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 "libavutil/avassert.h"
22 #include "libavutil/channel_layout.h"
23 #include "libavutil/common.h"
24 #include "libavutil/float_dsp.h"
25 #include "libavutil/opt.h"
26
27 #include "audio.h"
28 #include "avfilter.h"
29 #include "formats.h"
30 #include "filters.h"
31 #include "internal.h"
32
33 enum OutModes {
34 IN_MODE,
35 DESIRED_MODE,
36 OUT_MODE,
37 NOISE_MODE,
38 NB_OMODES
39 };
40
41 typedef struct AudioNLMSContext {
42 const AVClass *class;
43
44 int order;
45 float mu;
46 float eps;
47 float leakage;
48 int output_mode;
49
50 int kernel_size;
51 AVFrame *offset;
52 AVFrame *delay;
53 AVFrame *coeffs;
54 AVFrame *tmp;
55
56 AVFrame *frame[2];
57
58 AVFloatDSPContext *fdsp;
59 } AudioNLMSContext;
60
61 #define OFFSET(x) offsetof(AudioNLMSContext, x)
62 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
63 #define AT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
64
65 static const AVOption anlms_options[] = {
66 { "order", "set the filter order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=256}, 1, INT16_MAX, A },
67 { "mu", "set the filter mu", OFFSET(mu), AV_OPT_TYPE_FLOAT, {.dbl=0.75}, 0, 2, AT },
68 { "eps", "set the filter eps", OFFSET(eps), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AT },
69 { "leakage", "set the filter leakage", OFFSET(leakage), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, AT },
70 { "out_mode", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_OMODES-1, AT, "mode" },
71 { "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AT, "mode" },
72 { "d", "desired", 0, AV_OPT_TYPE_CONST, {.i64=DESIRED_MODE}, 0, 0, AT, "mode" },
73 { "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AT, "mode" },
74 { "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE}, 0, 0, AT, "mode" },
75 { NULL }
76 };
77
78 AVFILTER_DEFINE_CLASS(anlms);
79
query_formats(AVFilterContext * ctx)80 static int query_formats(AVFilterContext *ctx)
81 {
82 AVFilterFormats *formats;
83 AVFilterChannelLayouts *layouts;
84 static const enum AVSampleFormat sample_fmts[] = {
85 AV_SAMPLE_FMT_FLTP,
86 AV_SAMPLE_FMT_NONE
87 };
88 int ret;
89
90 layouts = ff_all_channel_counts();
91 if (!layouts)
92 return AVERROR(ENOMEM);
93 ret = ff_set_common_channel_layouts(ctx, layouts);
94 if (ret < 0)
95 return ret;
96
97 formats = ff_make_format_list(sample_fmts);
98 if (!formats)
99 return AVERROR(ENOMEM);
100 ret = ff_set_common_formats(ctx, formats);
101 if (ret < 0)
102 return ret;
103
104 formats = ff_all_samplerates();
105 if (!formats)
106 return AVERROR(ENOMEM);
107 return ff_set_common_samplerates(ctx, formats);
108 }
109
fir_sample(AudioNLMSContext * s,float sample,float * delay,float * coeffs,float * tmp,int * offset)110 static float fir_sample(AudioNLMSContext *s, float sample, float *delay,
111 float *coeffs, float *tmp, int *offset)
112 {
113 const int order = s->order;
114 float output;
115
116 delay[*offset] = sample;
117
118 memcpy(tmp, coeffs + order - *offset, order * sizeof(float));
119
120 output = s->fdsp->scalarproduct_float(delay, tmp, s->kernel_size);
121
122 if (--(*offset) < 0)
123 *offset = order - 1;
124
125 return output;
126 }
127
process_sample(AudioNLMSContext * s,float input,float desired,float * delay,float * coeffs,float * tmp,int * offsetp)128 static float process_sample(AudioNLMSContext *s, float input, float desired,
129 float *delay, float *coeffs, float *tmp, int *offsetp)
130 {
131 const int order = s->order;
132 const float leakage = s->leakage;
133 const float mu = s->mu;
134 const float a = 1.f - leakage * mu;
135 float sum, output, e, norm, b;
136 int offset = *offsetp;
137
138 delay[offset + order] = input;
139
140 output = fir_sample(s, input, delay, coeffs, tmp, offsetp);
141 e = desired - output;
142
143 sum = s->fdsp->scalarproduct_float(delay, delay, s->kernel_size);
144
145 norm = s->eps + sum;
146 b = mu * e / norm;
147
148 memcpy(tmp, delay + offset, order * sizeof(float));
149
150 s->fdsp->vector_fmul_scalar(coeffs, coeffs, a, s->kernel_size);
151
152 s->fdsp->vector_fmac_scalar(coeffs, tmp, b, s->kernel_size);
153
154 memcpy(coeffs + order, coeffs, order * sizeof(float));
155
156 switch (s->output_mode) {
157 case IN_MODE: output = input; break;
158 case DESIRED_MODE: output = desired; break;
159 case OUT_MODE: /*output = output;*/ break;
160 case NOISE_MODE: output = desired - output; break;
161 }
162 return output;
163 }
164
process_channels(AVFilterContext * ctx,void * arg,int jobnr,int nb_jobs)165 static int process_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
