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/opt.h"
29 #include "libavutil/samplefmt.h"
30 #include "libavutil/avassert.h"
31 #include "libswresample/swresample.h"
38 struct SwrContext *swr;
43 static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
45 AResampleContext *aresample = ctx->priv;
47 char *argd = av_strdup(args);
49 aresample->next_pts = AV_NOPTS_VALUE;
50 aresample->swr = swr_alloc();
52 return AVERROR(ENOMEM);
55 char *ptr=argd, *token;
57 while(token = av_strtok(ptr, ":", &ptr)) {
59 av_strtok(token, "=", &value);
62 if((ret=av_opt_set(aresample->swr, token, value, 0)) < 0)
66 if ((ret = ff_parse_sample_rate(&out_rate, token, ctx)) < 0)
68 if((ret = av_opt_set_int(aresample->swr, "osr", out_rate, 0)) < 0)
78 static av_cold void uninit(AVFilterContext *ctx)
80 AResampleContext *aresample = ctx->priv;
81 swr_free(&aresample->swr);
84 static int query_formats(AVFilterContext *ctx)
86 AResampleContext *aresample = ctx->priv;
87 int out_rate = av_get_int(aresample->swr, "osr", NULL);
88 uint64_t out_layout = av_get_int(aresample->swr, "ocl", NULL);
89 enum AVSampleFormat out_format = av_get_int(aresample->swr, "osf", NULL);
91 AVFilterLink *inlink = ctx->inputs[0];
92 AVFilterLink *outlink = ctx->outputs[0];
94 AVFilterFormats *in_formats = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
95 AVFilterFormats *out_formats;
96 AVFilterFormats *in_samplerates = ff_all_samplerates();
97 AVFilterFormats *out_samplerates;
98 AVFilterChannelLayouts *in_layouts = ff_all_channel_layouts();
99 AVFilterChannelLayouts *out_layouts;
101 avfilter_formats_ref (in_formats, &inlink->out_formats);
102 avfilter_formats_ref (in_samplerates, &inlink->out_samplerates);
103 ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
106 out_samplerates = avfilter_make_format_list((int[]){ out_rate, -1 });
108 out_samplerates = ff_all_samplerates();
110 avfilter_formats_ref(out_samplerates, &outlink->in_samplerates);
112 if(out_format != AV_SAMPLE_FMT_NONE) {
113 out_formats = avfilter_make_format_list((int[]){ out_format, -1 });
115 out_formats = avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO);
116 avfilter_formats_ref(out_formats, &outlink->in_formats);
119 out_layouts = avfilter_make_format64_list((int64_t[]){ out_layout, -1 });
121 out_layouts = ff_all_channel_layouts();
122 ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
128 static int config_output(AVFilterLink *outlink)
131 AVFilterContext *ctx = outlink->src;
132 AVFilterLink *inlink = ctx->inputs[0];
133 AResampleContext *aresample = ctx->priv;
136 enum AVSampleFormat out_format;
137 char inchl_buf[128], outchl_buf[128];
139 aresample->swr = swr_alloc_set_opts(aresample->swr,
140 outlink->channel_layout, outlink->format, outlink->sample_rate,
141 inlink->channel_layout, inlink->format, inlink->sample_rate,
144 return AVERROR(ENOMEM);
146 ret = swr_init(aresample->swr);
150 out_rate = av_get_int(aresample->swr, "osr", NULL);
151 out_layout = av_get_int(aresample->swr, "ocl", NULL);
152 out_format = av_get_int(aresample->swr, "osf", NULL);
153 outlink->time_base = (AVRational) {1, out_rate};
155 av_assert0(outlink->sample_rate == out_rate);
156 av_assert0(outlink->channel_layout == out_layout);
157 av_assert0(outlink->format == out_format);
159 aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
161 av_get_channel_layout_string(inchl_buf, sizeof(inchl_buf), -1, inlink ->channel_layout);
162 av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), -1, outlink->channel_layout);
163 av_log(ctx, AV_LOG_INFO, "chl:%s fmt:%s r:%"PRId64"Hz -> chl:%s fmt:%s r:%"PRId64"Hz\n",
164 inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,
165 outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
169 static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
171 AResampleContext *aresample = inlink->dst->priv;
172 const int n_in = insamplesref->audio->nb_samples;
173 int n_out = n_in * aresample->ratio * 2 ;
174 AVFilterLink *const outlink = inlink->dst->outputs[0];
175 AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
178 avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
180 if(insamplesref->pts != AV_NOPTS_VALUE) {
181 int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
182 int64_t outpts= swr_next_pts(aresample->swr, inpts);
183 aresample->next_pts =
184 outsamplesref->pts = (outpts + inlink->sample_rate/2) / inlink->sample_rate;
186 outsamplesref->pts = AV_NOPTS_VALUE;
189 n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
190 (void *)insamplesref->extended_data, n_in);
192 avfilter_unref_buffer(outsamplesref);
193 avfilter_unref_buffer(insamplesref);
197 outsamplesref->audio->sample_rate = outlink->sample_rate;
198 outsamplesref->audio->nb_samples = n_out;
200 ff_filter_samples(outlink, outsamplesref);
201 aresample->req_fullfilled= 1;
202 avfilter_unref_buffer(insamplesref);
205 static int request_frame(AVFilterLink *outlink)
207 AVFilterContext *ctx = outlink->src;
208 AResampleContext *aresample = ctx->priv;
209 AVFilterLink *const inlink = outlink->src->inputs[0];
212 aresample->req_fullfilled = 0;
214 ret = avfilter_request_frame(ctx->inputs[0]);
215 }while(!aresample->req_fullfilled && ret>=0);
217 if (ret == AVERROR_EOF) {
218 AVFilterBufferRef *outsamplesref;
221 outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
223 return AVERROR(ENOMEM);
224 n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, 0, 0);
226 avfilter_unref_buffer(outsamplesref);
227 return (n_out == 0) ? AVERROR_EOF : n_out;
230 outsamplesref->audio->sample_rate = outlink->sample_rate;
231 outsamplesref->audio->nb_samples = n_out;
233 outsamplesref->pts = aresample->next_pts;
234 if(aresample->next_pts != AV_NOPTS_VALUE)
235 aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base);
237 outsamplesref->pts = (swr_next_pts(aresample->swr, INT64_MIN) + inlink->sample_rate/2) / inlink->sample_rate;
240 ff_filter_samples(outlink, outsamplesref);
246 AVFilter avfilter_af_aresample = {
248 .description = NULL_IF_CONFIG_SMALL("Resample audio data."),
251 .query_formats = query_formats,
252 .priv_size = sizeof(AResampleContext),
254 .inputs = (const AVFilterPad[]) {{ .name = "default",
255 .type = AVMEDIA_TYPE_AUDIO,
256 .filter_samples = filter_samples,
257 .min_perms = AV_PERM_READ, },
259 .outputs = (const AVFilterPad[]) {{ .name = "default",
260 .config_props = config_output,
261 .request_frame = request_frame,
262 .type = AVMEDIA_TYPE_AUDIO, },