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 %

Line Branch Exec Source
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/*
2
 * Copyright (c) 2011 Stefano Sabatini
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 * Copyright (c) 2011 Mina Nagy Zaki
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 *
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 * This file is part of FFmpeg.
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 *
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 * FFmpeg is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * FFmpeg is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
21
22
/**
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 * @file
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 * resampling audio filter
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 */
26
27
#include "libavutil/avstring.h"
28
#include "libavutil/channel_layout.h"
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#include "libavutil/opt.h"
30
#include "libavutil/samplefmt.h"
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#include "libavutil/avassert.h"
32
#include "libswresample/swresample.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "internal.h"
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37
typedef struct AResampleContext {
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    const AVClass *class;
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    int sample_rate_arg;
40
    double ratio;
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    struct SwrContext *swr;
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    int64_t next_pts;
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    int more_data;
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} AResampleContext;
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46
1201
static av_cold int init_dict(AVFilterContext *ctx, AVDictionary **opts)
47
{
48
1201
    AResampleContext *aresample = ctx->priv;
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1201
    int ret = 0;
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51
1201
    aresample->next_pts = AV_NOPTS_VALUE;
52
1201
    aresample->swr = swr_alloc();
53
1201
    if (!aresample->swr) {
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        ret = AVERROR(ENOMEM);
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        goto end;
56
    }
57
58
1201
    if (opts) {
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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)
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                goto end;
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        }
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1201
        av_dict_free(opts);
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    }
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1201
    if (aresample->sample_rate_arg > 0)
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628
        av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0);
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573
end:
70
1201
    return ret;
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}
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73
1201
static av_cold void uninit(AVFilterContext *ctx)
74
{
75
1201
    AResampleContext *aresample = ctx->priv;
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1201
    swr_free(&aresample->swr);
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1201
}
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79
1201
static int query_formats(AVFilterContext *ctx)
80
{
81
1201
    AResampleContext *aresample = ctx->priv;
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    enum AVSampleFormat out_format;
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    int64_t out_rate, out_layout;
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85
1201
    AVFilterLink *inlink  = ctx->inputs[0];
86
1201
    AVFilterLink *outlink = ctx->outputs[0];
87
88
    AVFilterFormats        *in_formats, *out_formats;
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    AVFilterFormats        *in_samplerates, *out_samplerates;
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    AVFilterChannelLayouts *in_layouts, *out_layouts;
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    int ret;
92
93
1201
    av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
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1201
    av_opt_get_int(aresample->swr, "osr", 0, &out_rate);
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1201
    av_opt_get_int(aresample->swr, "ocl", 0, &out_layout);
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97
1201
    in_formats      = ff_all_formats(AVMEDIA_TYPE_AUDIO);
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1201
    if ((ret = ff_formats_ref(in_formats, &inlink->out_formats)) < 0)
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        return ret;
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101
1201
    in_samplerates  = ff_all_samplerates();
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1201
    if ((ret = ff_formats_ref(in_samplerates, &inlink->out_samplerates)) < 0)
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        return ret;
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105
1201
    in_layouts      = ff_all_channel_counts();
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1201
    if ((ret = ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts)) < 0)
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        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();
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    }
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116
1201
    if ((ret = ff_formats_ref(out_samplerates, &outlink->in_samplerates)) < 0)
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        return ret;
118
119
1201
    if(out_format != AV_SAMPLE_FMT_NONE) {
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        int formatlist[] = { out_format, -1 };
121
        out_formats = ff_make_format_list(formatlist);
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    } 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);
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1201
    av_opt_get_sample_fmt(aresample->swr, "osf", 0, &out_format);
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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);
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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));
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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
};