1 |
|
|
/* |
2 |
|
|
* Dynamic Audio Normalizer |
3 |
|
|
* Copyright (c) 2015 LoRd_MuldeR <mulder2@gmx.de>. Some rights reserved. |
4 |
|
|
* |
5 |
|
|
* This file is part of FFmpeg. |
6 |
|
|
* |
7 |
|
|
* FFmpeg is free software; you can redistribute it and/or |
8 |
|
|
* modify it under the terms of the GNU Lesser General Public |
9 |
|
|
* License as published by the Free Software Foundation; either |
10 |
|
|
* version 2.1 of the License, or (at your option) any later version. |
11 |
|
|
* |
12 |
|
|
* FFmpeg is distributed in the hope that it will be useful, |
13 |
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of |
14 |
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
15 |
|
|
* Lesser General Public License for more details. |
16 |
|
|
* |
17 |
|
|
* You should have received a copy of the GNU Lesser General Public |
18 |
|
|
* License along with FFmpeg; if not, write to the Free Software |
19 |
|
|
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
20 |
|
|
*/ |
21 |
|
|
|
22 |
|
|
/** |
23 |
|
|
* @file |
24 |
|
|
* Dynamic Audio Normalizer |
25 |
|
|
*/ |
26 |
|
|
|
27 |
|
|
#include <float.h> |
28 |
|
|
|
29 |
|
|
#include "libavutil/avassert.h" |
30 |
|
|
#include "libavutil/opt.h" |
31 |
|
|
|
32 |
|
|
#define MIN_FILTER_SIZE 3 |
33 |
|
|
#define MAX_FILTER_SIZE 301 |
34 |
|
|
|
35 |
|
|
#define FF_BUFQUEUE_SIZE (MAX_FILTER_SIZE + 1) |
36 |
|
|
#include "libavfilter/bufferqueue.h" |
37 |
|
|
|
38 |
|
|
#include "audio.h" |
39 |
|
|
#include "avfilter.h" |
40 |
|
|
#include "filters.h" |
41 |
|
|
#include "internal.h" |
42 |
|
|
|
43 |
|
|
typedef struct local_gain { |
44 |
|
|
double max_gain; |
45 |
|
|
double threshold; |
46 |
|
|
} local_gain; |
47 |
|
|
|
48 |
|
|
typedef struct cqueue { |
49 |
|
|
double *elements; |
50 |
|
|
int size; |
51 |
|
|
int max_size; |
52 |
|
|
int nb_elements; |
53 |
|
|
} cqueue; |
54 |
|
|
|
55 |
|
|
typedef struct DynamicAudioNormalizerContext { |
56 |
|
|
const AVClass *class; |
57 |
|
|
|
58 |
|
|
struct FFBufQueue queue; |
59 |
|
|
|
60 |
|
|
int frame_len; |
61 |
|
|
int frame_len_msec; |
62 |
|
|
int filter_size; |
63 |
|
|
int dc_correction; |
64 |
|
|
int channels_coupled; |
65 |
|
|
int alt_boundary_mode; |
66 |
|
|
|
67 |
|
|
double peak_value; |
68 |
|
|
double max_amplification; |
69 |
|
|
double target_rms; |
70 |
|
|
double compress_factor; |
71 |
|
|
double threshold; |
72 |
|
|
double *prev_amplification_factor; |
73 |
|
|
double *dc_correction_value; |
74 |
|
|
double *compress_threshold; |
75 |
|
|
double *weights; |
76 |
|
|
|
77 |
|
|
int channels; |
78 |
|
|
int eof; |
79 |
|
|
int64_t pts; |
80 |
|
|
|
81 |
|
|
cqueue **gain_history_original; |
82 |
|
|
cqueue **gain_history_minimum; |
83 |
|
|
cqueue **gain_history_smoothed; |
84 |
|
|
cqueue **threshold_history; |
85 |
|
|
|
86 |
|
|
cqueue *is_enabled; |
87 |
|
|
} DynamicAudioNormalizerContext; |
88 |
|
|
|
89 |
|
|
#define OFFSET(x) offsetof(DynamicAudioNormalizerContext, x) |
90 |
|
|
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM |
91 |
|
|
|
92 |
|
|
static const AVOption dynaudnorm_options[] = { |
93 |
|
|
{ "framelen", "set the frame length in msec", OFFSET(frame_len_msec), AV_OPT_TYPE_INT, {.i64 = 500}, 10, 8000, FLAGS }, |
94 |
|
|
{ "f", "set the frame length in msec", OFFSET(frame_len_msec), AV_OPT_TYPE_INT, {.i64 = 500}, 10, 8000, FLAGS }, |
95 |
|
|
{ "gausssize", "set the filter size", OFFSET(filter_size), AV_OPT_TYPE_INT, {.i64 = 31}, 3, 301, FLAGS }, |
96 |
|
|
{ "g", "set the filter size", OFFSET(filter_size), AV_OPT_TYPE_INT, {.i64 = 31}, 3, 301, FLAGS }, |
97 |
|
|
{ "peak", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0.95}, 0.0, 1.0, FLAGS }, |
98 |
|
|
{ "p", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl = 0.95}, 0.0, 1.0, FLAGS }, |
99 |
|
|
{ "maxgain", "set the max amplification", OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 10.0}, 1.0, 100.0, FLAGS }, |
100 |
|
|
{ "m", "set the max amplification", OFFSET(max_amplification), AV_OPT_TYPE_DOUBLE, {.dbl = 10.0}, 1.0, 100.0, FLAGS }, |
101 |
|
|
{ "targetrms", "set the target RMS", OFFSET(target_rms), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS }, |
102 |
|
|
{ "r", "set the target RMS", OFFSET(target_rms), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS }, |
103 |
|
|
