avfilter/af_asoftclip: rewrite oversampling

Fixes most aliasing issues.

Paul B Mahol committed 5 years ago

Commit 94d4cc21 parent 3e127b52 files · +171 −135Browse files

Showing the changes to libavfilter (1 of 2 files). Show all

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@@ -21,11 +21,12 @@ · 20 unmodified lines
21	21	#include "libavutil/avassert.h"
22	22	#include "libavutil/channel_layout.h"
23	23	#include "libavutil/opt.h"
24		−#include "libswresample/swresample.h"
25	24	#include "avfilter.h"
26	25	#include "audio.h"
27	26	#include "formats.h"
28	27
	28	+#define MAX_OVERSAMPLE 64
	29	+
29	30	enum ASoftClipTypes {
30	31	ASC_HARD = -1,
31	32	ASC_TANH,
@@ -39,6 +40,14 @@ enum ASoftClipTypes { · 7 unmodified lines
39	40	NB_TYPES,
40	41	};
41	42
	43	+typedef struct Lowpass {
	44	+ float fb0, fb1, fb2;
	45	+ float fa0, fa1, fa2;
	46	+
	47	+ double db0, db1, db2;
	48	+ double da0, da1, da2;
	49	+} Lowpass;
	50	+
42	51	typedef struct ASoftClipContext {
43	52	const AVClass *class;
44	53
@@ -49,10 +58,8 @@ typedef struct ASoftClipContext { · 4 unmodified lines
49	58	double output;
50	59	double param;
51	60
52		− SwrContext *up_ctx;
53		− SwrContext *down_ctx;
54		−
55		− AVFrame *frame;
	61	+ Lowpass lowpass[MAX_OVERSAMPLE];
	62	+ AVFrame *frame[2];
56	63
57	64	void (filter)(struct ASoftClipContext s, void dst, const void src,
58	65	int nb_samples, int channels, int start, int end);
@@ -60,7 +67,6 @@ typedef struct ASoftClipContext { · 1 unmodified line
60	67
61	68	#define OFFSET(x) offsetof(ASoftClipContext, x)
62	69	#define A AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM\|AV_OPT_FLAG_RUNTIME_PARAM
63		−#define F AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM
64	70
65	71	static const AVOption asoftclip_options[] = {
66	72	{ "type", "set softclip type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=0}, -1, NB_TYPES-1, A, "types" },
@@ -76,7 +82,7 @@ static const AVOption asoftclip_options[] = { · 9 unmodified lines
76	82	{ "threshold", "set softclip threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 1, A },
77	83	{ "output", "set softclip output gain", OFFSET(output), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 16, A },
78	84	{ "param", "set softclip parameter", OFFSET(param), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01, 3, A },
79		− { "oversample", "set oversample factor", OFFSET(oversample), AV_OPT_TYPE_INT, {.i64=1}, 1, 32, F },
	85	+ { "oversample", "set oversample factor", OFFSET(oversample), AV_OPT_TYPE_INT, {.i64=1}, 1, MAX_OVERSAMPLE, A },
80	86	{ NULL }
81	87	};
82	88
@@ -85,8 +91,7 @@ AVFILTER_DEFINE_CLASS(asoftclip); · 2 unmodified lines
85	91	static int query_formats(AVFilterContext *ctx)
86	92	{
87	93	static const enum AVSampleFormat sample_fmts[] = {
88		− AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
89		− AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
	94	+ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
90	95	AV_SAMPLE_FMT_NONE
91	96	};
92	97	int ret = ff_set_common_formats_from_list(ctx, sample_fmts);
@@ -100,42 +105,103 @@ static int query_formats(AVFilterContext *ctx) · 7 unmodified lines
