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
37typedef struct {
38    const AVClass *class;
39    int sample_rate_arg;
40    double ratio;
41    struct SwrContext *swr;
42    int64_t next_pts;
43    int req_fullfilled;
44} AResampleContext;
45
46static av_cold int init_dict(AVFilterContext *ctx, AVDictionary **opts)
47{
48    AResampleContext *aresample = ctx->priv;
49    int ret = 0;
50
51    aresample->next_pts = AV_NOPTS_VALUE;
52    aresample->swr = swr_alloc();
53    if (!aresample->swr) {
54        ret = AVERROR(ENOMEM);
55        goto end;
56    }
57
58    if (opts) {
59        AVDictionaryEntry *e = NULL;
60
61        while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
62            if ((ret = av_opt_set(aresample->swr, e->key, e->value, 0)) < 0)
63                goto end;
64        }
65        av_dict_free(opts);
66    }
67    if (aresample->sample_rate_arg > 0)
68        av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0);
69end:
70    return ret;
71}
72
73static av_cold void uninit(AVFilterContext *ctx)
74{
75    AResampleContext *aresample = ctx->priv;
76    swr_free(&aresample->swr);
77}
78
79static int query_formats(AVFilterContext *ctx)
80{
81    AResampleContext *aresample = ctx->priv;
82    int out_rate                   = av_get_int(aresample->swr, "osr", NULL);
83    uint64_t out_layout            = av_get_int(aresample->swr, "ocl", NULL);
84    enum AVSampleFormat out_format = av_get_int(aresample->swr, "osf", NULL);
85
86    AVFilterLink *inlink  = ctx->inputs[0];
87    AVFilterLink *outlink = ctx->outputs[0];
88
89    AVFilterFormats        *in_formats      = ff_all_formats(AVMEDIA_TYPE_AUDIO);
90    AVFilterFormats        *out_formats;
91    AVFilterFormats        *in_samplerates  = ff_all_samplerates();
92    AVFilterFormats        *out_samplerates;
93    AVFilterChannelLayouts *in_layouts      = ff_all_channel_counts();
94    AVFilterChannelLayouts *out_layouts;
95
96    ff_formats_ref  (in_formats,      &inlink->out_formats);
97    ff_formats_ref  (in_samplerates,  &inlink->out_samplerates);
98    ff_channel_layouts_ref(in_layouts,      &inlink->out_channel_layouts);
99
100    if(out_rate > 0) {
101        int ratelist[] = { out_rate, -1 };
102        out_samplerates = ff_make_format_list(ratelist);
103    } else {
104        out_samplerates = ff_all_samplerates();
105    }
106    ff_formats_ref(out_samplerates, &outlink->in_samplerates);
107
108    if(out_format != AV_SAMPLE_FMT_NONE) {
109        int formatlist[] = { out_format, -1 };
110        out_formats = ff_make_format_list(formatlist);
111    } else
112        out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
113    ff_formats_ref(out_formats, &outlink->in_formats);
114
115    if(out_layout) {
116        int64_t layout_list[] = { out_layout, -1 };
117        out_layouts = avfilter_make_format64_list(layout_list);
118    } else
119        out_layouts = ff_all_channel_counts();
120    ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
121
122    return 0;
123}
124
125
126static int config_output(AVFilterLink *outlink)
127{
128    int ret;
129    AVFilterContext *ctx = outlink->src;
130    AVFilterLink *inlink = ctx->inputs[0];
131    AResampleContext *aresample = ctx->priv;
132    int out_rate;
133    uint64_t out_layout;
134    enum AVSampleFormat out_format;
135    char inchl_buf[128], outchl_buf[128];
136
137    aresample->swr = swr_alloc_set_opts(aresample->swr,
138                                        outlink->channel_layout, outlink->format, outlink->sample_rate,
139                                        inlink->channel_layout, inlink->format, inlink->sample_rate,
140                                        0, ctx);
141    if (!aresample->swr)
142        return AVERROR(ENOMEM);
143    if (!inlink->channel_layout)
144        av_opt_set_int(aresample->swr, "ich", inlink->channels, 0);
145    if (!outlink->channel_layout)
146        av_opt_set_int(aresample->swr, "och", outlink->channels, 0);
147
148    ret = swr_init(aresample->swr);
149    if (ret < 0)
150        return ret;
151
152    out_rate   = av_get_int(aresample->swr, "osr", NULL);
153    out_layout = av_get_int(aresample->swr, "ocl", NULL);
154    out_format = av_get_int(aresample->swr, "osf", NULL);
155    outlink->time_base = (AVRational) {1, out_rate};
156
157    av_assert0(outlink->sample_rate == out_rate);
158    av_assert0(outlink->channel_layout == out_layout || !outlink->channel_layout);
159    av_assert0(outlink->format == out_format);
160
161    aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
162
163    av_get_channel_layout_string(inchl_buf,  sizeof(inchl_buf),  inlink ->channels, inlink ->channel_layout);
164    av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), outlink->channels, outlink->channel_layout);
165
166    av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",
167           inlink ->channels, inchl_buf,  av_get_sample_fmt_name(inlink->format),  inlink->sample_rate,