{ "coupling", "set channel coupling", OFFSET(channels_coupled), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS }, |
104 |
|
|
{ "n", "set channel coupling", OFFSET(channels_coupled), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, FLAGS }, |
105 |
|
|
{ "correctdc", "set DC correction", OFFSET(dc_correction), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS }, |
106 |
|
|
{ "c", "set DC correction", OFFSET(dc_correction), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS }, |
107 |
|
|
{ "altboundary", "set alternative boundary mode", OFFSET(alt_boundary_mode), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS }, |
108 |
|
|
{ "b", "set alternative boundary mode", OFFSET(alt_boundary_mode), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, FLAGS }, |
109 |
|
|
{ "compress", "set the compress factor", OFFSET(compress_factor), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 30.0, FLAGS }, |
110 |
|
|
{ "s", "set the compress factor", OFFSET(compress_factor), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 30.0, FLAGS }, |
111 |
|
|
{ "threshold", "set the threshold value", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS }, |
112 |
|
|
{ "t", "set the threshold value", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl = 0.0}, 0.0, 1.0, FLAGS }, |
113 |
|
|
{ NULL } |
114 |
|
|
}; |
115 |
|
|
|
116 |
|
|
AVFILTER_DEFINE_CLASS(dynaudnorm); |
117 |
|
|
|
118 |
|
|
static av_cold int init(AVFilterContext *ctx) |
119 |
|
|
{ |
120 |
|
|
DynamicAudioNormalizerContext *s = ctx->priv; |
121 |
|
|
|
122 |
|
|
if (!(s->filter_size & 1)) { |
123 |
|
|
av_log(ctx, AV_LOG_WARNING, "filter size %d is invalid. Changing to an odd value.\n", s->filter_size); |
124 |
|
|
s->filter_size |= 1; |
125 |
|
|
} |
126 |
|
|
|
127 |
|
|
return 0; |
128 |
|
|
} |
129 |
|
|
|
130 |
|
|
static int query_formats(AVFilterContext *ctx) |
131 |
|
|
{ |
132 |
|
|
AVFilterFormats *formats; |
133 |
|
|
AVFilterChannelLayouts *layouts; |
134 |
|
|
static const enum AVSampleFormat sample_fmts[] = { |
135 |
|
|
AV_SAMPLE_FMT_DBLP, |
136 |
|
|
AV_SAMPLE_FMT_NONE |
137 |
|
|
}; |
138 |
|
|
int ret; |
139 |
|
|
|
140 |
|
|
layouts = ff_all_channel_counts(); |
141 |
|
|
if (!layouts) |
142 |
|
|
return AVERROR(ENOMEM); |
143 |
|
|
ret = ff_set_common_channel_layouts(ctx, layouts); |
144 |
|
|
if (ret < 0) |
145 |
|
|
return ret; |
146 |
|
|
|
147 |
|
|
formats = ff_make_format_list(sample_fmts); |
148 |
|
|
if (!formats) |
149 |
|
|
return AVERROR(ENOMEM); |
150 |
|
|
ret = ff_set_common_formats(ctx, formats); |
151 |
|
|
if (ret < 0) |
152 |
|
|
return ret; |
153 |
|
|
|
154 |
|
|
formats = ff_all_samplerates(); |
155 |
|
|
if (!formats) |
156 |
|
|
return AVERROR(ENOMEM); |
157 |
|
|
return ff_set_common_samplerates(ctx, formats); |
158 |
|
|
} |
159 |
|
|
|
160 |
|
|
static inline int frame_size(int sample_rate, int frame_len_msec) |
161 |
|
|
{ |
162 |
|
|
const int frame_size = lrint((double)sample_rate * (frame_len_msec / 1000.0)); |
163 |
|
|
return frame_size + (frame_size % 2); |
164 |
|
|
} |
165 |
|
|
|
166 |
|
|
static cqueue *cqueue_create(int size, int max_size) |
167 |
|
|
{ |
168 |
|
|
cqueue *q; |
169 |
|
|
|
170 |
|
|
if (max_size < size) |
171 |
|
|
return NULL; |
172 |
|
|
|
173 |
|
|
q = av_malloc(sizeof(cqueue)); |
174 |
|
|
if (!q) |
175 |
|
|
return NULL; |
176 |
|
|
|
177 |
|
|
q->max_size = max_size; |
178 |
|
|
q->size = size; |
179 |
|
|
q->nb_elements = 0; |
180 |
|
|
|
181 |
|
|
q->elements = av_malloc_array(max_size, sizeof(double)); |
182 |
|
|
if (!q->elements) { |
183 |
|
|
av_free(q); |
184 |
|
|
return NULL; |
185 |
|
|
} |
186 |
|
|
|
187 |
|
|
return q; |
188 |
|
|
} |
189 |
|
|
|
190 |
|
|
static void cqueue_free(cqueue *q) |
191 |
|
|
{ |
192 |
|
|
if (q) |
193 |
|
|
av_free(q->elements); |
194 |
|
|
av_free(q); |
195 |
|
|
} |
196 |
|
|
|
197 |
|
|
static int cqueue_size(cqueue *q) |
198 |
|
|
{ |
199 |
|
|
return q->nb_elements; |
200 |
|
|
} |
201 |
|
|
|
202 |
|
|
static int cqueue_empty(cqueue *q) |
203 |
|
|
{ |
204 |
|
|
return q->nb_elements <= 0; |
205 |
|
|
} |
206 |
|
|
|
207 |
|
|
static int cqueue_enqueue(cqueue *q, double element) |
208 |
|
|
{ |
209 |
|
|
av_assert2(q->nb_elements < q->max_size); |
210 |
|
|
|
211 |
|
|
q->elements[q->nb_elements] = element; |
212 |
|
|
q->nb_elements++; |
213 |
|
|
|
214 |
|
|
