[deb_ffmpeg.git] / ffmpeg / libavfilter / af_resample.c

/*
 * 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
 */

/**
 * @file
 * sample format and channel layout conversion audio filter
 */

#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/common.h"
#include "libavutil/dict.h"
#include "libavutil/mathematics.h"
#include "libavutil/opt.h"

#include "libavresample/avresample.h"

#include "audio.h"
#include "avfilter.h"
#include "formats.h"
#include "internal.h"

typedef struct ResampleContext {
    const AVClass *class;
    AVAudioResampleContext *avr;
    AVDictionary *options;

    int64_t next_pts;
    int64_t next_in_pts;

    /* set by filter_frame() to signal an output frame to request_frame() */
    int got_output;
} ResampleContext;

static av_cold int init(AVFilterContext *ctx, AVDictionary **opts)
{
    ResampleContext *s = ctx->priv;
    const AVClass *avr_class = avresample_get_class();
    AVDictionaryEntry *e = NULL;

    while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
        if (av_opt_find(&avr_class, e->key, NULL, 0,
                        AV_OPT_SEARCH_FAKE_OBJ | AV_OPT_SEARCH_CHILDREN))
            av_dict_set(&s->options, e->key, e->value, 0);
    }

    e = NULL;
    while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
        av_dict_set(opts, e->key, NULL, 0);

    /* do not allow the user to override basic format options */
    av_dict_set(&s->options,  "in_channel_layout", NULL, 0);
    av_dict_set(&s->options, "out_channel_layout", NULL, 0);
    av_dict_set(&s->options,  "in_sample_fmt",     NULL, 0);
    av_dict_set(&s->options, "out_sample_fmt",     NULL, 0);
    av_dict_set(&s->options,  "in_sample_rate",    NULL, 0);
    av_dict_set(&s->options, "out_sample_rate",    NULL, 0);

    return 0;
}

static av_cold void uninit(AVFilterContext *ctx)
{
    ResampleContext *s = ctx->priv;

    if (s->avr) {
        avresample_close(s->avr);
        avresample_free(&s->avr);
    }
    av_dict_free(&s->options);
}

static int query_formats(AVFilterContext *ctx)
{
    AVFilterLink *inlink  = ctx->inputs[0];
    AVFilterLink *outlink = ctx->outputs[0];

    AVFilterFormats        *in_formats      = ff_all_formats(AVMEDIA_TYPE_AUDIO);
    AVFilterFormats        *out_formats     = ff_all_formats(AVMEDIA_TYPE_AUDIO);
    AVFilterFormats        *in_samplerates  = ff_all_samplerates();
    AVFilterFormats        *out_samplerates = ff_all_samplerates();
    AVFilterChannelLayouts *in_layouts      = ff_all_channel_layouts();
    AVFilterChannelLayouts *out_layouts     = ff_all_channel_layouts();

    ff_formats_ref(in_formats,  &inlink->out_formats);
    ff_formats_ref(out_formats, &outlink->in_formats);

    ff_formats_ref(in_samplerates,  &inlink->out_samplerates);
    ff_formats_ref(out_samplerates, &outlink->in_samplerates);

    ff_channel_layouts_ref(in_layouts,  &inlink->out_channel_layouts);
    ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);

    return 0;
}

static int config_output(AVFilterLink *outlink)
{
    AVFilterContext *ctx = outlink->src;
    AVFilterLink *inlink = ctx->inputs[0];
    ResampleContext   *s = ctx->priv;
    char buf1[64], buf2[64];
    int ret;

    if (s->avr) {
        avresample_close(s->avr);
        avresample_free(&s->avr);
    }

    if (inlink->channel_layout == outlink->channel_layout &&
        inlink->sample_rate    == outlink->sample_rate    &&
        (inlink->format        == outlink->format ||
        (av_get_channel_layout_nb_channels(inlink->channel_layout)  == 1 &&
         av_get_channel_layout_nb_channels(outlink->channel_layout) == 1 &&
         av_get_planar_sample_fmt(inlink->format) ==
         av_get_planar_sample_fmt(outlink->format))))
        return 0;

    if (!(s->avr = avresample_alloc_context()))
        return AVERROR(ENOMEM);

    if (s->options) {
        int ret;
        AVDictionaryEntry *e = NULL;
        while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
            av_log(ctx, AV_LOG_VERBOSE, "lavr option: %s=%s\n", e->key, e->value);

        ret = av_opt_set_dict(s->avr, &s->options);
        if (ret < 0)
            return ret;
    }

