2 * Copyright (c) 2011 Stefano Sabatini
3 * Copyright (c) 2011 Mina Nagy Zaki
5 * This file is part of FFmpeg.
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.
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.
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
24 * resampling audio filter
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"
41 struct SwrContext *swr;
47 static av_cold int init_dict(AVFilterContext *ctx, AVDictionary **opts)
49 AResampleContext *aresample = ctx->priv;
52 aresample->next_pts = AV_NOPTS_VALUE;
53 aresample->swr = swr_alloc();
54 if (!aresample->swr) {
55 ret = AVERROR(ENOMEM);
60 AVDictionaryEntry *e = NULL;
62 while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
63 if ((ret = av_opt_set(aresample->swr, e->key, e->value, 0)) < 0)
68 if (aresample->sample_rate_arg > 0)
69 av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0);
74 static av_cold void uninit(AVFilterContext *ctx)
76 AResampleContext *aresample = ctx->priv;
77 swr_free(&aresample->swr);
80 static int query_formats(AVFilterContext *ctx)
82 AResampleContext *aresample = ctx->priv;
83 int out_rate = av_get_int(aresample->swr, "osr", NULL);
84 uint64_t out_layout = av_get_int(aresample->swr, "ocl", NULL);
85 enum AVSampleFormat out_format = av_get_int(aresample->swr, "osf", NULL);
87 AVFilterLink *inlink = ctx->inputs[0];
88 AVFilterLink *outlink = ctx->outputs[0];
90 AVFilterFormats *in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
91 AVFilterFormats *out_formats;
92 AVFilterFormats *in_samplerates = ff_all_samplerates();
93 AVFilterFormats *out_samplerates;
94 AVFilterChannelLayouts *in_layouts = ff_all_channel_counts();
95 AVFilterChannelLayouts *out_layouts;
97 ff_formats_ref (in_formats, &inlink->out_formats);
98 ff_formats_ref (in_samplerates, &inlink->out_samplerates);
99 ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
102 int ratelist[] = { out_rate, -1 };
103 out_samplerates = ff_make_format_list(ratelist);
105 out_samplerates = ff_all_samplerates();
107 if (!out_samplerates) {
108 av_log(ctx, AV_LOG_ERROR, "Cannot allocate output samplerates.\n");
109 return AVERROR(ENOMEM);
112 ff_formats_ref(out_samplerates, &outlink->in_samplerates);
114 if(out_format != AV_SAMPLE_FMT_NONE) {
115 int formatlist[] = { out_format, -1 };
116 out_formats = ff_make_format_list(formatlist);
118 out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
119 ff_formats_ref(out_formats, &outlink->in_formats);
122 int64_t layout_list[] = { out_layout, -1 };
123 out_layouts = avfilter_make_format64_list(layout_list);
125 out_layouts = ff_all_channel_counts();
126 ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
132 static int config_output(AVFilterLink *outlink)
135 AVFilterContext *ctx = outlink->src;
136 AVFilterLink *inlink = ctx->inputs[0];
137 AResampleContext *aresample = ctx->priv;
140 enum AVSampleFormat out_format;
141 char inchl_buf[128], outchl_buf[128];
143 aresample->swr = swr_alloc_set_opts(aresample->swr,
144 outlink->channel_layout, outlink->format, outlink->sample_rate,
145 inlink->channel_layout, inlink->format, inlink->sample_rate,
148 return AVERROR(ENOMEM);
149 if (!inlink->channel_layout)
150 av_opt_set_int(aresample->swr, "ich", inlink->channels, 0);
151 if (!outlink->channel_layout)
152 av_opt_set_int(aresample->swr, "och", outlink->channels, 0);
154 ret = swr_init(aresample->swr);
158 out_rate = av_get_int(aresample->swr, "osr", NULL);
159 out_layout = av_get_int(aresample->swr, "ocl", NULL);
160 out_format = av_get_int(aresample->swr, "osf", NULL);
161 outlink->time_base = (AVRational) {1, out_rate};
163 av_assert0(outlink->sample_rate == out_rate);
164 av_assert0(outlink->channel_layout == out_layout || !outlink->channel_layout);
165 av_assert0(outlink->format == out_format);
167 aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
169 av_get_channel_layout_string(inchl_buf, sizeof(inchl_buf), inlink ->channels, inlink ->channel_layout);
170 av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), outlink->channels, outlink->channel_layout);
172 av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",
173 inlink ->channels, inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,
174 outlink->channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
178 static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref)