return 0; |
215 |
|
|
} |
216 |
|
|
|
217 |
|
|
static double cqueue_peek(cqueue *q, int index) |
218 |
|
|
{ |
219 |
|
|
av_assert2(index < q->nb_elements); |
220 |
|
|
return q->elements[index]; |
221 |
|
|
} |
222 |
|
|
|
223 |
|
|
static int cqueue_dequeue(cqueue *q, double *element) |
224 |
|
|
{ |
225 |
|
|
av_assert2(!cqueue_empty(q)); |
226 |
|
|
|
227 |
|
|
*element = q->elements[0]; |
228 |
|
|
memmove(&q->elements[0], &q->elements[1], (q->nb_elements - 1) * sizeof(double)); |
229 |
|
|
q->nb_elements--; |
230 |
|
|
|
231 |
|
|
return 0; |
232 |
|
|
} |
233 |
|
|
|
234 |
|
|
static int cqueue_pop(cqueue *q) |
235 |
|
|
{ |
236 |
|
|
av_assert2(!cqueue_empty(q)); |
237 |
|
|
|
238 |
|
|
memmove(&q->elements[0], &q->elements[1], (q->nb_elements - 1) * sizeof(double)); |
239 |
|
|
q->nb_elements--; |
240 |
|
|
|
241 |
|
|
return 0; |
242 |
|
|
} |
243 |
|
|
|
244 |
|
|
static void cqueue_resize(cqueue *q, int new_size) |
245 |
|
|
{ |
246 |
|
|
av_assert2(q->max_size >= new_size); |
247 |
|
|
av_assert2(MIN_FILTER_SIZE <= new_size); |
248 |
|
|
|
249 |
|
|
if (new_size > q->nb_elements) { |
250 |
|
|
const int side = (new_size - q->nb_elements) / 2; |
251 |
|
|
|
252 |
|
|
memmove(q->elements + side, q->elements, sizeof(double) * q->nb_elements); |
253 |
|
|
for (int i = 0; i < side; i++) |
254 |
|
|
q->elements[i] = q->elements[side]; |
255 |
|
|
q->nb_elements = new_size - 1 - side; |
256 |
|
|
} else { |
257 |
|
|
int count = (q->size - new_size + 1) / 2; |
258 |
|
|
|
259 |
|
|
while (count-- > 0) |
260 |
|
|
cqueue_pop(q); |
261 |
|
|
} |
262 |
|
|
|
263 |
|
|
q->size = new_size; |
264 |
|
|
} |
265 |
|
|
|
266 |
|
|
static void init_gaussian_filter(DynamicAudioNormalizerContext *s) |
267 |
|
|
{ |
268 |
|
|
double total_weight = 0.0; |
269 |
|
|
const double sigma = (((s->filter_size / 2.0) - 1.0) / 3.0) + (1.0 / 3.0); |
270 |
|
|
double adjust; |
271 |
|
|
int i; |
272 |
|
|
|
273 |
|
|
// Pre-compute constants |
274 |
|
|
const int offset = s->filter_size / 2; |
275 |
|
|
const double c1 = 1.0 / (sigma * sqrt(2.0 * M_PI)); |
276 |
|
|
const double c2 = 2.0 * sigma * sigma; |
277 |
|
|
|
278 |
|
|
// Compute weights |
279 |
|
|
for (i = 0; i < s->filter_size; i++) { |
280 |
|
|
const int x = i - offset; |
281 |
|
|
|
282 |
|
|
s->weights[i] = c1 * exp(-x * x / c2); |
283 |
|
|
total_weight += s->weights[i]; |
284 |
|
|
} |
285 |
|
|
|
286 |
|
|
// Adjust weights |
287 |
|
|
adjust = 1.0 / total_weight; |
288 |
|
|
for (i = 0; i < s->filter_size; i++) { |
289 |
|
|
s->weights[i] *= adjust; |
290 |
|
|
} |
291 |
|
|
} |
292 |
|
|
|
293 |
|
|
static av_cold void uninit(AVFilterContext *ctx) |
294 |
|
|
{ |
295 |
|
|
DynamicAudioNormalizerContext *s = ctx->priv; |
296 |
|
|
int c; |
297 |
|
|
|
298 |
|
|
av_freep(&s->prev_amplification_factor); |
299 |
|
|
av_freep(&s->dc_correction_value); |
300 |
|
|
av_freep(&s->compress_threshold); |
301 |
|
|
|
302 |
|
|
for (c = 0; c < s->channels; c++) { |
303 |
|
|
if (s->gain_history_original) |
304 |
|
|
cqueue_free(s->gain_history_original[c]); |
305 |
|
|
if (s->gain_history_minimum) |
306 |
|
|
cqueue_free(s->gain_history_minimum[c]); |
307 |
|
|
if (s->gain_history_smoothed) |
308 |
|
|
cqueue_free(s->gain_history_smoothed[c]); |
309 |
|
|
if (s->threshold_history) |
310 |
|
|
cqueue_free(s->threshold_history[c]); |
311 |
|
|
} |
312 |
|
|
|
313 |
|
|
av_freep(&s->gain_history_original); |
314 |
|
|
av_freep(&s->gain_history_minimum); |
315 |
|
|
av_freep(&s->gain_history_smoothed); |
316 |
|
|
av_freep(&s->threshold_history); |
317 |
|
|
|
318 |
|
|
cqueue_free(s->is_enabled); |
319 |
|
|
s->is_enabled = NULL; |
320 |
|
|
|
321 |
|
|
av_freep(&s->weights); |
322 |
|
|
|
323 |
|
|
ff_bufqueue_discard_all(&s->queue); |
324 |
|
|
} |
325 |
|
|
|
326 |
|
|
static int config_input(AVFilterLink *inlink) |
327 |
|
|
{ |
328 |
|
|
AVFilterContext *ctx = inlink->dst; |
329 |
|
|
DynamicAudioNormalizerContext *s = ctx->priv; |
330 |
|
|
int c; |
331 |
|
|
|
332 |
|
|
uninit(ctx); |
333 |
|
|
|
334 |
|
|
s->channels = inlink->channels; |
335 |
|
|
s->frame_len = frame_size(inlink->sample_rate, s->frame_len_msec); |
336 |
|
|
av_log(ctx, AV_LOG_DEBUG, "frame len %d\n", s->frame_len); |
337 |
|
|
|
338 |
|
|
s->prev_amplification_factor = av_malloc_array(inlink->channels, sizeof(*s->prev_amplification_factor)); |