    av_opt_set_int(s->avr,  "in_channel_layout", inlink ->channel_layout, 0);
    av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
    av_opt_set_int(s->avr,  "in_sample_fmt",     inlink ->format,         0);
    av_opt_set_int(s->avr, "out_sample_fmt",     outlink->format,         0);
    av_opt_set_int(s->avr,  "in_sample_rate",    inlink ->sample_rate,    0);
    av_opt_set_int(s->avr, "out_sample_rate",    outlink->sample_rate,    0);

    if ((ret = avresample_open(s->avr)) < 0)
        return ret;

    outlink->time_base = (AVRational){ 1, outlink->sample_rate };
    s->next_pts        = AV_NOPTS_VALUE;
    s->next_in_pts     = AV_NOPTS_VALUE;

    av_get_channel_layout_string(buf1, sizeof(buf1),
                                 -1, inlink ->channel_layout);
    av_get_channel_layout_string(buf2, sizeof(buf2),
                                 -1, outlink->channel_layout);
    av_log(ctx, AV_LOG_VERBOSE,
           "fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
           av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
           av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);

    return 0;
}

static int request_frame(AVFilterLink *outlink)
{
    AVFilterContext *ctx = outlink->src;
    ResampleContext   *s = ctx->priv;
    int ret = 0;

    s->got_output = 0;
    while (ret >= 0 && !s->got_output)
        ret = ff_request_frame(ctx->inputs[0]);

    /* flush the lavr delay buffer */
    if (ret == AVERROR_EOF && s->avr) {
        AVFrame *frame;
        int nb_samples = avresample_get_out_samples(s->avr, 0);

        if (!nb_samples)
            return ret;

        frame = ff_get_audio_buffer(outlink, nb_samples);
        if (!frame)
            return AVERROR(ENOMEM);

        ret = avresample_convert(s->avr, frame->extended_data,
                                 frame->linesize[0], nb_samples,
                                 NULL, 0, 0);
        if (ret <= 0) {
            av_frame_free(&frame);
            return (ret == 0) ? AVERROR_EOF : ret;
        }

        frame->pts = s->next_pts;
        return ff_filter_frame(outlink, frame);
    }
    return ret;
}

static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
    AVFilterContext  *ctx = inlink->dst;
    ResampleContext    *s = ctx->priv;
    AVFilterLink *outlink = ctx->outputs[0];
    int ret;

    if (s->avr) {
        AVFrame *out;
        int delay, nb_samples;

        /* maximum possible samples lavr can output */
        delay      = avresample_get_delay(s->avr);
        nb_samples = avresample_get_out_samples(s->avr, in->nb_samples);

        out = ff_get_audio_buffer(outlink, nb_samples);
        if (!out) {
            ret = AVERROR(ENOMEM);
            goto fail;
        }

        ret = avresample_convert(s->avr, out->extended_data, out->linesize[0],
                                 nb_samples, in->extended_data, in->linesize[0],
                                 in->nb_samples);
        if (ret <= 0) {
            av_frame_free(&out);
            if (ret < 0)
                goto fail;
        }

        av_assert0(!avresample_available(s->avr));

        if (s->next_pts == AV_NOPTS_VALUE) {
            if (in->pts == AV_NOPTS_VALUE) {
                av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
                       "assuming 0.\n");
                s->next_pts = 0;
            } else
                s->next_pts = av_rescale_q(in->pts, inlink->time_base,
                                           outlink->time_base);
        }

        if (ret > 0) {
            out->nb_samples = ret;

            ret = av_frame_copy_props(out, in);
            if (ret < 0) {
                av_frame_free(&out);
                goto fail;
            }

            out->sample_rate = outlink->sample_rate;
            /* Only convert in->pts if there is a discontinuous jump.
               This ensures that out->pts tracks the number of samples actually
               output by the resampler in the absence of such a jump.
               Otherwise, the rounding in av_rescale_q() and av_rescale()
               causes off-by-1 errors. */
            if (in->pts != AV_NOPTS_VALUE && in->pts != s->next_in_pts) {
                out->pts = av_rescale_q(in->pts, inlink->time_base,
                                            outlink->time_base) -
                               av_rescale(delay, outlink->sample_rate,
                                          inlink->sample_rate);
            } else
                out->pts = s->next_pts;

            s->next_pts = out->pts + out->nb_samples;
            s->next_in_pts = in->pts + in->nb_samples;

            ret = ff_filter_frame(outlink, out);
            s->got_output = 1;
        }

fail:
        av_frame_free(&in);
    } else {
        in->format = outlink->format;
        ret = ff_filter_frame(outlink, in);
        s->got_output = 1;
    }

