FFmpeg coverage

GCC Code Coverage Report

Directory:	../../../ffmpeg/		Exec	Total	Coverage
File:	src/libavfilter/af_resample.c	Lines:	0	167	0.0 %
Date:	2020-10-23 17:01:47	Branches:	0	110	0.0 %


/*
 * 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;

    int resampling;
    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, *out_formats, *in_samplerates, *out_samplerates;
    AVFilterChannelLayouts *in_layouts, *out_layouts;
    int ret;

    if (!(in_formats      = ff_all_formats         (AVMEDIA_TYPE_AUDIO)) ||
        !(out_formats     = ff_all_formats         (AVMEDIA_TYPE_AUDIO)) ||
        !(in_samplerates  = ff_all_samplerates     (                  )) ||
        !(out_samplerates = ff_all_samplerates     (                  )) ||
        !(in_layouts      = ff_all_channel_layouts (                  )) ||
        !(out_layouts     = ff_all_channel_layouts (                  )))
        return AVERROR(ENOMEM);

    if ((ret = ff_formats_ref         (in_formats,      &inlink->outcfg.formats        )) < 0 ||
        (ret = ff_formats_ref         (out_formats,     &outlink->incfg.formats        )) < 0 ||
        (ret = ff_formats_ref         (in_samplerates,  &inlink->outcfg.samplerates    )) < 0 ||
        (ret = ff_formats_ref         (out_samplerates, &outlink->incfg.samplerates    )) < 0 ||
        (ret = ff_channel_layouts_ref (in_layouts,      &inlink->outcfg.channel_layouts)) < 0 ||
        (ret = ff_channel_layouts_ref (out_layouts,     &outlink->incfg.channel_layouts)) < 0)
        return ret;

    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;

    int64_t resampling_forced;

    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;

    av_opt_get_int(s->avr, "force_resampling", 0, &resampling_forced);
    s->resampling = resampling_forced || (inlink->sample_rate != outlink->sample_rate);

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

    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->nb_samples = 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->resampling && 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;
            }

            if (s->resampling) {
                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;
            } else
                out->pts = in->pts;

            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;
}

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

static const AVClass *resample_child_class_iterate(void **iter)
{
    const AVClass *c = *iter ? NULL : avresample_get_class();
    *iter = (void*)(uintptr_t)c;
    return c;
}

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,
#if FF_API_CHILD_CLASS_NEXT
    .child_class_next = resample_child_class_next,
#endif
    .child_class_iterate = resample_child_class_iterate,
    .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,
};


Generated by: GCOVR (Version 4.2)

