FFmpeg coverage

GCC Code Coverage Report

Directory:	../../../ffmpeg/		Exec	Total	Coverage
File:	src/libavfilter/af_aresample.c	Lines:	137	160	85.6 %
Date:	2020-08-14 10:39:37	Branches:	49	76	64.5 %


/*
 * Copyright (c) 2011 Stefano Sabatini
 * Copyright (c) 2011 Mina Nagy Zaki
 *
 * 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
 * resampling audio filter
 */

#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "libavutil/avassert.h"
#include "libswresample/swresample.h"
#include "avfilter.h"
#include "audio.h"
#include "internal.h"

typedef struct AResampleContext {
    const AVClass *class;
    int sample_rate_arg;
    double ratio;
    struct SwrContext *swr;
    int64_t next_pts;
    int more_data;
} AResampleContext;

static av_cold int init_dict(AVFilterContext *ctx, AVDictionary **opts)
{
    AResampleContext *aresample = ctx->priv;
    int ret = 0;

    aresample->next_pts = AV_NOPTS_VALUE;
    aresample->swr = swr_alloc();
    if (!aresample->swr) {
        ret = AVERROR(ENOMEM);
        goto end;
    }

    if (opts) {
        AVDictionaryEntry *e = NULL;

        while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
            if ((ret = av_opt_set(aresample->swr, e->key, e->value, 0)) < 0)
                goto end;
        }
        av_dict_free(opts);
    }
    if (aresample->sample_rate_arg > 0)
        av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0);
end:
    return ret;
}

static av_cold void uninit(AVFilterContext *ctx)
{
    AResampleContext *aresample = ctx->priv;
    swr_free(&aresample->swr);
}

static int query_formats(AVFilterContext *ctx)
{
    AResampleContext *aresample = ctx->priv;
    enum AVSampleFormat out_format;
    int64_t out_rate, out_layout;

    AVFilterLink *inlink  = ctx->inputs[0];
    AVFilterLink *outlink = ctx->outputs[0];

    AVFilterFormats        *in_formats, *out_formats;
    AVFilterFormats        *in_samplerates, *out_samplerates;
    AVFilterChannelLayouts *in_layouts, *out_layouts;
    int ret;

    av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
    av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
    av_opt_get_int(aresample->swr, "ocl", 0, &out_layout);

    in_formats      = ff_all_formats(AVMEDIA_TYPE_AUDIO);
    if ((ret = ff_formats_ref(in_formats, &inlink->out_formats)) < 0)
        return ret;

    in_samplerates  = ff_all_samplerates();
    if ((ret = ff_formats_ref(in_samplerates, &inlink->out_samplerates)) < 0)
        return ret;

    in_layouts      = ff_all_channel_counts();
    if ((ret = ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts)) < 0)
        return ret;

    if(out_rate > 0) {
        int ratelist[] = { out_rate, -1 };
        out_samplerates = ff_make_format_list(ratelist);
    } else {
        out_samplerates = ff_all_samplerates();
    }

    if ((ret = ff_formats_ref(out_samplerates, &outlink->in_samplerates)) < 0)
        return ret;

    if(out_format != AV_SAMPLE_FMT_NONE) {
        int formatlist[] = { out_format, -1 };
        out_formats = ff_make_format_list(formatlist);
    } else
        out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
    if ((ret = ff_formats_ref(out_formats, &outlink->in_formats)) < 0)
        return ret;

    if(out_layout) {
        int64_t layout_list[] = { out_layout, -1 };
        out_layouts = ff_make_format64_list(layout_list);
    } else
        out_layouts = ff_all_channel_counts();

    return ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
}


static int config_output(AVFilterLink *outlink)
{
    int ret;
    AVFilterContext *ctx = outlink->src;
    AVFilterLink *inlink = ctx->inputs[0];
    AResampleContext *aresample = ctx->priv;
    int64_t out_rate, out_layout;
    enum AVSampleFormat out_format;
    char inchl_buf[128], outchl_buf[128];