    return ret;
}

static const AVClass *resample_child_class_next(const AVClass *prev)
{
    return prev ? NULL : avresample_get_class();
}

static void *resample_child_next(void *obj, void *prev)
{
    ResampleContext *s = obj;
    return prev ? NULL : s->avr;
}

static const AVClass resample_class = {
    .class_name       = "resample",
    .item_name        = av_default_item_name,
    .version          = LIBAVUTIL_VERSION_INT,
    .child_class_next = resample_child_class_next,
    .child_next       = resample_child_next,
};

static const AVFilterPad avfilter_af_resample_inputs[] = {
    {
        .name          = "default",
        .type          = AVMEDIA_TYPE_AUDIO,
        .filter_frame  = filter_frame,
    },
    { NULL }
};

static const AVFilterPad avfilter_af_resample_outputs[] = {
    {
        .name          = "default",
        .type          = AVMEDIA_TYPE_AUDIO,
        .config_props  = config_output,
        .request_frame = request_frame
    },
    { NULL }
};

AVFilter ff_af_resample = {
    .name          = "resample",
    .description   = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
    .priv_size     = sizeof(ResampleContext),
    .priv_class    = &resample_class,
    .init_dict     = init,
    .uninit        = uninit,
    .query_formats = query_formats,
    .inputs        = avfilter_af_resample_inputs,
    .outputs       = avfilter_af_resample_outputs,
};
Commit	Line	Data
2ba45a60 DM	1	/*
	2	* This file is part of FFmpeg.
	3	*
	4	* FFmpeg is free software; you can redistribute it and/or
	5	* modify it under the terms of the GNU Lesser General Public
	6	* License as published by the Free Software Foundation; either
	7	* version 2.1 of the License, or (at your option) any later version.
	8	*
	9	* FFmpeg is distributed in the hope that it will be useful,
	10	* but WITHOUT ANY WARRANTY; without even the implied warranty of
	11	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	12	* Lesser General Public License for more details.
	13	*
	14	* You should have received a copy of the GNU Lesser General Public
	15	* License along with FFmpeg; if not, write to the Free Software
	16	* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
	17	*/
	18
	19	/**
	20	* @file
	21	* sample format and channel layout conversion audio filter
	22	*/
	23
	24	#include "libavutil/avassert.h"
	25	#include "libavutil/avstring.h"
	26	#include "libavutil/common.h"
	27	#include "libavutil/dict.h"
	28	#include "libavutil/mathematics.h"
	29	#include "libavutil/opt.h"
	30
	31	#include "libavresample/avresample.h"
	32
	33	#include "audio.h"
	34	#include "avfilter.h"
	35	#include "formats.h"
	36	#include "internal.h"
	37
	38	typedef struct ResampleContext {
	39	const AVClass *class;
	40	AVAudioResampleContext *avr;
	41	AVDictionary *options;
	42
	43	int64_t next_pts;
	44	int64_t next_in_pts;
	45
	46	/* set by filter_frame() to signal an output frame to request_frame() */
	47	int got_output;
	48	} ResampleContext;
	49
	50	static av_cold int init(AVFilterContext ctx, AVDictionary *opts)
	51	{
	52	ResampleContext *s = ctx->priv;
	53	const AVClass *avr_class = avresample_get_class();
	54	AVDictionaryEntry *e = NULL;
	55
	56	while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
	57	if (av_opt_find(&avr_class, e->key, NULL, 0,
	58	AV_OPT_SEARCH_FAKE_OBJ \| AV_OPT_SEARCH_CHILDREN))
	59	av_dict_set(&s->options, e->key, e->value, 0);
	60	}
	61
	62	e = NULL;
	63	while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
	64	av_dict_set(opts, e->key, NULL, 0);
65
66	/* do not allow the user to override basic format options */
67	av_dict_set(&s->options, "in_channel_layout", NULL, 0);
68	av_dict_set(&s->options, "out_channel_layout", NULL, 0);
69	av_dict_set(&s->options, "in_sample_fmt", NULL, 0);
70	av_dict_set(&s->options, "out_sample_fmt", NULL, 0);
71	av_dict_set(&s->options, "in_sample_rate", NULL, 0);
72	av_dict_set(&s->options, "out_sample_rate", NULL, 0);
73
