1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
|
/*
* Copyright (c) 2021 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/channel_layout.h"
#include "libavutil/common.h"
#include "avfilter.h"
#include "filters.h"
#include "internal.h"
typedef struct ChanStats {
double u;
double v;
double uv;
} ChanStats;
typedef struct AudioSDRContext {
int channels;
uint64_t nb_samples;
double max;
ChanStats *chs;
AVFrame *cache[2];
int (*filter)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
} AudioSDRContext;
#define SDR_FILTER(name, type) \
static int sdr_##name(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)\
{ \
AudioSDRContext *s = ctx->priv; \
AVFrame *u = s->cache[0]; \
AVFrame *v = s->cache[1]; \
const int channels = u->ch_layout.nb_channels; \
const int start = (channels * jobnr) / nb_jobs; \
const int end = (channels * (jobnr+1)) / nb_jobs; \
const int nb_samples = u->nb_samples; \
\
for (int ch = start; ch < end; ch++) { \
ChanStats *chs = &s->chs[ch]; \
const type *const us = (type *)u->extended_data[ch]; \
const type *const vs = (type *)v->extended_data[ch]; \
double sum_uv = 0.; \
double sum_u = 0.; \
\
for (int n = 0; n < nb_samples; n++) { \
sum_u += us[n] * us[n]; \
sum_uv += (us[n] - vs[n]) * (us[n] - vs[n]); \
} \
\
chs->uv += sum_uv; \
chs->u += sum_u; \
} \
\
return 0; \
}
SDR_FILTER(fltp, float)
SDR_FILTER(dblp, double)
#define SISDR_FILTER(name, type) \
static int sisdr_##name(AVFilterContext *ctx, void *arg,int jobnr,int nb_jobs)\
{ \
AudioSDRContext *s = ctx->priv; \
AVFrame *u = s->cache[0]; \
AVFrame *v = s->cache[1]; \
const int channels = u->ch_layout.nb_channels; \
const int start = (channels * jobnr) / nb_jobs; \
const int end = (channels * (jobnr+1)) / nb_jobs; \
const int nb_samples = u->nb_samples; \
\
for (int ch = start; ch < end; ch++) { \
ChanStats *chs = &s->chs[ch]; \
const type *const us = (type *)u->extended_data[ch]; \
const type *const vs = (type *)v->extended_data[ch]; \
double sum_uv = 0.; \
double sum_u = 0.; \
double sum_v = 0.; \
\
for (int n = 0; n < nb_samples; n++) { \
sum_u += us[n] * us[n]; \
sum_v += vs[n] * vs[n]; \
sum_uv += us[n] * vs[n]; \
} \
\
chs->uv += sum_uv; \
chs->u += sum_u; \
chs->v += sum_v; \
} \
\
return 0; \
}
SISDR_FILTER(fltp, float)
SISDR_FILTER(dblp, double)
#define PSNR_FILTER(name, type) \
static int psnr_##name(AVFilterContext *ctx, void *arg, int jobnr,int nb_jobs)\
{ \
AudioSDRContext *s = ctx->priv; \
AVFrame *u = s->cache[0]; \
AVFrame *v = s->cache[1]; \
const int channels = u->ch_layout.nb_channels; \
const int start = (channels * jobnr) / nb_jobs; \
const int end = (channels * (jobnr+1)) / nb_jobs; \
const int nb_samples = u->nb_samples; \
\
for (int ch = start; ch < end; ch++) { \
ChanStats *chs = &s->chs[ch]; \
const type *const us = (type *)u->extended_data[ch]; \
const type *const vs = (type *)v->extended_data[ch]; \
double sum_uv = 0.; \
\
for (int n = 0; n < nb_samples; n++) \
sum_uv += (us[n] - vs[n]) * (us[n] - vs[n]); \
\
chs->uv += sum_uv; \
} \
\
return 0; \
}
PSNR_FILTER(fltp, float)
PSNR_FILTER(dblp, double)
static int activate(AVFilterContext *ctx)
{
AudioSDRContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
int ret, status, available;
int64_t pts;
FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
available = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]), ff_inlink_queued_samples(ctx->inputs[1]));
if (available > 0) {
AVFrame *out;
for (int i = 0; i < 2; i++) {
ret = ff_inlink_consume_samples(ctx->inputs[i], available, available, &s->cache[i]);