180 AResampleContext *aresample = inlink->dst->priv;
181 const int n_in = insamplesref->nb_samples;
183 int n_out = n_in * aresample->ratio + 32;
184 AVFilterLink *const outlink = inlink->dst->outputs[0];
185 AVFrame *outsamplesref;
188 delay = swr_get_delay(aresample->swr, outlink->sample_rate);
190 n_out += FFMIN(delay, FFMAX(4096, n_out));
192 outsamplesref = ff_get_audio_buffer(outlink, n_out);
195 return AVERROR(ENOMEM);
197 av_frame_copy_props(outsamplesref, insamplesref);
198 outsamplesref->format = outlink->format;
199 av_frame_set_channels(outsamplesref, outlink->channels);
200 outsamplesref->channel_layout = outlink->channel_layout;
201 outsamplesref->sample_rate = outlink->sample_rate;
203 if(insamplesref->pts != AV_NOPTS_VALUE) {
204 int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
205 int64_t outpts= swr_next_pts(aresample->swr, inpts);
206 aresample->next_pts =
207 outsamplesref->pts = ROUNDED_DIV(outpts, inlink->sample_rate);
209 outsamplesref->pts = AV_NOPTS_VALUE;
211 n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
212 (void *)insamplesref->extended_data, n_in);
214 av_frame_free(&outsamplesref);
215 av_frame_free(&insamplesref);
219 aresample->more_data = outsamplesref->nb_samples == n_out; // Indicate that there is probably more data in our buffers
221 outsamplesref->nb_samples = n_out;
223 ret = ff_filter_frame(outlink, outsamplesref);
224 aresample->req_fullfilled= 1;
225 av_frame_free(&insamplesref);
229 static int flush_frame(AVFilterLink *outlink, int final, AVFrame **outsamplesref_ret)
231 AVFilterContext *ctx = outlink->src;
232 AResampleContext *aresample = ctx->priv;
233 AVFilterLink *const inlink = outlink->src->inputs[0];
234 AVFrame *outsamplesref;
238 outsamplesref = ff_get_audio_buffer(outlink, n_out);
239 *outsamplesref_ret = outsamplesref;
241 return AVERROR(ENOMEM);
243 pts = swr_next_pts(aresample->swr, INT64_MIN);
244 pts = ROUNDED_DIV(pts, inlink->sample_rate);
246 n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, final ? NULL : (void*)outsamplesref->extended_data, 0);
248 av_frame_free(&outsamplesref);
249 return (n_out == 0) ? AVERROR_EOF : n_out;
252 outsamplesref->sample_rate = outlink->sample_rate;
253 outsamplesref->nb_samples = n_out;
255 outsamplesref->pts = pts;
260 static int request_frame(AVFilterLink *outlink)
262 AVFilterContext *ctx = outlink->src;
263 AResampleContext *aresample = ctx->priv;
266 // First try to get data from the internal buffers
267 if (aresample->more_data) {
268 AVFrame *outsamplesref;
270 if (flush_frame(outlink, 0, &outsamplesref) >= 0) {
271 return ff_filter_frame(outlink, outsamplesref);
274 aresample->more_data = 0;
276 // Second request more data from the input
277 aresample->req_fullfilled = 0;
279 ret = ff_request_frame(ctx->inputs[0]);
280 }while(!aresample->req_fullfilled && ret>=0);
282 // Third if we hit the end flush
283 if (ret == AVERROR_EOF) {
284 AVFrame *outsamplesref;
286 if ((ret = flush_frame(outlink, 1, &outsamplesref)) < 0)
289 return ff_filter_frame(outlink, outsamplesref);
294 static const AVClass *resample_child_class_next(const AVClass *prev)
296 return prev ? NULL : swr_get_class();
299 static void *resample_child_next(void *obj, void *prev)
301 AResampleContext *s = obj;
302 return prev ? NULL : s->swr;
305 #define OFFSET(x) offsetof(AResampleContext, x)
306 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
308 static const AVOption options[] = {
309 {"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS },
313 static const AVClass aresample_class = {
314 .class_name = "aresample",
315 .item_name = av_default_item_name,
317 .version = LIBAVUTIL_VERSION_INT,
318 .child_class_next = resample_child_class_next,
319 .child_next = resample_child_next,
322 static const AVFilterPad aresample_inputs[] = {
325 .type = AVMEDIA_TYPE_AUDIO,
326 .filter_frame = filter_frame,
331 static const AVFilterPad aresample_outputs[] = {
334 .config_props = config_output,
335 .request_frame = request_frame,
336 .type = AVMEDIA_TYPE_AUDIO,
341 AVFilter ff_af_aresample = {
343 .description = NULL_IF_CONFIG_SMALL("Resample audio data."),
344 .init_dict = init_dict,
346 .query_formats = query_formats,
347 .priv_size = sizeof(AResampleContext),
348 .priv_class = &aresample_class,
349 .inputs = aresample_inputs,
350 .outputs = aresample_outputs,