100	105	return ff_set_common_all_samplerates(ctx);
101	106	}
102	107
	108	+static void get_lowpass(Lowpass *s,
	109	+ double frequency,
	110	+ double sample_rate)
	111	+{
	112	+ double w0 = 2 * M_PI * frequency / sample_rate;
	113	+ double alpha = sin(w0) / (2 * 0.8);
	114	+ double factor;
	115	+
	116	+ s->da0 = 1 + alpha;
	117	+ s->da1 = -2 * cos(w0);
	118	+ s->da2 = 1 - alpha;
	119	+ s->db0 = (1 - cos(w0)) / 2;
	120	+ s->db1 = 1 - cos(w0);
	121	+ s->db2 = (1 - cos(w0)) / 2;
	122	+
	123	+ s->da1 /= s->da0;
	124	+ s->da2 /= s->da0;
	125	+ s->db0 /= s->da0;
	126	+ s->db1 /= s->da0;
	127	+ s->db2 /= s->da0;
	128	+ s->da0 /= s->da0;
	129	+
	130	+ factor = (s->da0 + s->da1 + s->da2) / (s->db0 + s->db1 + s->db2);
	131	+ s->db0 *= factor;
	132	+ s->db1 *= factor;
	133	+ s->db2 *= factor;
	134	+
	135	+ s->fa0 = s->da0;
	136	+ s->fa1 = s->da1;
	137	+ s->fa2 = s->da2;
	138	+ s->fb0 = s->db0;
	139	+ s->fb1 = s->db1;
	140	+ s->fb2 = s->db2;
	141	+}
	142	+
	143	+static inline float run_lowpassf(const Lowpass *const s,
	144	+ float src, float *w)
	145	+{
	146	+ float dst;
	147	+
	148	+ dst = src * s->fb0 + w[0];
	149	+ w[0] = s->fb1 * src + w[1] - s->fa1 * dst;
	150	+ w[1] = s->fb2 * src - s->fa2 * dst;
	151	+
	152	+ return dst;
	153	+}
	154	+
103	155	static void filter_flt(ASoftClipContext *s,
104	156	void dptr, const void sptr,
105	157	int nb_samples, int channels,
106	158	int start, int end)
107	159	{
	160	+ const int oversample = s->oversample;
	161	+ const int nb_osamples = nb_samples * oversample;
	162	+ const float scale = oversample > 1 ? oversample * 0.5f : 1.f;
108	163	float threshold = s->threshold;
109	164	float gain = s->output * threshold;
110	165	float factor = 1.f / threshold;
111	166	float param = s->param;
112	167
113	168	for (int c = start; c < end; c++) {
	169	+ float w = (float )(s->frame[0]->extended_data[c]) + 2 * (oversample - 1);
114	170	const float *src = sptr[c];
115	171	float *dst = dptr[c];
116	172
	173	+ for (int n = 0; n < nb_samples; n++) {
	174	+ dst[oversample * n] = src[n];
	175	+
	176	+ for (int m = 1; m < oversample; m++)
	177	+ dst[oversample * n + m] = 0.f;
	178	+ }
	179	+
	180	+ for (int n = 0; n < nb_osamples && oversample > 1; n++)
	181	+ dst[n] = run_lowpassf(&s->lowpass[oversample - 1], dst[n], w);
	182	+
117	183	switch (s->type) {
118	184	case ASC_HARD:
119		− for (int n = 0; n < nb_samples; n++) {
120		− dst[n] = av_clipf(src[n] * factor, -1.f, 1.f);
	185	+ for (int n = 0; n < nb_osamples; n++) {
	186	+ dst[n] = av_clipf(dst[n] * factor, -1.f, 1.f);
121	187	dst[n] *= gain;
122	188	}
123	189	break;
124	190	case ASC_TANH:
125		− for (int n = 0; n < nb_samples; n++) {
126		− dst[n] = tanhf(src[n] * factor * param);
	191	+ for (int n = 0; n < nb_osamples; n++) {
	192	+ dst[n] = tanhf(dst[n] * factor * param);
127	193	dst[n] *= gain;
128	194	}
129	195	break;
130	196	case ASC_ATAN:
131		− for (int n = 0; n < nb_samples; n++) {
132		− dst[n] = 2.f / M_PI * atanf(src[n] * factor * param);
	197	+ for (int n = 0; n < nb_osamples; n++) {
	198	+ dst[n] = 2.f / M_PI * atanf(dst[n] * factor * param);
133	199	dst[n] *= gain;
134	200	}
135	201	break;