    aresample->swr = swr_alloc_set_opts(aresample->swr,
                                        outlink->channel_layout, outlink->format, outlink->sample_rate,
                                        inlink->channel_layout, inlink->format, inlink->sample_rate,
                                        0, ctx);
    if (!aresample->swr)
        return AVERROR(ENOMEM);
    if (!inlink->channel_layout)
        av_opt_set_int(aresample->swr, "ich", inlink->channels, 0);
    if (!outlink->channel_layout)
        av_opt_set_int(aresample->swr, "och", outlink->channels, 0);

    ret = swr_init(aresample->swr);
    if (ret < 0)
        return ret;

    av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
    av_opt_get_int(aresample->swr, "ocl", 0, &out_layout);
    av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
    outlink->time_base = (AVRational) {1, out_rate};

    av_assert0(outlink->sample_rate == out_rate);
    av_assert0(outlink->channel_layout == out_layout || !outlink->channel_layout);
    av_assert0(outlink->format == out_format);

    aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;

    av_get_channel_layout_string(inchl_buf,  sizeof(inchl_buf),  inlink ->channels, inlink ->channel_layout);
    av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), outlink->channels, outlink->channel_layout);

    av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",
           inlink ->channels, inchl_buf,  av_get_sample_fmt_name(inlink->format),  inlink->sample_rate,
           outlink->channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
    return 0;
}

static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref)
{
    AResampleContext *aresample = inlink->dst->priv;
    const int n_in  = insamplesref->nb_samples;
    int64_t delay;
    int n_out       = n_in * aresample->ratio + 32;
    AVFilterLink *const outlink = inlink->dst->outputs[0];
    AVFrame *outsamplesref;
    int ret;

    delay = swr_get_delay(aresample->swr, outlink->sample_rate);
    if (delay > 0)
        n_out += FFMIN(delay, FFMAX(4096, n_out));

    outsamplesref = ff_get_audio_buffer(outlink, n_out);

    if(!outsamplesref) {
        av_frame_free(&insamplesref);
        return AVERROR(ENOMEM);
    }

    av_frame_copy_props(outsamplesref, insamplesref);
    outsamplesref->format                = outlink->format;
    outsamplesref->channels              = outlink->channels;
    outsamplesref->channel_layout        = outlink->channel_layout;
    outsamplesref->sample_rate           = outlink->sample_rate;

    if(insamplesref->pts != AV_NOPTS_VALUE) {
        int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
        int64_t outpts= swr_next_pts(aresample->swr, inpts);
        aresample->next_pts =
        outsamplesref->pts  = ROUNDED_DIV(outpts, inlink->sample_rate);
    } else {
        outsamplesref->pts  = AV_NOPTS_VALUE;
    }
    n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
                                 (void *)insamplesref->extended_data, n_in);
    if (n_out <= 0) {
        av_frame_free(&outsamplesref);
        av_frame_free(&insamplesref);
        return 0;
    }

    aresample->more_data = outsamplesref->nb_samples == n_out; // Indicate that there is probably more data in our buffers

    outsamplesref->nb_samples  = n_out;

    ret = ff_filter_frame(outlink, outsamplesref);
    av_frame_free(&insamplesref);
    return ret;
}

static int flush_frame(AVFilterLink *outlink, int final, AVFrame **outsamplesref_ret)
{
    AVFilterContext *ctx = outlink->src;
    AResampleContext *aresample = ctx->priv;
    AVFilterLink *const inlink = outlink->src->inputs[0];
    AVFrame *outsamplesref;
    int n_out = 4096;
    int64_t pts;

    outsamplesref = ff_get_audio_buffer(outlink, n_out);
    *outsamplesref_ret = outsamplesref;
    if (!outsamplesref)
        return AVERROR(ENOMEM);

    pts = swr_next_pts(aresample->swr, INT64_MIN);
    pts = ROUNDED_DIV(pts, inlink->sample_rate);

