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;
46 static av_cold int init_dict(AVFilterContext *ctx, AVDictionary **opts)
48 AResampleContext *aresample = ctx->priv;
51 aresample->next_pts = AV_NOPTS_VALUE;
52 aresample->swr = swr_alloc();
53 if (!aresample->swr) {
54 ret = AVERROR(ENOMEM);
59 AVDictionaryEntry *e = NULL;
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)
67 if (aresample->sample_rate_arg > 0)
68 av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0);
73 static av_cold void uninit(AVFilterContext *ctx)
75 AResampleContext *aresample = ctx->priv;
76 swr_free(&aresample->swr);
79 static int query_formats(AVFilterContext *ctx)
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);
86 AVFilterLink *inlink = ctx->inputs[0];
87 AVFilterLink *outlink = ctx->outputs[0];
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;
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);
101 out_samplerates = ff_make_format_list((int[]){ out_rate, -1 });
103 out_samplerates = ff_all_samplerates();
105 ff_formats_ref(out_samplerates, &outlink->in_samplerates);
107 if(out_format != AV_SAMPLE_FMT_NONE) {
108 out_formats = ff_make_format_list((int[]){ out_format, -1 });
110 out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
111 ff_formats_ref(out_formats, &outlink->in_formats);
114 out_layouts = avfilter_make_format64_list((int64_t[]){ out_layout, -1 });
116 out_layouts = ff_all_channel_counts();
117 ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
123 static int config_output(AVFilterLink *outlink)
126 AVFilterContext *ctx = outlink->src;
127 AVFilterLink *inlink = ctx->inputs[0];
128 AResampleContext *aresample = ctx->priv;
131 enum AVSampleFormat out_format;
132 char inchl_buf[128], outchl_buf[128];
134 aresample->swr = swr_alloc_set_opts(aresample->swr,
135 outlink->channel_layout, outlink->format, outlink->sample_rate,
136 inlink->channel_layout, inlink->format, inlink->sample_rate,
139 return AVERROR(ENOMEM);
140 if (!inlink->channel_layout)
141 av_opt_set_int(aresample->swr, "ich", inlink->channels, 0);
142 if (!outlink->channel_layout)
143 av_opt_set_int(aresample->swr, "och", outlink->channels, 0);
145 ret = swr_init(aresample->swr);
149 out_rate = av_get_int(aresample->swr, "osr", NULL);
150 out_layout = av_get_int(aresample->swr, "ocl", NULL);
151 out_format = av_get_int(aresample->swr, "osf", NULL);
152 outlink->time_base = (AVRational) {1, out_rate};
154 av_assert0(outlink->sample_rate == out_rate);
155 av_assert0(outlink->channel_layout == out_layout || !outlink->channel_layout);
156 av_assert0(outlink->format == out_format);
158 aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
160 av_get_channel_layout_string(inchl_buf, sizeof(inchl_buf), inlink ->channels, inlink ->channel_layout);
161 av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), outlink->channels, outlink->channel_layout);
163 av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",
164 inlink ->channels, inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,
165 outlink->channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
169 static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref)
171 AResampleContext *aresample = inlink->dst->priv;
172 const int n_in = insamplesref->nb_samples;
173 int n_out = n_in * aresample->ratio * 2 + 256;
174 AVFilterLink *const outlink = inlink->dst->outputs[0];
175 AVFrame *outsamplesref = ff_get_audio_buffer(outlink, n_out);
179 return AVERROR(ENOMEM);
181 av_frame_copy_props(outsamplesref, insamplesref);
182 outsamplesref->format = outlink->format;
183 av_frame_set_channels(outsamplesref, outlink->channels);
184 outsamplesref->channel_layout = outlink->channel_layout;
185 outsamplesref->sample_rate = outlink->sample_rate;
187 if(insamplesref->pts != AV_NOPTS_VALUE) {
188 int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
189 int64_t outpts= swr_next_pts(aresample->swr, inpts);
190 aresample->next_pts =
191 outsamplesref->pts = ROUNDED_DIV(outpts, inlink->sample_rate);
193 outsamplesref->pts = AV_NOPTS_VALUE;
195 n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
196 (void *)insamplesref->extended_data, n_in);
198 av_frame_free(&outsamplesref);
199 av_frame_free(&insamplesref);
203 outsamplesref->nb_samples = n_out;
205 ret = ff_filter_frame(outlink, outsamplesref);
206 aresample->req_fullfilled= 1;
207 av_frame_free(&insamplesref);
211 static int request_frame(AVFilterLink *outlink)
213 AVFilterContext *ctx = outlink->src;
214 AResampleContext *aresample = ctx->priv;
215 AVFilterLink *const inlink = outlink->src->inputs[0];
218 aresample->req_fullfilled = 0;
220 ret = ff_request_frame(ctx->inputs[0]);
221 }while(!aresample->req_fullfilled && ret>=0);
223 if (ret == AVERROR_EOF) {
224 AVFrame *outsamplesref;
227 outsamplesref = ff_get_audio_buffer(outlink, n_out);
229 return AVERROR(ENOMEM);
230 n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, 0, 0);
232 av_frame_free(&outsamplesref);
233 return (n_out == 0) ? AVERROR_EOF : n_out;
236 outsamplesref->sample_rate = outlink->sample_rate;
237 outsamplesref->nb_samples = n_out;
239 outsamplesref->pts = aresample->next_pts;
240 if(aresample->next_pts != AV_NOPTS_VALUE)
241 aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base);
243 outsamplesref->pts = swr_next_pts(aresample->swr, INT64_MIN);
244 outsamplesref->pts = ROUNDED_DIV(outsamplesref->pts, inlink->sample_rate);
247 return ff_filter_frame(outlink, outsamplesref);
252 static const AVClass *resample_child_class_next(const AVClass *prev)
254 return prev ? NULL : swr_get_class();
257 static void *resample_child_next(void *obj, void *prev)
259 AResampleContext *s = obj;
260 return prev ? NULL : s->swr;
263 #define OFFSET(x) offsetof(AResampleContext, x)
264 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
266 static const AVOption options[] = {
267 {"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS },
271 static const AVClass aresample_class = {
272 .class_name = "aresample",
273 .item_name = av_default_item_name,
275 .version = LIBAVUTIL_VERSION_INT,
276 .child_class_next = resample_child_class_next,
277 .child_next = resample_child_next,
280 static const AVFilterPad aresample_inputs[] = {
283 .type = AVMEDIA_TYPE_AUDIO,
284 .filter_frame = filter_frame,
289 static const AVFilterPad aresample_outputs[] = {
292 .config_props = config_output,
293 .request_frame = request_frame,
294 .type = AVMEDIA_TYPE_AUDIO,
299 AVFilter avfilter_af_aresample = {
301 .description = NULL_IF_CONFIG_SMALL("Resample audio data."),
302 .init_dict = init_dict,
304 .query_formats = query_formats,
305 .priv_size = sizeof(AResampleContext),
306 .priv_class = &aresample_class,
307 .inputs = aresample_inputs,
308 .outputs = aresample_outputs,