339 |
|
|
s->dc_correction_value = av_calloc(inlink->channels, sizeof(*s->dc_correction_value)); |
340 |
|
|
s->compress_threshold = av_calloc(inlink->channels, sizeof(*s->compress_threshold)); |
341 |
|
|
s->gain_history_original = av_calloc(inlink->channels, sizeof(*s->gain_history_original)); |
342 |
|
|
s->gain_history_minimum = av_calloc(inlink->channels, sizeof(*s->gain_history_minimum)); |
343 |
|
|
s->gain_history_smoothed = av_calloc(inlink->channels, sizeof(*s->gain_history_smoothed)); |
344 |
|
|
s->threshold_history = av_calloc(inlink->channels, sizeof(*s->threshold_history)); |
345 |
|
|
s->weights = av_malloc_array(MAX_FILTER_SIZE, sizeof(*s->weights)); |
346 |
|
|
s->is_enabled = cqueue_create(s->filter_size, MAX_FILTER_SIZE); |
347 |
|
|
if (!s->prev_amplification_factor || !s->dc_correction_value || |
348 |
|
|
!s->compress_threshold || |
349 |
|
|
!s->gain_history_original || !s->gain_history_minimum || |
350 |
|
|
!s->gain_history_smoothed || !s->threshold_history || |
351 |
|
|
!s->is_enabled || !s->weights) |
352 |
|
|
return AVERROR(ENOMEM); |
353 |
|
|
|
354 |
|
|
for (c = 0; c < inlink->channels; c++) { |
355 |
|
|
s->prev_amplification_factor[c] = 1.0; |
356 |
|
|
|
357 |
|
|
s->gain_history_original[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE); |
358 |
|
|
s->gain_history_minimum[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE); |
359 |
|
|
s->gain_history_smoothed[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE); |
360 |
|
|
s->threshold_history[c] = cqueue_create(s->filter_size, MAX_FILTER_SIZE); |
361 |
|
|
|
362 |
|
|
if (!s->gain_history_original[c] || !s->gain_history_minimum[c] || |
363 |
|
|
!s->gain_history_smoothed[c] || !s->threshold_history[c]) |
364 |
|
|
return AVERROR(ENOMEM); |
365 |
|
|
} |
366 |
|
|
|
367 |
|
|
init_gaussian_filter(s); |
368 |
|
|
|
369 |
|
|
return 0; |
370 |
|
|
} |
371 |
|
|
|
372 |
|
|
static inline double fade(double prev, double next, int pos, int length) |
373 |
|
|
{ |
374 |
|
|
const double step_size = 1.0 / length; |
375 |
|
|
const double f0 = 1.0 - (step_size * (pos + 1.0)); |
376 |
|
|
const double f1 = 1.0 - f0; |
377 |
|
|
return f0 * prev + f1 * next; |
378 |
|
|
} |
379 |
|
|
|
380 |
|
|
static inline double pow_2(const double value) |
381 |
|
|
{ |
382 |
|
|
return value * value; |
383 |
|
|
} |
384 |
|
|
|
385 |
|
|
static inline double bound(const double threshold, const double val) |
386 |
|
|
{ |
387 |
|
|
const double CONST = 0.8862269254527580136490837416705725913987747280611935; //sqrt(PI) / 2.0 |
388 |
|
|
return erf(CONST * (val / threshold)) * threshold; |
389 |
|
|
} |
390 |
|
|
|
391 |
|
|
static double find_peak_magnitude(AVFrame *frame, int channel) |
392 |
|
|
{ |
393 |
|
|
double max = DBL_EPSILON; |
394 |
|
|
int c, i; |
395 |
|
|
|
396 |
|
|
if (channel == -1) { |
397 |
|
|
for (c = 0; c < frame->channels; c++) { |
398 |
|
|
double *data_ptr = (double *)frame->extended_data[c]; |
399 |
|
|
|
400 |
|
|
for (i = 0; i < frame->nb_samples; i++) |
401 |
|
|
max = FFMAX(max, fabs(data_ptr[i])); |
402 |
|
|
} |
403 |
|
|
} else { |
404 |
|
|
double *data_ptr = (double *)frame->extended_data[channel]; |
405 |
|
|
|
406 |
|
|
for (i = 0; i < frame->nb_samples; i++) |
407 |
|
|
max = FFMAX(max, fabs(data_ptr[i])); |
408 |
|
|
} |
409 |
|
|
|
410 |
|
|
return max; |
411 |
|
|
} |
412 |
|
|
|
413 |
|
|
static double compute_frame_rms(AVFrame *frame, int channel) |
414 |
|
|
{ |
415 |
|
|
double rms_value = 0.0; |
416 |
|
|
int c, i; |
417 |
|
|
|
418 |
|
|
if (channel == -1) { |
419 |
|
|
for (c = 0; c < frame->channels; c++) { |
420 |
|
|
const double *data_ptr = (double *)frame->extended_data[c]; |
421 |
|
|
|
422 |
|
|
for (i = 0; i < frame->nb_samples; i++) { |
423 |
|
|
rms_value += pow_2(data_ptr[i]); |
424 |
|
|
} |
425 |
|
|
} |
426 |
|
|
|
427 |
|
|
rms_value /= frame->nb_samples * frame->channels; |
428 |
|
|
} else { |
429 |
|
|
const double *data_ptr = (double *)frame->extended_data[channel]; |
430 |
|
|
for (i = 0; i < frame->nb_samples; i++) { |
431 |
|
|
rms_value += pow_2(data_ptr[i]); |
432 |
|
|
} |
433 |
|
|
|
434 |
|
|
rms_value /= frame->nb_samples; |
435 |
|
|
} |
436 |
|
|
|
437 |
|
|
return FFMAX(sqrt(rms_value), DBL_EPSILON); |
438 |
|
|
} |
439 |
|
|
|
440 |
|
|
static local_gain get_max_local_gain(DynamicAudioNormalizerContext *s, AVFrame *frame, |