136	202	case ASC_CUBIC:
137		− for (int n = 0; n < nb_samples; n++) {
138		− float sample = src[n] * factor;
	203	+ for (int n = 0; n < nb_osamples; n++) {
	204	+ float sample = dst[n] * factor;
139	205
140	206	if (FFABS(sample) >= 1.5f)
141	207	dst[n] = FFSIGN(sample);
@@ -145,22 +211,22 @@ static void filter_flt(ASoftClipContext *s, · 3 unmodified lines
145	211	}
146	212	break;
147	213	case ASC_EXP:
148		− for (int n = 0; n < nb_samples; n++) {
149		− dst[n] = 2.f / (1.f + expf(-2.f * src[n] * factor)) - 1.;
	214	+ for (int n = 0; n < nb_osamples; n++) {
	215	+ dst[n] = 2.f / (1.f + expf(-2.f * dst[n] * factor)) - 1.;
150	216	dst[n] *= gain;
151	217	}
152	218	break;
153	219	case ASC_ALG:
154		− for (int n = 0; n < nb_samples; n++) {
155		− float sample = src[n] * factor;
	220	+ for (int n = 0; n < nb_osamples; n++) {
	221	+ float sample = dst[n] * factor;
156	222
157	223	dst[n] = sample / (sqrtf(param + sample * sample));
158	224	dst[n] *= gain;
159	225	}
160	226	break;
161	227	case ASC_QUINTIC:
162		− for (int n = 0; n < nb_samples; n++) {
163		− float sample = src[n] * factor;
	228	+ for (int n = 0; n < nb_osamples; n++) {
	229	+ float sample = dst[n] * factor;
164	230
165	231	if (FFABS(sample) >= 1.25)
166	232	dst[n] = FFSIGN(sample);
@@ -170,8 +236,8 @@ static void filter_flt(ASoftClipContext *s, · 3 unmodified lines
170	236	}
171	237	break;
172	238	case ASC_SIN:
173		− for (int n = 0; n < nb_samples; n++) {
174		− float sample = src[n] * factor;
	239	+ for (int n = 0; n < nb_osamples; n++) {
	240	+ float sample = dst[n] * factor;
175	241
176	242	if (FFABS(sample) >= M_PI_2)
177	243	dst[n] = FFSIGN(sample);
@@ -181,53 +247,86 @@ static void filter_flt(ASoftClipContext *s, · 3 unmodified lines
181	247	}
182	248	break;
183	249	case ASC_ERF:
184		− for (int n = 0; n < nb_samples; n++) {
185		− dst[n] = erff(src[n] * factor);
	250	+ for (int n = 0; n < nb_osamples; n++) {
	251	+ dst[n] = erff(dst[n] * factor);
186	252	dst[n] *= gain;
187	253	}
188	254	break;
189	255	default:
190	256	av_assert0(0);
191	257	}
	258	+
	259	+ w = (float )(s->frame[1]->extended_data[c]) + 2 (oversample - 1);
	260	+ for (int n = 0; n < nb_osamples && oversample > 1; n++)
	261	+ dst[n] = run_lowpassf(&s->lowpass[oversample - 1], dst[n], w);
	262	+
	263	+ for (int n = 0; n < nb_samples; n++)
	264	+ dst[n] = dst[n * oversample] * scale;
192	265	}
193	266	}
194	267
	268	+static inline double run_lowpassd(const Lowpass *const s,
	269	+ double src, double *w)
	270	+{
	271	+ double dst;
	272	+
	273	+ dst = src * s->db0 + w[0];
	274	+ w[0] = s->db1 * src + w[1] - s->da1 * dst;
	275	+ w[1] = s->db2 * src - s->da2 * dst;
	276	+
	277	+ return dst;
	278	+}
	279	+
195	280	static void filter_dbl(ASoftClipContext *s,
196	281	void dptr, const void sptr,
197	282	int nb_samples, int channels,
198	283	int start, int end)
199	284	{
	285	+ const int oversample = s->oversample;
	286	+ const int nb_osamples = nb_samples * oversample;
	287	+ const double scale = oversample > 1 ? oversample * 0.5 : 1.;
200	288	double threshold = s->threshold;
201	289	double gain = s->output * threshold;
202	290	double factor = 1. / threshold;