74	return 0;
75	}
76
77	static av_cold void uninit(AVFilterContext *ctx)
78	{
79	ResampleContext *s = ctx->priv;
80
81	if (s->avr) {
82	avresample_close(s->avr);
83	avresample_free(&s->avr);
84	}
85	av_dict_free(&s->options);
86	}
87
88	static int query_formats(AVFilterContext *ctx)
89	{
90	AVFilterLink *inlink = ctx->inputs[0];
91	AVFilterLink *outlink = ctx->outputs[0];
92
93	AVFilterFormats *in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
94	AVFilterFormats *out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
95	AVFilterFormats *in_samplerates = ff_all_samplerates();
96	AVFilterFormats *out_samplerates = ff_all_samplerates();
97	AVFilterChannelLayouts *in_layouts = ff_all_channel_layouts();
98	AVFilterChannelLayouts *out_layouts = ff_all_channel_layouts();
99
100	ff_formats_ref(in_formats, &inlink->out_formats);
101	ff_formats_ref(out_formats, &outlink->in_formats);
102
103	ff_formats_ref(in_samplerates, &inlink->out_samplerates);
104	ff_formats_ref(out_samplerates, &outlink->in_samplerates);
105
106	ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
107	ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
108
109	return 0;
110	}
111
112	static int config_output(AVFilterLink *outlink)
113	{
114	AVFilterContext *ctx = outlink->src;
115	AVFilterLink *inlink = ctx->inputs[0];
116	ResampleContext *s = ctx->priv;
117	char buf1[64], buf2[64];
118	int ret;
119
120	if (s->avr) {
121	avresample_close(s->avr);
122	avresample_free(&s->avr);
123	}
124
125	if (inlink->channel_layout == outlink->channel_layout &&
126	inlink->sample_rate == outlink->sample_rate &&
127	(inlink->format == outlink->format \|\|
128	(av_get_channel_layout_nb_channels(inlink->channel_layout) == 1 &&
129	av_get_channel_layout_nb_channels(outlink->channel_layout) == 1 &&
130	av_get_planar_sample_fmt(inlink->format) ==
131	av_get_planar_sample_fmt(outlink->format))))
132	return 0;
133
134	if (!(s->avr = avresample_alloc_context()))
135	return AVERROR(ENOMEM);
136
137	if (s->options) {
f6fa7814	138	int ret;
2ba45a60 DM	139	AVDictionaryEntry *e = NULL;
	140	while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
	141	av_log(ctx, AV_LOG_VERBOSE, "lavr option: %s=%s\n", e->key, e->value);
	142
f6fa7814 DM	143	ret = av_opt_set_dict(s->avr, &s->options);
	144	if (ret < 0)
	145	return ret;
2ba45a60 DM	146	}
	147
	148	av_opt_set_int(s->avr, "in_channel_layout", inlink ->channel_layout, 0);
	149	av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
	150	av_opt_set_int(s->avr, "in_sample_fmt", inlink ->format, 0);
	151	av_opt_set_int(s->avr, "out_sample_fmt", outlink->format, 0);
	152	av_opt_set_int(s->avr, "in_sample_rate", inlink ->sample_rate, 0);
	153	av_opt_set_int(s->avr, "out_sample_rate", outlink->sample_rate, 0);
	154
	155	if ((ret = avresample_open(s->avr)) < 0)
	156	return ret;
	157
	158	outlink->time_base = (AVRational){ 1, outlink->sample_rate };
	159	s->next_pts = AV_NOPTS_VALUE;
	160	s->next_in_pts = AV_NOPTS_VALUE;
	161
	162	av_get_channel_layout_string(buf1, sizeof(buf1),
	163	-1, inlink ->channel_layout);
	164	av_get_channel_layout_string(buf2, sizeof(buf2),
	165	-1, outlink->channel_layout);
	166	av_log(ctx, AV_LOG_VERBOSE,
	167	"fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
	168	av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
	169	av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
	170
	171	return 0;
	172	}
	173
	174	static int request_frame(AVFilterLink *outlink)
	175	{
	176	AVFilterContext *ctx = outlink->src;
	177	ResampleContext *s = ctx->priv;
	178	int ret = 0;
	179
	180	s->got_output = 0;
	181	while (ret >= 0 && !s->got_output)
	182	ret = ff_request_frame(ctx->inputs[0]);
	183
	184	/* flush the lavr delay buffer */
	185	if (ret == AVERROR_EOF && s->avr) {
	186	AVFrame *frame;
	187	int nb_samples = avresample_get_out_samples(s->avr, 0);
	188
	189	if (!nb_samples)
	190	return ret;
	191
	192	frame = ff_get_audio_buffer(outlink, nb_samples);