441 |
|
|
int channel) |
442 |
|
|
{ |
443 |
|
|
const double peak_magnitude = find_peak_magnitude(frame, channel); |
444 |
|
|
const double maximum_gain = s->peak_value / peak_magnitude; |
445 |
|
|
const double rms_gain = s->target_rms > DBL_EPSILON ? (s->target_rms / compute_frame_rms(frame, channel)) : DBL_MAX; |
446 |
|
|
local_gain gain; |
447 |
|
|
|
448 |
|
|
gain.threshold = peak_magnitude > s->threshold; |
449 |
|
|
gain.max_gain = bound(s->max_amplification, FFMIN(maximum_gain, rms_gain)); |
450 |
|
|
|
451 |
|
|
return gain; |
452 |
|
|
} |
453 |
|
|
|
454 |
|
|
static double minimum_filter(cqueue *q) |
455 |
|
|
{ |
456 |
|
|
double min = DBL_MAX; |
457 |
|
|
int i; |
458 |
|
|
|
459 |
|
|
for (i = 0; i < cqueue_size(q); i++) { |
460 |
|
|
min = FFMIN(min, cqueue_peek(q, i)); |
461 |
|
|
} |
462 |
|
|
|
463 |
|
|
return min; |
464 |
|
|
} |
465 |
|
|
|
466 |
|
|
static double gaussian_filter(DynamicAudioNormalizerContext *s, cqueue *q, cqueue *tq) |
467 |
|
|
{ |
468 |
|
|
double result = 0.0, tsum = 0.0; |
469 |
|
|
int i; |
470 |
|
|
|
471 |
|
|
for (i = 0; i < cqueue_size(q); i++) { |
472 |
|
|
tsum += cqueue_peek(tq, i) * s->weights[i]; |
473 |
|
|
result += cqueue_peek(q, i) * s->weights[i] * cqueue_peek(tq, i); |
474 |
|
|
} |
475 |
|
|
|
476 |
|
|
if (tsum == 0.0) |
477 |
|
|
result = 1.0; |
478 |
|
|
|
479 |
|
|
return result; |
480 |
|
|
} |
481 |
|
|
|
482 |
|
|
static void update_gain_history(DynamicAudioNormalizerContext *s, int channel, |
483 |
|
|
local_gain gain) |
484 |
|
|
{ |
485 |
|
|
if (cqueue_empty(s->gain_history_original[channel])) { |
486 |
|
|
const int pre_fill_size = s->filter_size / 2; |
487 |
|
|
const double initial_value = s->alt_boundary_mode ? gain.max_gain : s->peak_value; |
488 |
|
|
|
489 |
|
|
s->prev_amplification_factor[channel] = initial_value; |
490 |
|
|
|
491 |
|
|
while (cqueue_size(s->gain_history_original[channel]) < pre_fill_size) { |
492 |
|
|
cqueue_enqueue(s->gain_history_original[channel], initial_value); |
493 |
|
|
cqueue_enqueue(s->threshold_history[channel], gain.threshold); |
494 |
|
|
} |
495 |
|
|
} |
496 |
|
|
|
497 |
|
|
cqueue_enqueue(s->gain_history_original[channel], gain.max_gain); |
498 |
|
|
|
499 |
|
|
while (cqueue_size(s->gain_history_original[channel]) >= s->filter_size) { |
500 |
|
|
double minimum; |
501 |
|
|
|
502 |
|
|
if (cqueue_empty(s->gain_history_minimum[channel])) { |
503 |
|
|
const int pre_fill_size = s->filter_size / 2; |
504 |
|
|
double initial_value = s->alt_boundary_mode ? cqueue_peek(s->gain_history_original[channel], 0) : 1.0; |
505 |
|
|
int input = pre_fill_size; |
506 |
|
|
|
507 |
|
|
while (cqueue_size(s->gain_history_minimum[channel]) < pre_fill_size) { |
508 |
|
|
input++; |
509 |
|
|
initial_value = FFMIN(initial_value, cqueue_peek(s->gain_history_original[channel], input)); |
510 |
|
|
cqueue_enqueue(s->gain_history_minimum[channel], initial_value); |
511 |
|
|
} |
512 |
|
|
} |
513 |
|
|
|
514 |
|
|
minimum = minimum_filter(s->gain_history_original[channel]); |
515 |
|
|
|
516 |
|
|
cqueue_enqueue(s->gain_history_minimum[channel], minimum); |
517 |
|
|
|
518 |
|
|
cqueue_enqueue(s->threshold_history[channel], gain.threshold); |
519 |
|
|
|
520 |
|
|
cqueue_pop(s->gain_history_original[channel]); |
521 |
|
|
} |
522 |
|
|
|
523 |
|
|
while (cqueue_size(s->gain_history_minimum[channel]) >= s->filter_size) { |
524 |
|
|
double smoothed, limit; |
525 |
|
|
|
526 |
|
|
smoothed = gaussian_filter(s, s->gain_history_minimum[channel], s->threshold_history[channel]); |
527 |
|
|
limit = cqueue_peek(s->gain_history_original[channel], 0); |
528 |
|
|
smoothed = FFMIN(smoothed, limit); |
529 |
|
|
|
530 |
|
|
cqueue_enqueue(s->gain_history_smoothed[channel], smoothed); |
531 |
|
|
|
532 |
|
|
cqueue_pop(s->gain_history_minimum[channel]); |
533 |
|
|
cqueue_pop(s->threshold_history[channel]); |
534 |
|
|
} |
535 |
|
|
} |
536 |
|
|
|
537 |
|
|
static inline double update_value(double new, double old, double aggressiveness) |
538 |
|
|
{ |
539 |
|
|
av_assert0((aggressiveness >= 0.0) && (aggressiveness <= 1.0)); |
540 |
|
|
return aggressiveness * new + (1.0 - aggressiveness) * old; |
541 |
|
|
} |
542 |
|
|
|
543 |
|
|
static void perform_dc_correction(DynamicAudioNormalizerContext *s, AVFrame *frame) |
544 |
|