Line	Branch	Exec	Source
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			int resampling;
44			int64_t next_pts;
45			int64_t next_in_pts;
46
47			/* set by filter_frame() to signal an output frame to request_frame() */
48			int got_output;
49			} ResampleContext;
50
51			static av_cold int init(AVFilterContext ctx, AVDictionary *opts)
52			{
53			ResampleContext *s = ctx->priv;
54			const AVClass *avr_class = avresample_get_class();
55			AVDictionaryEntry *e = NULL;
56
57			while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
58			if (av_opt_find(&avr_class, e->key, NULL, 0,
59			AV_OPT_SEARCH_FAKE_OBJ \| AV_OPT_SEARCH_CHILDREN))
60			av_dict_set(&s->options, e->key, e->value, 0);
61			}
62
63			e = NULL;
64			while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
65			av_dict_set(opts, e->key, NULL, 0);
66
67			/* do not allow the user to override basic format options */
68			av_dict_set(&s->options, "in_channel_layout", NULL, 0);
69			av_dict_set(&s->options, "out_channel_layout", NULL, 0);
70			av_dict_set(&s->options, "in_sample_fmt", NULL, 0);
71			av_dict_set(&s->options, "out_sample_fmt", NULL, 0);
72			av_dict_set(&s->options, "in_sample_rate", NULL, 0);
73			av_dict_set(&s->options, "out_sample_rate", NULL, 0);
74
75			return 0;
76			}
77
78			static av_cold void uninit(AVFilterContext *ctx)
79			{
80			ResampleContext *s = ctx->priv;
81
82			if (s->avr) {
83			avresample_close(s->avr);
84			avresample_free(&s->avr);
85			}
86			av_dict_free(&s->options);
87			}
88
89			static int query_formats(AVFilterContext *ctx)
90			{
91			AVFilterLink *inlink = ctx->inputs[0];
92			AVFilterLink *outlink = ctx->outputs[0];
93			AVFilterFormats in_formats, out_formats, in_samplerates, out_samplerates;
94			AVFilterChannelLayouts in_layouts, out_layouts;
95			int ret;
96
97			if (!(in_formats = ff_all_formats (AVMEDIA_TYPE_AUDIO)) \|\|
98			!(out_formats = ff_all_formats (AVMEDIA_TYPE_AUDIO)) \|\|
99			!(in_samplerates = ff_all_samplerates ( )) \|\|
100			!(out_samplerates = ff_all_samplerates ( )) \|\|
101			!(in_layouts = ff_all_channel_layouts ( )) \|\|
102			!(out_layouts = ff_all_channel_layouts ( )))
103			return AVERROR(ENOMEM);
104
105			if ((ret = ff_formats_ref (in_formats, &inlink->outcfg.formats )) < 0 \|\|
106			(ret = ff_formats_ref (out_formats, &outlink->incfg.formats )) < 0 \|\|
107			(ret = ff_formats_ref (in_samplerates, &inlink->outcfg.samplerates )) < 0 \|\|
108			(ret = ff_formats_ref (out_samplerates, &outlink->incfg.samplerates )) < 0 \|\|
109			(ret = ff_channel_layouts_ref (in_layouts, &inlink->outcfg.channel_layouts)) < 0 \|\|
110			(ret = ff_channel_layouts_ref (out_layouts, &outlink->incfg.channel_layouts)) < 0)
111			return ret;
112
113			return 0;
114			}
115
116			static int config_output(AVFilterLink *outlink)
117			{
118			AVFilterContext *ctx = outlink->src;
119			AVFilterLink *inlink = ctx->inputs[0];
120			ResampleContext *s = ctx->priv;
121			char buf1[64], buf2[64];
122			int ret;
123
124			int64_t resampling_forced;
125
126			if (s->avr) {
127			avresample_close(s->avr);
128			avresample_free(&s->avr);
129			}
130
131			if (inlink->channel_layout == outlink->channel_layout &&
132			inlink->sample_rate == outlink->sample_rate &&
133			(inlink->format == outlink->format \|\|
134			(av_get_channel_layout_nb_channels(inlink->channel_layout) == 1 &&
135			av_get_channel_layout_nb_channels(outlink->channel_layout) == 1 &&
136			av_get_planar_sample_fmt(inlink->format) ==
137			av_get_planar_sample_fmt(outlink->format))))
138			return 0;
139
140			if (!(s->avr = avresample_alloc_context()))
141			return AVERROR(ENOMEM);
142
143			if (s->options) {
144			int ret;
145			AVDictionaryEntry *e = NULL;
146			while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
147			av_log(ctx, AV_LOG_VERBOSE, "lavr option: %s=%s\n", e->key, e->value);
148
149			ret = av_opt_set_dict(s->avr, &s->options);
150			if (ret < 0)
151			return ret;
152			}
153
154			av_opt_set_int(s->avr, "in_channel_layout", inlink ->channel_layout, 0);
155			av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
156			av_opt_set_int(s->avr, "in_sample_fmt", inlink ->format, 0);
157			av_opt_set_int(s->avr, "out_sample_fmt", outlink->format, 0);
158			av_opt_set_int(s->avr, "in_sample_rate", inlink ->sample_rate, 0);
159			av_opt_set_int(s->avr, "out_sample_rate", outlink->sample_rate, 0);
160
161			if ((ret = avresample_open(s->avr)) < 0)
162			return ret;
163
164			av_opt_get_int(s->avr, "force_resampling", 0, &resampling_forced);
165			s->resampling = resampling_forced \|\| (inlink->sample_rate != outlink->sample_rate);
166
167			if (s->resampling) {
168			outlink->time_base = (AVRational){ 1, outlink->sample_rate };
169			s->next_pts = AV_NOPTS_VALUE;
170			s->next_in_pts = AV_NOPTS_VALUE;
171			} else
172			outlink->time_base = inlink->time_base;
173
174			av_get_channel_layout_string(buf1, sizeof(buf1),
175			-1, inlink ->channel_layout);
176			av_get_channel_layout_string(buf2, sizeof(buf2),
177			-1, outlink->channel_layout);
178			av_log(ctx, AV_LOG_VERBOSE,
179			"fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
180			av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
181			av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
