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/audioconvert.h"
28 #include "libavutil/avstring.h"
29 #include "libavutil/opt.h"
30 #include "libavutil/samplefmt.h"
31 #include "libavutil/avassert.h"
32 #include "libswresample/swresample.h"
39 struct SwrContext *swr;
44 static av_cold int init(AVFilterContext *ctx, const char *args)
46 AResampleContext *aresample = ctx->priv;
48 char *argd = av_strdup(args);
50 aresample->next_pts = AV_NOPTS_VALUE;
51 aresample->swr = swr_alloc();
53 return AVERROR(ENOMEM);
56 char *ptr=argd, *token;
58 while(token = av_strtok(ptr, ":", &ptr)) {
60 av_strtok(token, "=", &value);
63 if((ret=av_opt_set(aresample->swr, token, value, 0)) < 0)
67 if ((ret = ff_parse_sample_rate(&out_rate, token, ctx)) < 0)
69 if((ret = av_opt_set_int(aresample->swr, "osr", out_rate, 0)) < 0)
79 static av_cold void uninit(AVFilterContext *ctx)
81 AResampleContext *aresample = ctx->priv;
82 swr_free(&aresample->swr);
85 static int query_formats(AVFilterContext *ctx)
87 AResampleContext *aresample = ctx->priv;
88 int out_rate = av_get_int(aresample->swr, "osr", NULL);
89 uint64_t out_layout = av_get_int(aresample->swr, "ocl", NULL);
90 enum AVSampleFormat out_format = av_get_int(aresample->swr, "osf", NULL);
92 AVFilterLink *inlink = ctx->inputs[0];
93 AVFilterLink *outlink = ctx->outputs[0];
95 AVFilterFormats *in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
96 AVFilterFormats *out_formats;
97 AVFilterFormats *in_samplerates = ff_all_samplerates();
98 AVFilterFormats *out_samplerates;
99 AVFilterChannelLayouts *in_layouts = ff_all_channel_layouts();
100 AVFilterChannelLayouts *out_layouts;
102 ff_formats_ref (in_formats, &inlink->out_formats);
103 ff_formats_ref (in_samplerates, &inlink->out_samplerates);
104 ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
107 out_samplerates = ff_make_format_list((int[]){ out_rate, -1 });
109 out_samplerates = ff_all_samplerates();
111 ff_formats_ref(out_samplerates, &outlink->in_samplerates);
113 if(out_format != AV_SAMPLE_FMT_NONE) {
114 out_formats = ff_make_format_list((int[]){ out_format, -1 });
116 out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
117 ff_formats_ref(out_formats, &outlink->in_formats);
120 out_layouts = avfilter_make_format64_list((int64_t[]){ out_layout, -1 });
122 out_layouts = ff_all_channel_layouts();
123 ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
129 static int config_output(AVFilterLink *outlink)
132 AVFilterContext *ctx = outlink->src;
133 AVFilterLink *inlink = ctx->inputs[0];
134 AResampleContext *aresample = ctx->priv;
137 enum AVSampleFormat out_format;
138 char inchl_buf[128], outchl_buf[128];
140 aresample->swr = swr_alloc_set_opts(aresample->swr,
141 outlink->channel_layout, outlink->format, outlink->sample_rate,
142 inlink->channel_layout, inlink->format, inlink->sample_rate,
145 return AVERROR(ENOMEM);
147 ret = swr_init(aresample->swr);
151 out_rate = av_get_int(aresample->swr, "osr", NULL);
152 out_layout = av_get_int(aresample->swr, "ocl", NULL);
153 out_format = av_get_int(aresample->swr, "osf", NULL);
154 outlink->time_base = (AVRational) {1, out_rate};
156 av_assert0(outlink->sample_rate == out_rate);
157 av_assert0(outlink->channel_layout == out_layout);
158 av_assert0(outlink->format == out_format);
160 aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
162 av_get_channel_layout_string(inchl_buf, sizeof(inchl_buf), -1, inlink ->channel_layout);
163 av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), -1, outlink->channel_layout);
165 av_log(ctx, AV_LOG_INFO, "chl:%s fmt:%s r:%dHz -> chl:%s fmt:%s r:%dHz\n",
166 inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,
167 outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
171 static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
173 AResampleContext *aresample = inlink->dst->priv;
174 const int n_in = insamplesref->audio->nb_samples;
175 int n_out = n_in * aresample->ratio * 2 ;
176 AVFilterLink *const outlink = inlink->dst->outputs[0];
177 AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
180 avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
182 if(insamplesref->pts != AV_NOPTS_VALUE) {
183 int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
184 int64_t outpts= swr_next_pts(aresample->swr, inpts);
185 aresample->next_pts =
186 outsamplesref->pts = (outpts + inlink->sample_rate/2) / inlink->sample_rate;
188 outsamplesref->pts = AV_NOPTS_VALUE;
191 n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
192 (void *)insamplesref->extended_data, n_in);
194 avfilter_unref_buffer(outsamplesref);
195 avfilter_unref_buffer(insamplesref);
199 outsamplesref->audio->sample_rate = outlink->sample_rate;
200 outsamplesref->audio->nb_samples = n_out;
202 ff_filter_samples(outlink, outsamplesref);
203 aresample->req_fullfilled= 1;
204 avfilter_unref_buffer(insamplesref);
207 static int request_frame(AVFilterLink *outlink)
209 AVFilterContext *ctx = outlink->src;
210 AResampleContext *aresample = ctx->priv;
211 AVFilterLink *const inlink = outlink->src->inputs[0];
214 aresample->req_fullfilled = 0;
216 ret = ff_request_frame(ctx->inputs[0]);
217 }while(!aresample->req_fullfilled && ret>=0);
219 if (ret == AVERROR_EOF) {
220 AVFilterBufferRef *outsamplesref;
223 outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
225 return AVERROR(ENOMEM);
226 n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, 0, 0);
228 avfilter_unref_buffer(outsamplesref);
229 return (n_out == 0) ? AVERROR_EOF : n_out;
232 outsamplesref->audio->sample_rate = outlink->sample_rate;
233 outsamplesref->audio->nb_samples = n_out;
235 outsamplesref->pts = aresample->next_pts;
236 if(aresample->next_pts != AV_NOPTS_VALUE)
237 aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base);
239 outsamplesref->pts = (swr_next_pts(aresample->swr, INT64_MIN) + inlink->sample_rate/2) / inlink->sample_rate;
242 ff_filter_samples(outlink, outsamplesref);
248 AVFilter avfilter_af_aresample = {
250 .description = NULL_IF_CONFIG_SMALL("Resample audio data."),
253 .query_formats = query_formats,
254 .priv_size = sizeof(AResampleContext),
256 .inputs = (const AVFilterPad[]) {{ .name = "default",
257 .type = AVMEDIA_TYPE_AUDIO,
258 .filter_samples = filter_samples,
259 .min_perms = AV_PERM_READ, },
261 .outputs = (const AVFilterPad[]) {{ .name = "default",
262 .config_props = config_output,
263 .request_frame = request_frame,
264 .type = AVMEDIA_TYPE_AUDIO, },