	193	if (!frame)
	194	return AVERROR(ENOMEM);
	195
	196	ret = avresample_convert(s->avr, frame->extended_data,
	197	frame->linesize[0], nb_samples,
	198	NULL, 0, 0);
	199	if (ret <= 0) {
	200	av_frame_free(&frame);
	201	return (ret == 0) ? AVERROR_EOF : ret;
	202	}
	203
	204	frame->pts = s->next_pts;
	205	return ff_filter_frame(outlink, frame);
	206	}
	207	return ret;
	208	}
	209
210	static int filter_frame(AVFilterLink inlink, AVFrame in)
211	{
212	AVFilterContext *ctx = inlink->dst;
213	ResampleContext *s = ctx->priv;
214	AVFilterLink *outlink = ctx->outputs[0];
215	int ret;
216
217	if (s->avr) {
218	AVFrame *out;
219	int delay, nb_samples;
220
221	/* maximum possible samples lavr can output */
222	delay = avresample_get_delay(s->avr);
223	nb_samples = avresample_get_out_samples(s->avr, in->nb_samples);
224
225	out = ff_get_audio_buffer(outlink, nb_samples);
226	if (!out) {
227	ret = AVERROR(ENOMEM);
228	goto fail;
229	}
230
231	ret = avresample_convert(s->avr, out->extended_data, out->linesize[0],
232	nb_samples, in->extended_data, in->linesize[0],
233	in->nb_samples);
234	if (ret <= 0) {
235	av_frame_free(&out);
236	if (ret < 0)
237	goto fail;
238	}
239
240	av_assert0(!avresample_available(s->avr));
241
242	if (s->next_pts == AV_NOPTS_VALUE) {
243	if (in->pts == AV_NOPTS_VALUE) {
244	av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
245	"assuming 0.\n");
246	s->next_pts = 0;
247	} else
248	s->next_pts = av_rescale_q(in->pts, inlink->time_base,
249	outlink->time_base);
250	}
251
252	if (ret > 0) {
253	out->nb_samples = ret;
254
255	ret = av_frame_copy_props(out, in);
256	if (ret < 0) {
257	av_frame_free(&out);
258	goto fail;
259	}
260
261	out->sample_rate = outlink->sample_rate;
262	/* Only convert in->pts if there is a discontinuous jump.
263	This ensures that out->pts tracks the number of samples actually
264	output by the resampler in the absence of such a jump.
265	Otherwise, the rounding in av_rescale_q() and av_rescale()
266	causes off-by-1 errors. */
267	if (in->pts != AV_NOPTS_VALUE && in->pts != s->next_in_pts) {
268	out->pts = av_rescale_q(in->pts, inlink->time_base,
269	outlink->time_base) -
270	av_rescale(delay, outlink->sample_rate,
271	inlink->sample_rate);
272	} else
273	out->pts = s->next_pts;
274
275	s->next_pts = out->pts + out->nb_samples;
276	s->next_in_pts = in->pts + in->nb_samples;
277
278	ret = ff_filter_frame(outlink, out);
279	s->got_output = 1;
280	}
281
282	fail:
283	av_frame_free(&in);
284	} else {
285	in->format = outlink->format;
286	ret = ff_filter_frame(outlink, in);
287	s->got_output = 1;
288	}
289
290	return ret;
291	}
292
293	static const AVClass resample_child_class_next(const AVClass prev)
294	{
295	return prev ? NULL : avresample_get_class();
296	}
297
298	static void resample_child_next(void obj, void *prev)
299	{
300	ResampleContext *s = obj;
301	return prev ? NULL : s->avr;
302	}
303
304	static const AVClass resample_class = {
305	.class_name = "resample",
306	.item_name = av_default_item_name,
307	.version = LIBAVUTIL_VERSION_INT,
308	.child_class_next = resample_child_class_next,
309	.child_next = resample_child_next,
310	};
311
312	static const AVFilterPad avfilter_af_resample_inputs[] = {
313	{
314	.name = "default",
315	.type = AVMEDIA_TYPE_AUDIO,
316	.filter_frame = filter_frame,
317	},
318	{ NULL }
319	};
320
321	static const AVFilterPad avfilter_af_resample_outputs[] = {
322	{
323	.name = "default",
324	.type = AVMEDIA_TYPE_AUDIO,
325	.config_props = config_output,
326	.request_frame = request_frame
327	},
328	{ NULL }
329	};
330
331	AVFilter ff_af_resample = {
332	.name = "resample",
333	.description = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
334	.priv_size = sizeof(ResampleContext),
335	.priv_class = &resample_class,
336	.init_dict = init,
337	.uninit = uninit,
338	.query_formats = query_formats,
339	.inputs = avfilter_af_resample_inputs,
340	.outputs = avfilter_af_resample_outputs,
341	};