    n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, final ? NULL : (void*)outsamplesref->extended_data, 0);
    if (n_out <= 0) {
        av_frame_free(&outsamplesref);
        return (n_out == 0) ? AVERROR_EOF : n_out;
    }

    outsamplesref->sample_rate = outlink->sample_rate;
    outsamplesref->nb_samples  = n_out;

    outsamplesref->pts = pts;

    return 0;
}

static int request_frame(AVFilterLink *outlink)
{
    AVFilterContext *ctx = outlink->src;
    AResampleContext *aresample = ctx->priv;
    int ret;

    // First try to get data from the internal buffers
    if (aresample->more_data) {
        AVFrame *outsamplesref;

        if (flush_frame(outlink, 0, &outsamplesref) >= 0) {
            return ff_filter_frame(outlink, outsamplesref);
        }
    }
    aresample->more_data = 0;

    // Second request more data from the input
    ret = ff_request_frame(ctx->inputs[0]);

    // Third if we hit the end flush
    if (ret == AVERROR_EOF) {
        AVFrame *outsamplesref;

        if ((ret = flush_frame(outlink, 1, &outsamplesref)) < 0)
            return ret;

        return ff_filter_frame(outlink, outsamplesref);
    }
    return ret;
}

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

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

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

#define OFFSET(x) offsetof(AResampleContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

static const AVOption options[] = {
    {"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0},  0,        INT_MAX, FLAGS },
    {NULL}
};

static const AVClass aresample_class = {
    .class_name       = "aresample",
    .item_name        = av_default_item_name,
    .option           = options,
    .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 aresample_inputs[] = {
    {
        .name         = "default",
        .type         = AVMEDIA_TYPE_AUDIO,
        .filter_frame = filter_frame,
    },
    { NULL }
};

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

AVFilter ff_af_aresample = {
    .name          = "aresample",
    .description   = NULL_IF_CONFIG_SMALL("Resample audio data."),
    .init_dict     = init_dict,
    .uninit        = uninit,
    .query_formats = query_formats,
    .priv_size     = sizeof(AResampleContext),
    .priv_class    = &aresample_class,
    .inputs        = aresample_inputs,
    .outputs       = aresample_outputs,
};


Generated by: GCOVR (Version 4.2)