|
{ |
545 |
|
|
const double diff = 1.0 / frame->nb_samples; |
546 |
|
|
int is_first_frame = cqueue_empty(s->gain_history_original[0]); |
547 |
|
|
int c, i; |
548 |
|
|
|
549 |
|
|
for (c = 0; c < s->channels; c++) { |
550 |
|
|
double *dst_ptr = (double *)frame->extended_data[c]; |
551 |
|
|
double current_average_value = 0.0; |
552 |
|
|
double prev_value; |
553 |
|
|
|
554 |
|
|
for (i = 0; i < frame->nb_samples; i++) |
555 |
|
|
current_average_value += dst_ptr[i] * diff; |
556 |
|
|
|
557 |
|
|
prev_value = is_first_frame ? current_average_value : s->dc_correction_value[c]; |
558 |
|
|
s->dc_correction_value[c] = is_first_frame ? current_average_value : update_value(current_average_value, s->dc_correction_value[c], 0.1); |
559 |
|
|
|
560 |
|
|
for (i = 0; i < frame->nb_samples; i++) { |
561 |
|
|
dst_ptr[i] -= fade(prev_value, s->dc_correction_value[c], i, frame->nb_samples); |
562 |
|
|
} |
563 |
|
|
} |
564 |
|
|
} |
565 |
|
|
|
566 |
|
|
static double setup_compress_thresh(double threshold) |
567 |
|
|
{ |
568 |
|
|
if ((threshold > DBL_EPSILON) && (threshold < (1.0 - DBL_EPSILON))) { |
569 |
|
|
double current_threshold = threshold; |
570 |
|
|
double step_size = 1.0; |
571 |
|
|
|
572 |
|
|
while (step_size > DBL_EPSILON) { |
573 |
|
|
while ((llrint((current_threshold + step_size) * (UINT64_C(1) << 63)) > |
574 |
|
|
llrint(current_threshold * (UINT64_C(1) << 63))) && |
575 |
|
|
(bound(current_threshold + step_size, 1.0) <= threshold)) { |
576 |
|
|
current_threshold += step_size; |
577 |
|
|
} |
578 |
|
|
|
579 |
|
|
step_size /= 2.0; |
580 |
|
|
} |
581 |
|
|
|
582 |
|
|
return current_threshold; |
583 |
|
|
} else { |
584 |
|
|
return threshold; |
585 |
|
|
} |
586 |
|
|
} |
587 |
|
|
|
588 |
|
|
static double compute_frame_std_dev(DynamicAudioNormalizerContext *s, |
589 |
|
|
AVFrame *frame, int channel) |
590 |
|
|
{ |
591 |
|
|
double variance = 0.0; |
592 |
|
|
int i, c; |
593 |
|
|
|
594 |
|
|
if (channel == -1) { |
595 |
|
|
for (c = 0; c < s->channels; c++) { |
596 |
|
|
const double *data_ptr = (double *)frame->extended_data[c]; |
597 |
|
|
|
598 |
|
|
for (i = 0; i < frame->nb_samples; i++) { |
599 |
|
|
variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero* |
600 |
|
|
} |
601 |
|
|
} |
602 |
|
|
variance /= (s->channels * frame->nb_samples) - 1; |
603 |
|
|
} else { |
604 |
|
|
const double *data_ptr = (double *)frame->extended_data[channel]; |
605 |
|
|
|
606 |
|
|
for (i = 0; i < frame->nb_samples; i++) { |
607 |
|
|
variance += pow_2(data_ptr[i]); // Assume that MEAN is *zero* |
608 |
|
|
} |
609 |
|
|
variance /= frame->nb_samples - 1; |
610 |
|
|
} |
611 |
|
|
|
612 |
|
|
return FFMAX(sqrt(variance), DBL_EPSILON); |
613 |
|
|
} |
614 |
|
|
|
615 |
|
|
static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame) |
616 |
|
|
{ |
617 |
|
|
int is_first_frame = cqueue_empty(s->gain_history_original[0]); |
618 |
|
|
int c, i; |
619 |
|
|
|
620 |
|
|
if (s->channels_coupled) { |
621 |
|
|
const double standard_deviation = compute_frame_std_dev(s, frame, -1); |
622 |
|
|
const double current_threshold = FFMIN(1.0, s->compress_factor * standard_deviation); |
623 |
|
|
|
624 |
|
|
const double prev_value = is_first_frame ? current_threshold : s->compress_threshold[0]; |
625 |
|
|
double prev_actual_thresh, curr_actual_thresh; |
626 |
|
|
s->compress_threshold[0] = is_first_frame ? current_threshold : update_value(current_threshold, s->compress_threshold[0], (1.0/3.0)); |
627 |
|
|
|
628 |
|
|
prev_actual_thresh = setup_compress_thresh(prev_value); |
629 |
|
|
curr_actual_thresh = setup_compress_thresh(s->compress_threshold[0]); |
630 |
|
|
|
631 |
|
|
for (c = 0; c < s->channels; c++) { |
632 |
|
|
double *const dst_ptr = (double *)frame->extended_data[c]; |
633 |
|
|
for (i = 0; i < frame->nb_samples; i++) { |
634 |
|
|
const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, frame->nb_samples); |
635 |
|
|
dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]); |
636 |
|
|
} |
637 |
|
|
} |
638 |
|
|
} else { |
639 |
|
|
for (c = 0; c < s->channels; c++) { |
640 |
|
|
const double standard_deviation = compute_frame_std_dev(s, frame, c); |
641 |
|
|
const double current_threshold = setup_compress_thresh(FFMIN(1.0, s->compress_factor * standard_deviation)); |
642 |
|
|
|
643 |
|
|