182
183			return 0;
184			}
185
186			static int request_frame(AVFilterLink *outlink)
187			{
188			AVFilterContext *ctx = outlink->src;
189			ResampleContext *s = ctx->priv;
190			int ret = 0;
191
192			s->got_output = 0;
193			while (ret >= 0 && !s->got_output)
194			ret = ff_request_frame(ctx->inputs[0]);
195
196			/* flush the lavr delay buffer */
197			if (ret == AVERROR_EOF && s->avr) {
198			AVFrame *frame;
199			int nb_samples = avresample_get_out_samples(s->avr, 0);
200
201			if (!nb_samples)
202			return ret;
203
204			frame = ff_get_audio_buffer(outlink, nb_samples);
205			if (!frame)
206			return AVERROR(ENOMEM);
207
208			ret = avresample_convert(s->avr, frame->extended_data,
209			frame->linesize[0], nb_samples,
210			NULL, 0, 0);
211			if (ret <= 0) {
212			av_frame_free(&frame);
213			return (ret == 0) ? AVERROR_EOF : ret;
214			}
215
216			frame->nb_samples = ret;
217			frame->pts = s->next_pts;
218			return ff_filter_frame(outlink, frame);
219			}
220			return ret;
221			}
222
223			static int filter_frame(AVFilterLink inlink, AVFrame in)
224			{
225			AVFilterContext *ctx = inlink->dst;
226			ResampleContext *s = ctx->priv;
227			AVFilterLink *outlink = ctx->outputs[0];
228			int ret;
229
230			if (s->avr) {
231			AVFrame *out;
232			int delay, nb_samples;
233
234			/* maximum possible samples lavr can output */
235			delay = avresample_get_delay(s->avr);
236			nb_samples = avresample_get_out_samples(s->avr, in->nb_samples);
237
238			out = ff_get_audio_buffer(outlink, nb_samples);
239			if (!out) {
240			ret = AVERROR(ENOMEM);
241			goto fail;
242			}
243
244			ret = avresample_convert(s->avr, out->extended_data, out->linesize[0],
245			nb_samples, in->extended_data, in->linesize[0],
246			in->nb_samples);
247			if (ret <= 0) {
248			av_frame_free(&out);
249			if (ret < 0)
250			goto fail;
251			}
252
253			av_assert0(!avresample_available(s->avr));
254
255			if (s->resampling && s->next_pts == AV_NOPTS_VALUE) {
256			if (in->pts == AV_NOPTS_VALUE) {
257			av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
258			"assuming 0.\n");
259			s->next_pts = 0;
260			} else
261			s->next_pts = av_rescale_q(in->pts, inlink->time_base,
262			outlink->time_base);
263			}
264
265			if (ret > 0) {
266			out->nb_samples = ret;
267
268			ret = av_frame_copy_props(out, in);
269			if (ret < 0) {
270			av_frame_free(&out);
271			goto fail;
272			}
273
274			if (s->resampling) {
275			out->sample_rate = outlink->sample_rate;
276			/* Only convert in->pts if there is a discontinuous jump.
277			This ensures that out->pts tracks the number of samples actually
278			output by the resampler in the absence of such a jump.
279			Otherwise, the rounding in av_rescale_q() and av_rescale()
280			causes off-by-1 errors. */
281			if (in->pts != AV_NOPTS_VALUE && in->pts != s->next_in_pts) {
282			out->pts = av_rescale_q(in->pts, inlink->time_base,
283			outlink->time_base) -
284			av_rescale(delay, outlink->sample_rate,
285			inlink->sample_rate);
286			} else
287			out->pts = s->next_pts;
288
289			s->next_pts = out->pts + out->nb_samples;
290			s->next_in_pts = in->pts + in->nb_samples;
291			} else
292			out->pts = in->pts;
293
294			ret = ff_filter_frame(outlink, out);
295			s->got_output = 1;
296			}
297
298			fail:
299			av_frame_free(&in);
300			} else {
301			in->format = outlink->format;
302			ret = ff_filter_frame(outlink, in);
303			s->got_output = 1;
304			}
305
306			return ret;
307			}
308
309			#if FF_API_CHILD_CLASS_NEXT
310			static const AVClass resample_child_class_next(const AVClass prev)
311			{
312			return prev ? NULL : avresample_get_class();
313			}
314			#endif
315
316			static const AVClass resample_child_class_iterate(void *iter)
317			{
318			const AVClass c = iter ? NULL : avresample_get_class();
319			iter = (void)(uintptr_t)c;
320			return c;
321			}
322
323			static void resample_child_next(void obj, void *prev)
324			{
325			ResampleContext *s = obj;
326			return prev ? NULL : s->avr;
327			}
328
329			static const AVClass resample_class = {
330			.class_name = "resample",
331			.item_name = av_default_item_name,
332			.version = LIBAVUTIL_VERSION_INT,
333			#if FF_API_CHILD_CLASS_NEXT
334			.child_class_next = resample_child_class_next,
335			#endif
336			.child_class_iterate = resample_child_class_iterate,
337			.child_next = resample_child_next,
338			};
339
340			static const AVFilterPad avfilter_af_resample_inputs[] = {
341			{
342			.name = "default",
343			.type = AVMEDIA_TYPE_AUDIO,
344			.filter_frame = filter_frame,
345			},
346			{ NULL }
347			};
348
349			static const AVFilterPad avfilter_af_resample_outputs[] = {
350			{
351			.name = "default",
352			.type = AVMEDIA_TYPE_AUDIO,
353			.config_props = config_output,
354			.request_frame = request_frame
355			},
356			{ NULL }
357			};
358
359			AVFilter ff_af_resample = {
360			.name = "resample",
361			.description = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
362			.priv_size = sizeof(ResampleContext),
363			.priv_class = &resample_class,
364			.init_dict = init,
365			.uninit = uninit,
366			.query_formats = query_formats,
367			.inputs = avfilter_af_resample_inputs,
368			.outputs = avfilter_af_resample_outputs,
369			};