203	291	double param = s->param;
204	292
205	293	for (int c = start; c < end; c++) {
	294	+ double w = (double )(s->frame[0]->extended_data[c]) + 2 * (oversample - 1);
206	295	const double *src = sptr[c];
207	296	double *dst = dptr[c];
208	297
	298	+ for (int n = 0; n < nb_samples; n++) {
	299	+ dst[oversample * n] = src[n];
	300	+
	301	+ for (int m = 1; m < oversample; m++)
	302	+ dst[oversample * n + m] = 0.f;
	303	+ }
	304	+
	305	+ for (int n = 0; n < nb_osamples && oversample > 1; n++)
	306	+ dst[n] = run_lowpassd(&s->lowpass[oversample - 1], dst[n], w);
	307	+
209	308	switch (s->type) {
210	309	case ASC_HARD:
211		− for (int n = 0; n < nb_samples; n++) {
212		− dst[n] = av_clipd(src[n] * factor, -1., 1.);
	310	+ for (int n = 0; n < nb_osamples; n++) {
	311	+ dst[n] = av_clipd(dst[n] * factor, -1., 1.);
213	312	dst[n] *= gain;
214	313	}
215	314	break;
216	315	case ASC_TANH:
217		− for (int n = 0; n < nb_samples; n++) {
218		− dst[n] = tanh(src[n] * factor * param);
	316	+ for (int n = 0; n < nb_osamples; n++) {
	317	+ dst[n] = tanh(dst[n] * factor * param);
219	318	dst[n] *= gain;
220	319	}
221	320	break;
222	321	case ASC_ATAN:
223		− for (int n = 0; n < nb_samples; n++) {
224		− dst[n] = 2. / M_PI * atan(src[n] * factor * param);
	322	+ for (int n = 0; n < nb_osamples; n++) {
	323	+ dst[n] = 2. / M_PI * atan(dst[n] * factor * param);
225	324	dst[n] *= gain;
226	325	}
227	326	break;
228	327	case ASC_CUBIC:
229		− for (int n = 0; n < nb_samples; n++) {
230		− double sample = src[n] * factor;
	328	+ for (int n = 0; n < nb_osamples; n++) {
	329	+ double sample = dst[n] * factor;
231	330
232	331	if (FFABS(sample) >= 1.5)
233	332	dst[n] = FFSIGN(sample);
@@ -237,22 +336,22 @@ static void filter_dbl(ASoftClipContext *s, · 3 unmodified lines
237	336	}
238	337	break;
239	338	case ASC_EXP:
240		− for (int n = 0; n < nb_samples; n++) {
241		− dst[n] = 2. / (1. + exp(-2. * src[n] * factor)) - 1.;
	339	+ for (int n = 0; n < nb_osamples; n++) {
	340	+ dst[n] = 2. / (1. + exp(-2. * dst[n] * factor)) - 1.;
242	341	dst[n] *= gain;
243	342	}
244	343	break;
245	344	case ASC_ALG:
246		− for (int n = 0; n < nb_samples; n++) {
247		− double sample = src[n] * factor;
	345	+ for (int n = 0; n < nb_osamples; n++) {
	346	+ double sample = dst[n] * factor;
248	347
249	348	dst[n] = sample / (sqrt(param + sample * sample));
250	349	dst[n] *= gain;
251	350	}
252	351	break;
253	352	case ASC_QUINTIC:
254		− for (int n = 0; n < nb_samples; n++) {
255		− double sample = src[n] * factor;
	353	+ for (int n = 0; n < nb_osamples; n++) {
	354	+ double sample = dst[n] * factor;
256	355
257	356	if (FFABS(sample) >= 1.25)
258	357	dst[n] = FFSIGN(sample);
@@ -262,8 +361,8 @@ static void filter_dbl(ASoftClipContext *s, · 3 unmodified lines
262	361	}
263	362	break;
264	363	case ASC_SIN:
265		− for (int n = 0; n < nb_samples; n++) {
266		− double sample = src[n] * factor;
	364	+ for (int n = 0; n < nb_osamples; n++) {
	365	+ double sample = dst[n] * factor;
267	366
268	367	if (FFABS(sample) >= M_PI_2)
269	368	dst[n] = FFSIGN(sample);
@@ -273,14 +372,21 @@ static void filter_dbl(ASoftClipContext *s, · 3 unmodified lines