Line	Branch	Exec	Source
1			/*
2			* Copyright (c) 2011 Stefano Sabatini
3			* Copyright (c) 2011 Mina Nagy Zaki
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			* resampling audio filter
25			*/
26
27			#include "libavutil/avstring.h"
28			#include "libavutil/channel_layout.h"
29			#include "libavutil/opt.h"
30			#include "libavutil/samplefmt.h"
31			#include "libavutil/avassert.h"
32			#include "libswresample/swresample.h"
33			#include "avfilter.h"
34			#include "audio.h"
35			#include "internal.h"
36
37			typedef struct AResampleContext {
38			const AVClass *class;
39			int sample_rate_arg;
40			double ratio;
41			struct SwrContext *swr;
42			int64_t next_pts;
43			int more_data;
44			} AResampleContext;
45
46		1201	static av_cold int init_dict(AVFilterContext ctx, AVDictionary *opts)
47			{
48		1201	AResampleContext *aresample = ctx->priv;
49		1201	int ret = 0;
50
51		1201	aresample->next_pts = AV_NOPTS_VALUE;
52		1201	aresample->swr = swr_alloc();
53	✗✓	1201	if (!aresample->swr) {
54			ret = AVERROR(ENOMEM);
55			goto end;
56			}
57
58	✓✗	1201	if (opts) {
59		1201	AVDictionaryEntry *e = NULL;
60
61	✓✓	3284	while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
62	✗✓	2083	if ((ret = av_opt_set(aresample->swr, e->key, e->value, 0)) < 0)
63			goto end;
64			}
65		1201	av_dict_free(opts);
66			}
67	✓✓	1201	if (aresample->sample_rate_arg > 0)
68		628	av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0);
69		573	end:
70		1201	return ret;
71			}
72
73		1201	static av_cold void uninit(AVFilterContext *ctx)
74			{
75		1201	AResampleContext *aresample = ctx->priv;
76		1201	swr_free(&aresample->swr);
77		1201	}
78
79		1201	static int query_formats(AVFilterContext *ctx)
80			{
81		1201	AResampleContext *aresample = ctx->priv;
82			enum AVSampleFormat out_format;
83			int64_t out_rate, out_layout;
84
85		1201	AVFilterLink *inlink = ctx->inputs[0];
86		1201	AVFilterLink *outlink = ctx->outputs[0];
87
88			AVFilterFormats in_formats, out_formats;
89			AVFilterFormats in_samplerates, out_samplerates;
90			AVFilterChannelLayouts in_layouts, out_layouts;
91			int ret;
92
93		1201	av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
94		1201	av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
95		1201	av_opt_get_int(aresample->swr, "ocl", 0, &out_layout);
96
97		1201	in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
98	✗✓	1201	if ((ret = ff_formats_ref(in_formats, &inlink->out_formats)) < 0)
99			return ret;
100
101		1201	in_samplerates = ff_all_samplerates();
102	✗✓	1201	if ((ret = ff_formats_ref(in_samplerates, &inlink->out_samplerates)) < 0)
103			return ret;
104
105		1201	in_layouts = ff_all_channel_counts();
106	✗✓	1201	if ((ret = ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts)) < 0)
107			return ret;
108
109	✓✓	1201	if(out_rate > 0) {
110		628	int ratelist[] = { out_rate, -1 };
111		628	out_samplerates = ff_make_format_list(ratelist);
112			} else {
113		573	out_samplerates = ff_all_samplerates();
114			}
115
116	✗✓	1201	if ((ret = ff_formats_ref(out_samplerates, &outlink->in_samplerates)) < 0)
117			return ret;
118
119	✗✓	1201	if(out_format != AV_SAMPLE_FMT_NONE) {
120			int formatlist[] = { out_format, -1 };
121			out_formats = ff_make_format_list(formatlist);
122			} else
123		1201	out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
124	✗✓	1201	if ((ret = ff_formats_ref(out_formats, &outlink->in_formats)) < 0)
125			return ret;
126
127	✗✓	1201	if(out_layout) {
128			int64_t layout_list[] = { out_layout, -1 };
129			out_layouts = ff_make_format64_list(layout_list);
130			} else
131		1201	out_layouts = ff_all_channel_counts();
132
133		1201	return ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
134			}
135
136
137		1201	static int config_output(AVFilterLink *outlink)
138			{
139			int ret;
140		1201	AVFilterContext *ctx = outlink->src;
141		1201	AVFilterLink *inlink = ctx->inputs[0];