const double prev_value = is_first_frame ? current_threshold : s->compress_threshold[c]; |
644 |
|
|
double prev_actual_thresh, curr_actual_thresh; |
645 |
|
|
double *dst_ptr; |
646 |
|
|
s->compress_threshold[c] = is_first_frame ? current_threshold : update_value(current_threshold, s->compress_threshold[c], 1.0/3.0); |
647 |
|
|
|
648 |
|
|
prev_actual_thresh = setup_compress_thresh(prev_value); |
649 |
|
|
curr_actual_thresh = setup_compress_thresh(s->compress_threshold[c]); |
650 |
|
|
|
651 |
|
|
dst_ptr = (double *)frame->extended_data[c]; |
652 |
|
|
for (i = 0; i < frame->nb_samples; i++) { |
653 |
|
|
const double localThresh = fade(prev_actual_thresh, curr_actual_thresh, i, frame->nb_samples); |
654 |
|
|
dst_ptr[i] = copysign(bound(localThresh, fabs(dst_ptr[i])), dst_ptr[i]); |
655 |
|
|
} |
656 |
|
|
} |
657 |
|
|
} |
658 |
|
|
} |
659 |
|
|
|
660 |
|
|
static void analyze_frame(DynamicAudioNormalizerContext *s, AVFrame *frame) |
661 |
|
|
{ |
662 |
|
|
if (s->dc_correction) { |
663 |
|
|
perform_dc_correction(s, frame); |
664 |
|
|
} |
665 |
|
|
|
666 |
|
|
if (s->compress_factor > DBL_EPSILON) { |
667 |
|
|
perform_compression(s, frame); |
668 |
|
|
} |
669 |
|
|
|
670 |
|
|
if (s->channels_coupled) { |
671 |
|
|
const local_gain gain = get_max_local_gain(s, frame, -1); |
672 |
|
|
int c; |
673 |
|
|
|
674 |
|
|
for (c = 0; c < s->channels; c++) |
675 |
|
|
update_gain_history(s, c, gain); |
676 |
|
|
} else { |
677 |
|
|
int c; |
678 |
|
|
|
679 |
|
|
for (c = 0; c < s->channels; c++) |
680 |
|
|
update_gain_history(s, c, get_max_local_gain(s, frame, c)); |
681 |
|
|
} |
682 |
|
|
} |
683 |
|
|
|
684 |
|
|
static void amplify_frame(DynamicAudioNormalizerContext *s, AVFrame *frame, int enabled) |
685 |
|
|
{ |
686 |
|
|
int c, i; |
687 |
|
|
|
688 |
|
|
for (c = 0; c < s->channels; c++) { |
689 |
|
|
double *dst_ptr = (double *)frame->extended_data[c]; |
690 |
|
|
double current_amplification_factor; |
691 |
|
|
|
692 |
|
|
cqueue_dequeue(s->gain_history_smoothed[c], ¤t_amplification_factor); |
693 |
|
|
|
694 |
|
|
for (i = 0; i < frame->nb_samples && enabled; i++) { |
695 |
|
|
const double amplification_factor = fade(s->prev_amplification_factor[c], |
696 |
|
|
current_amplification_factor, i, |
697 |
|
|
frame->nb_samples); |
698 |
|
|
|
699 |
|
|
dst_ptr[i] *= amplification_factor; |
700 |
|
|
} |
701 |
|
|
|
702 |
|
|
s->prev_amplification_factor[c] = current_amplification_factor; |
703 |
|
|
} |
704 |
|
|
} |
705 |
|
|
|
706 |
|
|
static int filter_frame(AVFilterLink *inlink, AVFrame *in) |
707 |
|
|
{ |
708 |
|
|
AVFilterContext *ctx = inlink->dst; |
709 |
|
|
DynamicAudioNormalizerContext *s = ctx->priv; |
710 |
|
|
AVFilterLink *outlink = ctx->outputs[0]; |
711 |
|
|
int ret = 1; |
712 |
|
|
|
713 |
|
|
while (((s->queue.available >= s->filter_size) || |
714 |
|
|
(s->eof && s->queue.available)) && |
715 |
|
|
!cqueue_empty(s->gain_history_smoothed[0])) { |
716 |
|
|
AVFrame *out = ff_bufqueue_get(&s->queue); |
717 |
|
|
double is_enabled; |
718 |
|
|
|
719 |
|
|
cqueue_dequeue(s->is_enabled, &is_enabled); |
720 |
|
|
|
721 |
|
|
amplify_frame(s, out, is_enabled > 0.); |
722 |
|
|
ret = ff_filter_frame(outlink, out); |
723 |
|
|
} |
724 |
|
|
|
725 |
|
|
av_frame_make_writable(in); |
726 |
|
|
analyze_frame(s, in); |
727 |
|
|
if (!s->eof) { |
728 |
|
|
ff_bufqueue_add(ctx, &s->queue, in); |
729 |
|
|
cqueue_enqueue(s->is_enabled, !ctx->is_disabled); |
730 |
|
|
} else { |
731 |
|
|
av_frame_free(&in); |
732 |
|
|
} |
733 |
|
|
|
734 |
|
|
return ret; |
735 |
|
|
} |
736 |
|
|
|
737 |
|
|
static int flush_buffer(DynamicAudioNormalizerContext *s, AVFilterLink *inlink, |
738 |
|
|
AVFilterLink *outlink) |
739 |
|
|
{ |
740 |
|
|
AVFrame *out = ff_get_audio_buffer(outlink, s->frame_len); |
741 |
|
|
int c, i; |
742 |
|
|
|
743 |
|
|
if (!out) |
744 |
|
|
return AVERROR(ENOMEM); |
745 |
|
|
|
746 |
|
|
for (c = 0; c < s->channels; c++) { |
747 |
|
|
double *dst_ptr = (double *)out->extended_data[c]; |
748 |
|
|
|
749 |
|
|
for (i = 0; i < out->nb_samples; i++) { |
750 |
|
|
dst_ptr[i] = s->alt_boundary_mode ? DBL_EPSILON : ((s->target_rms > DBL_EPSILON) ? FFMIN(s->peak_value, s->target_rms) : s->peak_value); |
751 |
|
|
if (s->dc_correction) { |
752 |
|
|
dst_ptr[i] *= ((i % 2) == 1) ? -1 : 1; |
753 |
|
|
dst_ptr[i] += s->dc_correction_value[c]; |
754 |
|
|