273	372	}
274	373	break;
275	374	case ASC_ERF:
276		− for (int n = 0; n < nb_samples; n++) {
277		− dst[n] = erf(src[n] * factor);
	375	+ for (int n = 0; n < nb_osamples; n++) {
	376	+ dst[n] = erf(dst[n] * factor);
278	377	dst[n] *= gain;
279	378	}
280	379	break;
281	380	default:
282	381	av_assert0(0);
283	382	}
	383	+
	384	+ w = (double )(s->frame[1]->extended_data[c]) + 2 (oversample - 1);
	385	+ for (int n = 0; n < nb_osamples && oversample > 1; n++)
	386	+ dst[n] = run_lowpassd(&s->lowpass[oversample - 1], dst[n], w);
	387	+
	388	+ for (int n = 0; n < nb_samples; n++)
	389	+ dst[n] = dst[n * oversample] * scale;
284	390	}
285	391	}
286	392
@@ -288,47 +394,21 @@ static int config_input(AVFilterLink *inlink) · 1 unmodified line
288	394	{
289	395	AVFilterContext *ctx = inlink->dst;
290	396	ASoftClipContext *s = ctx->priv;
291		− int ret;
292	397
293	398	switch (inlink->format) {
294		− case AV_SAMPLE_FMT_FLT:
295	399	case AV_SAMPLE_FMT_FLTP: s->filter = filter_flt; break;
296		− case AV_SAMPLE_FMT_DBL:
297	400	case AV_SAMPLE_FMT_DBLP: s->filter = filter_dbl; break;
298	401	default: av_assert0(0);
299	402	}
300	403
301		− if (s->oversample <= 1)
302		− return 0;
303		−
304		− s->up_ctx = swr_alloc();
305		− s->down_ctx = swr_alloc();
306		− if (!s->up_ctx \|\| !s->down_ctx)
	404	+ s->frame[0] = ff_get_audio_buffer(inlink, 2 * MAX_OVERSAMPLE);
	405	+ s->frame[1] = ff_get_audio_buffer(inlink, 2 * MAX_OVERSAMPLE);
	406	+ if (!s->frame[0] \|\| !s->frame[1])
307	407	return AVERROR(ENOMEM);
308	408
309		− av_opt_set_int(s->up_ctx, "in_channel_layout", inlink->channel_layout, 0);
310		− av_opt_set_int(s->up_ctx, "in_sample_rate", inlink->sample_rate, 0);
311		− av_opt_set_sample_fmt(s->up_ctx, "in_sample_fmt", inlink->format, 0);
312		−
313		− av_opt_set_int(s->up_ctx, "out_channel_layout", inlink->channel_layout, 0);
314		− av_opt_set_int(s->up_ctx, "out_sample_rate", inlink->sample_rate * s->oversample, 0);
315		− av_opt_set_sample_fmt(s->up_ctx, "out_sample_fmt", inlink->format, 0);
316		−
317		− av_opt_set_int(s->down_ctx, "in_channel_layout", inlink->channel_layout, 0);
318		− av_opt_set_int(s->down_ctx, "in_sample_rate", inlink->sample_rate * s->oversample, 0);
319		− av_opt_set_sample_fmt(s->down_ctx, "in_sample_fmt", inlink->format, 0);
320		−
321		− av_opt_set_int(s->down_ctx, "out_channel_layout", inlink->channel_layout, 0);
322		− av_opt_set_int(s->down_ctx, "out_sample_rate", inlink->sample_rate, 0);
323		− av_opt_set_sample_fmt(s->down_ctx, "out_sample_fmt", inlink->format, 0);
324		−
325		− ret = swr_init(s->up_ctx);
326		− if (ret < 0)
327		− return ret;
328		−
329		− ret = swr_init(s->down_ctx);
330		− if (ret < 0)
331		− return ret;
	409	+ for (int i = 0; i < MAX_OVERSAMPLE; i++) {
	410	+ get_lowpass(&s->lowpass[i], inlink->sample_rate / 2, inlink->sample_rate * (i + 1));
	411	+ }
332	412
333	413	return 0;
334	414	}
@@ -361,14 +441,14 @@ static int filter_frame(AVFilterLink inlink, AVFrame in) · 26 unmodified lines
361	441	AVFilterContext *ctx = inlink->dst;
362	442	ASoftClipContext *s = ctx->priv;
363	443	AVFilterLink *outlink = ctx->outputs[0];
364		− int ret, nb_samples, channels;
