compute_cache(cache, f, S, K, i, i - S);
234		✗	s->dsp.compute_cache(cache + S, f, S, K, i, i + 1);
235			}
236
237		✗	for (int j = 0; j < 2 * S && !ctx->is_disabled; j++) {
238		✗	float distance = cache[j];
239			unsigned weight_lut_idx;
240			float w;
241
242		✗	if (distance < 0.f)
243		✗	cache[j] = distance = 0.f;
244		✗	w = distance * sw;
245		✗	if (w >= smooth)
246		✗	continue;
247		✗	weight_lut_idx = w * pdiff_lut_scale;
248			av_assert2(weight_lut_idx < WEIGHT_LUT_SIZE);
249		✗	w = weight_lut[weight_lut_idx];
250		✗	P += w * f[i - S + j + (j >= S)];
251		✗	Q += w;
252			}
253
254		✗	P += f[i];
255		✗	Q += 1.f;
256
257		✗	switch (om) {
258		✗	case IN_MODE: dst[i - S] = f[i]; break;
259		✗	case OUT_MODE: dst[i - S] = P / Q; break;
260		✗	case NOISE_MODE: dst[i - S] = f[i] - (P / Q); break;
261			}
262			}
263
264		✗	return 0;
265			}
266
267		✗	static int filter_frame(AVFilterLink *inlink, AVFrame *in)
268			{
269		✗	AVFilterContext *ctx = inlink->dst;
270		✗	AVFilterLink *outlink = ctx->outputs[0];
271		✗	AudioNLMeansContext *s = ctx->priv;
272			AVFrame *out;
273
274		✗	if (av_frame_is_writable(in)) {
275		✗	out = in;
276			} else {
277		✗	out = ff_get_audio_buffer(outlink, in->nb_samples);
278		✗	if (!out) {
279		✗	av_frame_free(&in);
280		✗	return AVERROR(ENOMEM);
281			}
282
283		✗	out->pts = in->pts;
284			}
285
286		✗	s->in = in;
287		✗	ff_filter_execute(ctx, filter_channel, out, NULL, inlink->ch_layout.nb_channels);
288
289		✗	if (out != in)
290		✗	av_frame_free(&in);
291		✗	return ff_filter_frame(outlink, out);
292			}
293
294		✗	static int activate(AVFilterContext *ctx)
295			{
296		✗	AVFilterLink *inlink = ctx->inputs[0];
297		✗	AVFilterLink *outlink = ctx->outputs[0];
298		✗	AudioNLMeansContext *s = ctx->priv;
299		✗	AVFrame *in = NULL;
300		✗	int ret = 0, status;
301			int64_t pts;
302
303		✗	FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
304
305		✗	ret = ff_inlink_consume_samples(inlink, s->H, s->H, &in);
306		✗	if (ret < 0)
307		✗	return ret;
308
309		✗	if (ret > 0) {
310		✗	return filter_frame(inlink, in);
311		✗	} else if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
312		✗	ff_outlink_set_status(outlink, status, pts);
313		✗	return 0;
314			} else {
315		✗	if (ff_inlink_queued_samples(inlink) >= s->H) {
316		✗	ff_filter_set_ready(ctx, 10);
317		✗	} else if (ff_outlink_frame_wanted(outlink)) {
318		✗	ff_inlink_request_frame(inlink);
319			}
320		✗	return 0;
321			}
322			}
323
324		✗	static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
325			char *res, int res_len, int flags)
326			{
327			int ret;
328
329		✗	ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
330		✗	if (ret < 0)
331		✗	return ret;
332
333		✗	return config_filter(ctx);
334			}
335
336		✗	static av_cold void uninit(AVFilterContext *ctx)
337			{
338		✗	AudioNLMeansContext *s = ctx->priv;
339
340		✗	av_frame_free(&s->cache);
341		✗	av_frame_free(&s->window);
342		✗	}
343
344			static const AVFilterPad outputs[] = {
345			{
346			.name = "default",
347			.type = AVMEDIA_TYPE_AUDIO,
348			.config_props = config_output,
349			},
350			};
351
352			const FFFilter ff_af_anlmdn = {
353			.p.name = "anlmdn",
354			.p.description = NULL_IF_CONFIG_SMALL("Reduce broadband noise from stream using Non-Local Means."),
355			.p.priv_class = &anlmdn_class,
356			.p.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
357			AVFILTER_FLAG_SLICE_THREADS,
358			.priv_size = sizeof(AudioNLMeansContext),
359			.activate = activate,
360			.uninit = uninit,
361			FILTER_INPUTS(ff_audio_default_filterpad),
362			FILTER_OUTPUTS(outputs),
363			FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_FLTP),
364			.process_command = process_command,
365			};
366