} |
755 |
|
|
} |
756 |
|
|
} |
757 |
|
|
|
758 |
|
|
return filter_frame(inlink, out); |
759 |
|
|
} |
760 |
|
|
|
761 |
|
|
static int flush(AVFilterLink *outlink) |
762 |
|
|
{ |
763 |
|
|
AVFilterContext *ctx = outlink->src; |
764 |
|
|
DynamicAudioNormalizerContext *s = ctx->priv; |
765 |
|
|
int ret = 0; |
766 |
|
|
|
767 |
|
|
if (!cqueue_empty(s->gain_history_smoothed[0])) { |
768 |
|
|
ret = flush_buffer(s, ctx->inputs[0], outlink); |
769 |
|
|
} else if (s->queue.available) { |
770 |
|
|
AVFrame *out = ff_bufqueue_get(&s->queue); |
771 |
|
|
|
772 |
|
|
s->pts = out->pts; |
773 |
|
|
ret = ff_filter_frame(outlink, out); |
774 |
|
|
} |
775 |
|
|
|
776 |
|
|
return ret; |
777 |
|
|
} |
778 |
|
|
|
779 |
|
|
static int activate(AVFilterContext *ctx) |
780 |
|
|
{ |
781 |
|
|
AVFilterLink *inlink = ctx->inputs[0]; |
782 |
|
|
AVFilterLink *outlink = ctx->outputs[0]; |
783 |
|
|
DynamicAudioNormalizerContext *s = ctx->priv; |
784 |
|
|
AVFrame *in = NULL; |
785 |
|
|
int ret = 0, status; |
786 |
|
|
int64_t pts; |
787 |
|
|
|
788 |
|
|
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); |
789 |
|
|
|
790 |
|
|
if (!s->eof) { |
791 |
|
|
ret = ff_inlink_consume_samples(inlink, s->frame_len, s->frame_len, &in); |
792 |
|
|
if (ret < 0) |
793 |
|
|
return ret; |
794 |
|
|
if (ret > 0) { |
795 |
|
|
ret = filter_frame(inlink, in); |
796 |
|
|
if (ret <= 0) |
797 |
|
|
return ret; |
798 |
|
|
} |
799 |
|
|
|
800 |
|
|
if (ff_inlink_queued_samples(inlink) >= s->frame_len) { |
801 |
|
|
ff_filter_set_ready(ctx, 10); |
802 |
|
|
return 0; |
803 |
|
|
} |
804 |
|
|
} |
805 |
|
|
|
806 |
|
|
if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) { |
807 |
|
|
if (status == AVERROR_EOF) |
808 |
|
|
s->eof = 1; |
809 |
|
|
} |
810 |
|
|
|
811 |
|
|
if (s->eof && s->queue.available) |
812 |
|
|
return flush(outlink); |
813 |
|
|
|
814 |
|
|
if (s->eof && !s->queue.available) { |
815 |
|
|
ff_outlink_set_status(outlink, AVERROR_EOF, s->pts); |
816 |
|
|
return 0; |
817 |
|
|
} |
818 |
|
|
|
819 |
|
|
if (!s->eof) |
820 |
|
|
FF_FILTER_FORWARD_WANTED(outlink, inlink); |
821 |
|
|
|
822 |
|
|
return FFERROR_NOT_READY; |
823 |
|
|
} |
824 |
|
|
|
825 |
|
|
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, |
826 |
|
|
char *res, int res_len, int flags) |
827 |
|
|
{ |
828 |
|
|
DynamicAudioNormalizerContext *s = ctx->priv; |
829 |
|
|
AVFilterLink *inlink = ctx->inputs[0]; |
830 |
|
|
int prev_filter_size = s->filter_size; |
831 |
|
|
int ret; |
832 |
|
|
|
833 |
|
|
ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags); |
834 |
|
|
if (ret < 0) |
835 |
|
|
return ret; |
836 |
|
|
|
837 |
|
|
s->filter_size |= 1; |
838 |
|
|
if (prev_filter_size != s->filter_size) { |
839 |
|
|
init_gaussian_filter(s); |
840 |
|
|
|
841 |
|
|
for (int c = 0; c < s->channels; c++) { |
842 |
|
|
cqueue_resize(s->gain_history_original[c], s->filter_size); |
843 |
|
|
cqueue_resize(s->gain_history_minimum[c], s->filter_size); |
844 |
|
|
cqueue_resize(s->threshold_history[c], s->filter_size); |
845 |
|
|
} |
846 |
|
|
} |
847 |
|
|
|
848 |
|
|
s->frame_len = frame_size(inlink->sample_rate, s->frame_len_msec); |
849 |
|
|
|
850 |
|
|
return 0; |
851 |
|
|
} |
852 |
|
|
|
853 |
|
|
static const AVFilterPad avfilter_af_dynaudnorm_inputs[] = { |
854 |
|
|
{ |
855 |
|
|
.name = "default", |
856 |
|
|
.type = AVMEDIA_TYPE_AUDIO, |
857 |
|
|
.config_props = config_input, |
858 |
|
|
}, |
859 |
|
|
{ NULL } |
860 |
|
|
}; |
861 |
|
|
|
862 |
|
|
static const AVFilterPad avfilter_af_dynaudnorm_outputs[] = { |
863 |
|
|
{ |
864 |
|
|
.name = "default", |
865 |
|
|
.type = AVMEDIA_TYPE_AUDIO, |
866 |
|
|
}, |
867 |
|
|
{ NULL } |
868 |
|
|
}; |
869 |
|
|
|
870 |
|
|
AVFilter ff_af_dynaudnorm = { |
871 |
|
|
.name = "dynaudnorm", |
872 |
|
|
.description = NULL_IF_CONFIG_SMALL("Dynamic Audio Normalizer."), |
873 |
|
|
.query_formats = query_formats, |
874 |
|
|
.priv_size = sizeof(DynamicAudioNormalizerContext), |
875 |
|
|
.init = init, |
876 |
|
|
.uninit = uninit, |
877 |
|
|
.activate = activate, |
878 |
|
|
.inputs = avfilter_af_dynaudnorm_inputs, |
879 |
|
|
.outputs = avfilter_af_dynaudnorm_outputs, |
880 |
|
|
.priv_class = &dynaudnorm_class, |
881 |
|
|
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL, |
882 |
|
|
.process_command = process_command, |
883 |
|
|
}; |