168           outlink->channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
169    return 0;
170}
171
172static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref)
173{
174    AResampleContext *aresample = inlink->dst->priv;
175    const int n_in  = insamplesref->nb_samples;
176    int64_t delay;
177    int n_out       = n_in * aresample->ratio + 32;
178    AVFilterLink *const outlink = inlink->dst->outputs[0];
179    AVFrame *outsamplesref;
180    int ret;
181
182    delay = swr_get_delay(aresample->swr, outlink->sample_rate);
183    if (delay > 0)
184        n_out += delay;
185
186    outsamplesref = ff_get_audio_buffer(outlink, n_out);
187
188    if(!outsamplesref)
189        return AVERROR(ENOMEM);
190
191    av_frame_copy_props(outsamplesref, insamplesref);
192    outsamplesref->format                = outlink->format;
193    av_frame_set_channels(outsamplesref, outlink->channels);
194    outsamplesref->channel_layout        = outlink->channel_layout;
195    outsamplesref->sample_rate           = outlink->sample_rate;
196
197    if(insamplesref->pts != AV_NOPTS_VALUE) {
198        int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
199        int64_t outpts= swr_next_pts(aresample->swr, inpts);
200        aresample->next_pts =
201        outsamplesref->pts  = ROUNDED_DIV(outpts, inlink->sample_rate);
202    } else {
203        outsamplesref->pts  = AV_NOPTS_VALUE;
204    }
205    n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
206                                 (void *)insamplesref->extended_data, n_in);
207    if (n_out <= 0) {
208        av_frame_free(&outsamplesref);
209        av_frame_free(&insamplesref);
210        return 0;
211    }
212
213    outsamplesref->nb_samples  = n_out;
214
215    ret = ff_filter_frame(outlink, outsamplesref);
216    aresample->req_fullfilled= 1;
217    av_frame_free(&insamplesref);
218    return ret;
219}
220
221static int request_frame(AVFilterLink *outlink)
222{
223    AVFilterContext *ctx = outlink->src;
224    AResampleContext *aresample = ctx->priv;
225    AVFilterLink *const inlink = outlink->src->inputs[0];
226    int ret;
227
228    aresample->req_fullfilled = 0;
229    do{
230        ret = ff_request_frame(ctx->inputs[0]);
231    }while(!aresample->req_fullfilled && ret>=0);
232
233    if (ret == AVERROR_EOF) {
234        AVFrame *outsamplesref;
235        int n_out = 4096;
236        int64_t pts;
237
238        outsamplesref = ff_get_audio_buffer(outlink, n_out);
239        if (!outsamplesref)
240            return AVERROR(ENOMEM);
241
242        pts = swr_next_pts(aresample->swr, INT64_MIN);
243        pts = ROUNDED_DIV(pts, inlink->sample_rate);
244
245        n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, 0, 0);
246        if (n_out <= 0) {
247            av_frame_free(&outsamplesref);
248            return (n_out == 0) ? AVERROR_EOF : n_out;
249        }
250
251        outsamplesref->sample_rate = outlink->sample_rate;
252        outsamplesref->nb_samples  = n_out;
253
254        outsamplesref->pts = pts;
255
256        return ff_filter_frame(outlink, outsamplesref);
257    }
258    return ret;
259}
260
261static const AVClass *resample_child_class_next(const AVClass *prev)
262{
263    return prev ? NULL : swr_get_class();
264}
265
266static void *resample_child_next(void *obj, void *prev)
267{
268    AResampleContext *s = obj;
269    return prev ? NULL : s->swr;
270}
271
272#define OFFSET(x) offsetof(AResampleContext, x)
273#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
274
275static const AVOption options[] = {
276    {"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0},  0,        INT_MAX, FLAGS },
277    {NULL}
278};
279
280static const AVClass aresample_class = {
281    .class_name       = "aresample",
282    .item_name        = av_default_item_name,
283    .option           = options,
284    .version          = LIBAVUTIL_VERSION_INT,
285    .child_class_next = resample_child_class_next,
286    .child_next       = resample_child_next,
287};
288
289static const AVFilterPad aresample_inputs[] = {
290    {
291        .name         = "default",
292        .type         = AVMEDIA_TYPE_AUDIO,
293        .filter_frame = filter_frame,
294    },
295    { NULL }
296};
297
298static const AVFilterPad aresample_outputs[] = {
299    {
300        .name          = "default",
301        .config_props  = config_output,
302        .request_frame = request_frame,
303        .type          = AVMEDIA_TYPE_AUDIO,
304    },
305    { NULL }
306};
307
308AVFilter ff_af_aresample = {
309    .name          = "aresample",
310    .description   = NULL_IF_CONFIG_SMALL("Resample audio data."),
311    .init_dict     = init_dict,
312    .uninit        = uninit,
313    .query_formats = query_formats,
314    .priv_size     = sizeof(AResampleContext),
315    .priv_class    = &aresample_class,
316    .inputs        = aresample_inputs,
317    .outputs       = aresample_outputs,
318};
319