	444	+ int nb_samples, channels;
365	445	ThreadData td;
366	446	AVFrame *out;
367	447
368		− if (av_frame_is_writable(in)) {
	448	+ if (av_frame_is_writable(in) && s->oversample == 1) {
369	449	out = in;
370	450	} else {
371		− out = ff_get_audio_buffer(outlink, in->nb_samples);
	451	+ out = ff_get_audio_buffer(outlink, in->nb_samples * s->oversample);
372	452	if (!out) {
373	453	av_frame_free(&in);
374	454	return AVERROR(ENOMEM);
@@ -376,72 +456,29 @@ static int filter_frame(AVFilterLink inlink, AVFrame in) · 1 unmodified line
376	456	av_frame_copy_props(out, in);
377	457	}
378	458
379		− if (av_sample_fmt_is_planar(in->format)) {
380		− nb_samples = in->nb_samples;
381		− channels = in->channels;
382		− } else {
383		− nb_samples = in->channels * in->nb_samples;
384		− channels = 1;
385		− }
386		−
387		− if (s->oversample > 1) {
388		− s->frame = ff_get_audio_buffer(outlink, in->nb_samples * s->oversample);
389		− if (!s->frame) {
390		− ret = AVERROR(ENOMEM);
391		− goto fail;
392		− }
	459	+ nb_samples = in->nb_samples;
	460	+ channels = in->channels;
393	461
394		− ret = swr_convert(s->up_ctx, (uint8_t*)s->frame->extended_data, in->nb_samples s->oversample,
395		− (const uint8_t **)in->extended_data, in->nb_samples);
396		− if (ret < 0)
397		− goto fail;
398		−
399		− td.in = s->frame;
400		− td.out = s->frame;
401		− td.nb_samples = av_sample_fmt_is_planar(in->format) ? ret : ret * in->channels;
402		− td.channels = channels;
403		− ff_filter_execute(ctx, filter_channels, &td, NULL,
404		− FFMIN(channels, ff_filter_get_nb_threads(ctx)));
405		−
406		− ret = swr_convert(s->down_ctx, (uint8_t**)out->extended_data, out->nb_samples,
407		− (const uint8_t **)s->frame->extended_data, ret);
408		− if (ret < 0)
409		− goto fail;
410		−
411		− if (out->pts)
412		− out->pts -= s->delay;
413		− s->delay += in->nb_samples - ret;
414		− out->nb_samples = ret;
415		−
416		− av_frame_free(&s->frame);
417		− } else {
418		− td.in = in;
419		− td.out = out;
420		− td.nb_samples = nb_samples;
421		− td.channels = channels;
422		− ff_filter_execute(ctx, filter_channels, &td, NULL,
423		− FFMIN(channels, ff_filter_get_nb_threads(ctx)));
424		− }
	462	+ td.in = in;
	463	+ td.out = out;
	464	+ td.nb_samples = nb_samples;
	465	+ td.channels = channels;
	466	+ ff_filter_execute(ctx, filter_channels, &td, NULL,
	467	+ FFMIN(channels, ff_filter_get_nb_threads(ctx)));
425	468
426	469	if (out != in)
427	470	av_frame_free(&in);
428	471
	472	+ out->nb_samples /= s->oversample;
429	473	return ff_filter_frame(outlink, out);
430		−fail:
431		− if (out != in)
432		− av_frame_free(&out);
433		− av_frame_free(&in);
434		− av_frame_free(&s->frame);
435		−
436		− return ret;
437	474	}
438	475
439	476	static av_cold void uninit(AVFilterContext *ctx)
440	477	{
441	478	ASoftClipContext *s = ctx->priv;
442	479
443		− swr_free(&s->up_ctx);
444		− swr_free(&s->down_ctx);
	480	+ av_frame_free(&s->frame[0]);
	481	+ av_frame_free(&s->frame[1]);
445	482	}
446	483
447	484	static const AVFilterPad inputs[] = {

avfilter/af_asoftclip: rewrite oversampling

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