166 {
167 AudioNLMSContext *s = ctx->priv;
168 AVFrame *out = arg;
169 const int start = (out->channels * jobnr) / nb_jobs;
170 const int end = (out->channels * (jobnr+1)) / nb_jobs;
171
172 for (int c = start; c < end; c++) {
173 const float *input = (const float *)s->frame[0]->extended_data[c];
174 const float *desired = (const float *)s->frame[1]->extended_data[c];
175 float *delay = (float *)s->delay->extended_data[c];
176 float *coeffs = (float *)s->coeffs->extended_data[c];
177 float *tmp = (float *)s->tmp->extended_data[c];
178 int *offset = (int *)s->offset->extended_data[c];
179 float *output = (float *)out->extended_data[c];
180
181 for (int n = 0; n < out->nb_samples; n++)
182 output[n] = process_sample(s, input[n], desired[n], delay, coeffs, tmp, offset);
183 }
184
185 return 0;
186 }
187
activate(AVFilterContext * ctx)188 static int activate(AVFilterContext *ctx)
189 {
190 AudioNLMSContext *s = ctx->priv;
191 int i, ret, status;
192 int nb_samples;
193 int64_t pts;
194
195 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
196
197 nb_samples = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]),
198 ff_inlink_queued_samples(ctx->inputs[1]));
199 for (i = 0; i < ctx->nb_inputs && nb_samples > 0; i++) {
200 if (s->frame[i])
201 continue;
202
203 if (ff_inlink_check_available_samples(ctx->inputs[i], nb_samples) > 0) {
204 ret = ff_inlink_consume_samples(ctx->inputs[i], nb_samples, nb_samples, &s->frame[i]);
205 if (ret < 0)
206 return ret;
207 }
208 }
209
210 if (s->frame[0] && s->frame[1]) {
211 AVFrame *out;
212
213 out = ff_get_audio_buffer(ctx->outputs[0], s->frame[0]->nb_samples);
214 if (!out) {
215 av_frame_free(&s->frame[0]);
216 av_frame_free(&s->frame[1]);
217 return AVERROR(ENOMEM);
218 }
219
220 ctx->internal->execute(ctx, process_channels, out, NULL, FFMIN(ctx->outputs[0]->channels,
221 ff_filter_get_nb_threads(ctx)));
222
223 out->pts = s->frame[0]->pts;
224
225 av_frame_free(&s->frame[0]);
226 av_frame_free(&s->frame[1]);
227
228 ret = ff_filter_frame(ctx->outputs[0], out);
229 if (ret < 0)
230 return ret;
231 }
232
233 if (!nb_samples) {
234 for (i = 0; i < 2; i++) {
235 if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
236 ff_outlink_set_status(ctx->outputs[0], status, pts);
237 return 0;
238 }
239 }
240 }
241
242 if (ff_outlink_frame_wanted(ctx->outputs[0])) {
243 for (i = 0; i < 2; i++) {
244 if (ff_inlink_queued_samples(ctx->inputs[i]) > 0)
245 continue;
246 ff_inlink_request_frame(ctx->inputs[i]);
247 return 0;
248 }
249 }
250 return 0;
251 }
252
config_output(AVFilterLink * outlink)253 static int config_output(AVFilterLink *outlink)
254 {
255 AVFilterContext *ctx = outlink->src;
256 AudioNLMSContext *s = ctx->priv;
257
258 s->kernel_size = FFALIGN(s->order, 16);
259
260 if (!s->offset)
261 s->offset = ff_get_audio_buffer(outlink, 1);
262 if (!s->delay)
263 s->delay = ff_get_audio_buffer(outlink, 2 * s->kernel_size);
264 if (!s->coeffs)
265 s->coeffs = ff_get_audio_buffer(outlink, 2 * s->kernel_size);
266 if (!s->tmp)
267 s->tmp = ff_get_audio_buffer(outlink, s->kernel_size);
268 if (!s->delay || !s->coeffs || !s->offset || !s->tmp)
269 return AVERROR(ENOMEM);
270
271 return 0;
272 }
273
init(AVFilterContext * ctx)274 static av_cold int init(AVFilterContext *ctx)
275 {
276 AudioNLMSContext *s = ctx->priv;
277
278 s->fdsp = avpriv_float_dsp_alloc(0);
279 if (!s->fdsp)
280 return AVERROR(ENOMEM);
281
282 return 0;
283 }
284
uninit(AVFilterContext * ctx)285 static av_cold void uninit(AVFilterContext *ctx)
286 {
287 AudioNLMSContext *s = ctx->priv;
288
289 av_freep(&s->fdsp);
290 av_frame_free(&s->delay);
291 av_frame_free(&s->coeffs);
292 av_frame_free(&s->offset);
293 av_frame_free(&s->tmp);
294 }
295
296 static const AVFilterPad inputs[] = {
297 {
298 .name = "input",
299 .type = AVMEDIA_TYPE_AUDIO,
300 },
301 {
302 .name = "desired",
303 .type = AVMEDIA_TYPE_AUDIO,
304 },
305 { NULL }
306 };
307
308 static const AVFilterPad outputs[] = {
309 {
310 .name = "default",
311 .type = AVMEDIA_TYPE_AUDIO,
312 .config_props = config_output,
313 },
314 { NULL }
315 };
316
317 AVFilter ff_af_anlms = {
318 .name = "anlms",
319 .description = NULL_IF_CONFIG_SMALL("Apply Normalized Least-Mean-Squares algorithm to first audio stream."),
320 .priv_size = sizeof(AudioNLMSContext),
321 .priv_class = &anlms_class,
322 .init = init,
323 .uninit = uninit,
324 .activate = activate,
325 .query_formats = query_formats,
326 .inputs = inputs,
327 .outputs = outputs,
328 .flags = AVFILTER_FLAG_SLICE_THREADS,
329 .process_command = ff_filter_process_command,
330 };
331