142		1201	AResampleContext *aresample = ctx->priv;
143			int64_t out_rate, out_layout;
144			enum AVSampleFormat out_format;
145			char inchl_buf[128], outchl_buf[128];
146
147		2402	aresample->swr = swr_alloc_set_opts(aresample->swr,
148		1201	outlink->channel_layout, outlink->format, outlink->sample_rate,
149		1201	inlink->channel_layout, inlink->format, inlink->sample_rate,
150			0, ctx);
151	✗✓	1201	if (!aresample->swr)
152			return AVERROR(ENOMEM);
153	✓✓	1201	if (!inlink->channel_layout)
154		3	av_opt_set_int(aresample->swr, "ich", inlink->channels, 0);
155	✓✓	1201	if (!outlink->channel_layout)
156		2	av_opt_set_int(aresample->swr, "och", outlink->channels, 0);
157
158		1201	ret = swr_init(aresample->swr);
159	✗✓	1201	if (ret < 0)
160			return ret;
161
162		1201	av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
163		1201	av_opt_get_int(aresample->swr, "ocl", 0, &out_layout);
164		1201	av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
165		1201	outlink->time_base = (AVRational) {1, out_rate};
166
167	✗✓	1201	av_assert0(outlink->sample_rate == out_rate);
168	✗✓✗✗	1201	av_assert0(outlink->channel_layout == out_layout \|\| !outlink->channel_layout);
169	✗✓	1201	av_assert0(outlink->format == out_format);
170
171		1201	aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
172
173		1201	av_get_channel_layout_string(inchl_buf, sizeof(inchl_buf), inlink ->channels, inlink ->channel_layout);
174		1201	av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), outlink->channels, outlink->channel_layout);
175
176		1201	av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",
177		1201	inlink ->channels, inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,
178		1201	outlink->channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
179		1201	return 0;
180			}
181
182		279722	static int filter_frame(AVFilterLink inlink, AVFrame insamplesref)
183			{
184		279722	AResampleContext *aresample = inlink->dst->priv;
185		279722	const int n_in = insamplesref->nb_samples;
186			int64_t delay;
187		279722	int n_out = n_in * aresample->ratio + 32;
188		279722	AVFilterLink *const outlink = inlink->dst->outputs[0];
189			AVFrame *outsamplesref;
190			int ret;
191
192		279722	delay = swr_get_delay(aresample->swr, outlink->sample_rate);
193	✓✓	279722	if (delay > 0)
194		4253	n_out += FFMIN(delay, FFMAX(4096, n_out));
195
196		279722	outsamplesref = ff_get_audio_buffer(outlink, n_out);
197
198	✗✓	279722	if(!outsamplesref) {
199			av_frame_free(&insamplesref);
200			return AVERROR(ENOMEM);
201			}
202
203		279722	av_frame_copy_props(outsamplesref, insamplesref);
204		279722	outsamplesref->format = outlink->format;
205		279722	outsamplesref->channels = outlink->channels;
206		279722	outsamplesref->channel_layout = outlink->channel_layout;
207		279722	outsamplesref->sample_rate = outlink->sample_rate;
208
209	✓✗	279722	if(insamplesref->pts != AV_NOPTS_VALUE) {
210		279722	int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
211		279722	int64_t outpts= swr_next_pts(aresample->swr, inpts);
212		279722	aresample->next_pts =
213	✓✓	279722	outsamplesref->pts = ROUNDED_DIV(outpts, inlink->sample_rate);
214			} else {
215			outsamplesref->pts = AV_NOPTS_VALUE;
216			}
217		279722	n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
218		279722	(void *)insamplesref->extended_data, n_in);
219	✗✓	279722	if (n_out <= 0) {
220			av_frame_free(&outsamplesref);
221			av_frame_free(&insamplesref);
222			return 0;
223			}
224
225		279722	aresample->more_data = outsamplesref->nb_samples == n_out; // Indicate that there is probably more data in our buffers
226
227		279722	outsamplesref->nb_samples = n_out;
228
229		279722	ret = ff_filter_frame(outlink, outsamplesref);
230		279722	av_frame_free(&insamplesref);
231		279722	return ret;
232			}
233
234		1953	static int flush_frame(AVFilterLink outlink, int final, AVFrame *outsamplesref_ret)
235			{