if (ret < 0) {
av_frame_free(&s->cache[0]);
av_frame_free(&s->cache[1]);
return ret;
}
}
if (!ctx->is_disabled)
ff_filter_execute(ctx, s->filter, NULL, NULL,
FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
av_frame_free(&s->cache[1]);
out = s->cache[0];
s->cache[0] = NULL;
s->nb_samples += available;
return ff_filter_frame(outlink, out);
}
for (int i = 0; i < 2; i++) {
if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
ff_outlink_set_status(outlink, status, pts);
return 0;
}
}
if (ff_outlink_frame_wanted(outlink)) {
for (int i = 0; i < 2; i++) {
if (ff_inlink_queued_samples(ctx->inputs[i]) > 0)
continue;
ff_inlink_request_frame(ctx->inputs[i]);
}
return 0;
}
return FFERROR_NOT_READY;
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AVFilterLink *inlink = ctx->inputs[0];
AudioSDRContext *s = ctx->priv;
s->channels = inlink->ch_layout.nb_channels;
if (!strcmp(ctx->filter->name, "asdr"))
s->filter = inlink->format == AV_SAMPLE_FMT_FLTP ? sdr_fltp : sdr_dblp;
else if (!strcmp(ctx->filter->name, "asisdr"))
s->filter = inlink->format == AV_SAMPLE_FMT_FLTP ? sisdr_fltp : sisdr_dblp;
else
s->filter = inlink->format == AV_SAMPLE_FMT_FLTP ? psnr_fltp : psnr_dblp;
s->max = inlink->format == AV_SAMPLE_FMT_FLTP ? FLT_MAX : DBL_MAX;
s->chs = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->chs));
if (!s->chs)
return AVERROR(ENOMEM);
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioSDRContext *s = ctx->priv;
if (!strcmp(ctx->filter->name, "asdr")) {
for (int ch = 0; ch < s->channels; ch++)
av_log(ctx, AV_LOG_INFO, "SDR ch%d: %g dB\n", ch, 10. * log10(s->chs[ch].u / s->chs[ch].uv));
} else if (!strcmp(ctx->filter->name, "asisdr")) {
for (int ch = 0; ch < s->channels; ch++) {
double scale = s->chs[ch].uv / s->chs[ch].v;
double sisdr = scale * scale * s->chs[ch].v / fmax(0., s->chs[ch].u + scale*scale*s->chs[ch].v - 2.0*scale*s->chs[ch].uv);
av_log(ctx, AV_LOG_INFO, "SI-SDR ch%d: %g dB\n", ch, 10. * log10(sisdr));
}
} else {
for (int ch = 0; ch < s->channels; ch++) {
double psnr = s->chs[ch].uv > 0.0 ? 2.0 * log(s->max) - log(s->nb_samples / s->chs[ch].uv) : INFINITY;
av_log(ctx, AV_LOG_INFO, "PSNR ch%d: %g dB\n", ch, psnr);
}
}
av_frame_free(&s->cache[0]);
av_frame_free(&s->cache[1]);
av_freep(&s->chs);
}
static const AVFilterPad inputs[] = {
{
.name = "input0",
.type = AVMEDIA_TYPE_AUDIO,
},
{
.name = "input1",
.type = AVMEDIA_TYPE_AUDIO,
},
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
};
const AVFilter ff_af_asdr = {
.name = "asdr",
.description = NULL_IF_CONFIG_SMALL("Measure Audio Signal-to-Distortion Ratio."),
.priv_size = sizeof(AudioSDRContext),
.activate = activate,
.uninit = uninit,
.flags = AVFILTER_FLAG_METADATA_ONLY |
AVFILTER_FLAG_SLICE_THREADS |
AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_DBLP),
};
const AVFilter ff_af_apsnr = {
.name = "apsnr",
.description = NULL_IF_CONFIG_SMALL("Measure Audio Peak Signal-to-Noise Ratio."),
.priv_size = sizeof(AudioSDRContext),
.activate = activate,
.uninit = uninit,
.flags = AVFILTER_FLAG_METADATA_ONLY |
AVFILTER_FLAG_SLICE_THREADS |
AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_DBLP),
};
const AVFilter ff_af_asisdr = {
.name = "asisdr",
.description = NULL_IF_CONFIG_SMALL("Measure Audio Scale-Invariant Signal-to-Distortion Ratio."),
.priv_size = sizeof(AudioSDRContext),
.activate = activate,
.uninit = uninit,
.flags = AVFILTER_FLAG_METADATA_ONLY |
AVFILTER_FLAG_SLICE_THREADS |
AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_DBLP),
};
|