236		1953	AVFilterContext *ctx = outlink->src;
237		1953	AResampleContext *aresample = ctx->priv;
238		1953	AVFilterLink *const inlink = outlink->src->inputs[0];
239			AVFrame *outsamplesref;
240		1953	int n_out = 4096;
241			int64_t pts;
242
243		1953	outsamplesref = ff_get_audio_buffer(outlink, n_out);
244		1953	*outsamplesref_ret = outsamplesref;
245	✗✓	1953	if (!outsamplesref)
246			return AVERROR(ENOMEM);
247
248		1953	pts = swr_next_pts(aresample->swr, INT64_MIN);
249	✓✗	1953	pts = ROUNDED_DIV(pts, inlink->sample_rate);
250
251	✓✓	1953	n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, final ? NULL : (void*)outsamplesref->extended_data, 0);
252	✓✓	1953	if (n_out <= 0) {
253		1191	av_frame_free(&outsamplesref);
254	✗✓	1191	return (n_out == 0) ? AVERROR_EOF : n_out;
255			}
256
257		762	outsamplesref->sample_rate = outlink->sample_rate;
258		762	outsamplesref->nb_samples = n_out;
259
260		762	outsamplesref->pts = pts;
261
262		762	return 0;
263			}
264
265		260131	static int request_frame(AVFilterLink *outlink)
266			{
267		260131	AVFilterContext *ctx = outlink->src;
268		260131	AResampleContext *aresample = ctx->priv;
269			int ret;
270
271			// First try to get data from the internal buffers
272	✓✓	260131	if (aresample->more_data) {
273			AVFrame *outsamplesref;
274
275	✓✓	123	if (flush_frame(outlink, 0, &outsamplesref) >= 0) {
276		121	return ff_filter_frame(outlink, outsamplesref);
277			}
278			}
279		260010	aresample->more_data = 0;
280
281			// Second request more data from the input
282		260010	ret = ff_request_frame(ctx->inputs[0]);
283
284			// Third if we hit the end flush
285	✓✓	260010	if (ret == AVERROR_EOF) {
286			AVFrame *outsamplesref;
287
288	✓✓	1830	if ((ret = flush_frame(outlink, 1, &outsamplesref)) < 0)
289		1189	return ret;
290
291		641	return ff_filter_frame(outlink, outsamplesref);
292			}
293		258180	return ret;
294			}
295
296			#if FF_API_CHILD_CLASS_NEXT
297			static const AVClass resample_child_class_next(const AVClass prev)
298			{
299			return prev ? NULL : swr_get_class();
300			}
301			#endif
302
303		2083	static const AVClass resample_child_class_iterate(void *iter)
304			{
305	✓✗	2083	const AVClass c = iter ? NULL : swr_get_class();
306		2083	iter = (void)(uintptr_t)c;
307		2083	return c;
308			}
309
310		5422	static void resample_child_next(void obj, void *prev)
311			{
312		5422	AResampleContext *s = obj;
313	✓✗	5422	return prev ? NULL : s->swr;
314			}
315
316			#define OFFSET(x) offsetof(AResampleContext, x)
317			#define FLAGS AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM
318
319			static const AVOption options[] = {
320			{"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS },
321			{NULL}
322			};
323
324			static const AVClass aresample_class = {
325			.class_name = "aresample",
326			.item_name = av_default_item_name,
327			.option = options,
328			.version = LIBAVUTIL_VERSION_INT,
329			#if FF_API_CHILD_CLASS_NEXT
330			.child_class_next = resample_child_class_next,
331			#endif
332			.child_class_iterate = resample_child_class_iterate,
333			.child_next = resample_child_next,
334			};
335
336			static const AVFilterPad aresample_inputs[] = {
337			{
338			.name = "default",
339			.type = AVMEDIA_TYPE_AUDIO,
340			.filter_frame = filter_frame,
341			},
342			{ NULL }
343			};
344
345			static const AVFilterPad aresample_outputs[] = {
346			{
347			.name = "default",
348			.config_props = config_output,
349			.request_frame = request_frame,
350			.type = AVMEDIA_TYPE_AUDIO,
351			},
352			{ NULL }
353			};
354
355			AVFilter ff_af_aresample = {
356			.name = "aresample",
357			.description = NULL_IF_CONFIG_SMALL("Resample audio data."),
358			.init_dict = init_dict,
359			.uninit = uninit,
360			.query_formats = query_formats,
361			.priv_size = sizeof(AResampleContext),
362			.priv_class = &aresample_class,
363			.inputs = aresample_inputs,
364			.outputs = aresample_outputs,
365			};