3 * This file is part of Libav.
5 * Libav is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Lesser General Public
7 * License as published by the Free Software Foundation; either
8 * version 2.1 of the License, or (at your option) any later version.
10 * Libav is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Lesser General Public License for more details.
15 * You should have received a copy of the GNU Lesser General Public
16 * License along with Libav; if not, write to the Free Software
17 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 * sample format and channel layout conversion audio filter
25 #include "libavutil/avassert.h"
26 #include "libavutil/avstring.h"
27 #include "libavutil/common.h"
28 #include "libavutil/mathematics.h"
29 #include "libavutil/opt.h"
31 #include "libavresample/avresample.h"
38 typedef struct ResampleContext {
39 AVAudioResampleContext *avr;
43 /* set by filter_samples() to signal an output frame to request_frame() */
47 static av_cold void uninit(AVFilterContext *ctx)
49 ResampleContext *s = ctx->priv;
52 avresample_close(s->avr);
53 avresample_free(&s->avr);
57 static int query_formats(AVFilterContext *ctx)
59 AVFilterLink *inlink = ctx->inputs[0];
60 AVFilterLink *outlink = ctx->outputs[0];
62 AVFilterFormats *in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
63 AVFilterFormats *out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
64 AVFilterFormats *in_samplerates = ff_all_samplerates();
65 AVFilterFormats *out_samplerates = ff_all_samplerates();
66 AVFilterChannelLayouts *in_layouts = ff_all_channel_layouts();
67 AVFilterChannelLayouts *out_layouts = ff_all_channel_layouts();
69 ff_formats_ref(in_formats, &inlink->out_formats);
70 ff_formats_ref(out_formats, &outlink->in_formats);
72 ff_formats_ref(in_samplerates, &inlink->out_samplerates);
73 ff_formats_ref(out_samplerates, &outlink->in_samplerates);
75 ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
76 ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
81 static int config_output(AVFilterLink *outlink)
83 AVFilterContext *ctx = outlink->src;
84 AVFilterLink *inlink = ctx->inputs[0];
85 ResampleContext *s = ctx->priv;
86 char buf1[64], buf2[64];
90 avresample_close(s->avr);
91 avresample_free(&s->avr);
94 if (inlink->channel_layout == outlink->channel_layout &&
95 inlink->sample_rate == outlink->sample_rate &&
96 inlink->format == outlink->format)
99 if (!(s->avr = avresample_alloc_context()))
100 return AVERROR(ENOMEM);
102 av_opt_set_int(s->avr, "in_channel_layout", inlink ->channel_layout, 0);
103 av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
104 av_opt_set_int(s->avr, "in_sample_fmt", inlink ->format, 0);
105 av_opt_set_int(s->avr, "out_sample_fmt", outlink->format, 0);
106 av_opt_set_int(s->avr, "in_sample_rate", inlink ->sample_rate, 0);
107 av_opt_set_int(s->avr, "out_sample_rate", outlink->sample_rate, 0);
109 if ((ret = avresample_open(s->avr)) < 0)
112 outlink->time_base = (AVRational){ 1, outlink->sample_rate };
113 s->next_pts = AV_NOPTS_VALUE;
115 av_get_channel_layout_string(buf1, sizeof(buf1),
116 -1, inlink ->channel_layout);
117 av_get_channel_layout_string(buf2, sizeof(buf2),
118 -1, outlink->channel_layout);
119 av_log(ctx, AV_LOG_VERBOSE,
120 "fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
121 av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
122 av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
127 static int request_frame(AVFilterLink *outlink)
129 AVFilterContext *ctx = outlink->src;
130 ResampleContext *s = ctx->priv;
134 while (ret >= 0 && !s->got_output)
135 ret = ff_request_frame(ctx->inputs[0]);
137 /* flush the lavr delay buffer */
138 if (ret == AVERROR_EOF && s->avr) {
139 AVFilterBufferRef *buf;
140 int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
141 outlink->sample_rate,
142 ctx->inputs[0]->sample_rate,
148 buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
150 return AVERROR(ENOMEM);
152 ret = avresample_convert(s->avr, (void**)buf->extended_data,
153 buf->linesize[0], nb_samples,
156 avfilter_unref_buffer(buf);
157 return (ret == 0) ? AVERROR_EOF : ret;
160 buf->pts = s->next_pts;
161 return ff_filter_samples(outlink, buf);
166 static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
168 AVFilterContext *ctx = inlink->dst;
169 ResampleContext *s = ctx->priv;
170 AVFilterLink *outlink = ctx->outputs[0];
174 AVFilterBufferRef *buf_out;
175 int delay, nb_samples;
177 /* maximum possible samples lavr can output */
178 delay = avresample_get_delay(s->avr);
179 nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,
180 outlink->sample_rate, inlink->sample_rate,
183 buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
185 ret = AVERROR(ENOMEM);
189 ret = avresample_convert(s->avr, (void**)buf_out->extended_data,
190 buf_out->linesize[0], nb_samples,
191 (void**)buf->extended_data, buf->linesize[0],
192 buf->audio->nb_samples);
194 avfilter_unref_buffer(buf_out);
198 av_assert0(!avresample_available(s->avr));
200 if (s->next_pts == AV_NOPTS_VALUE) {
201 if (buf->pts == AV_NOPTS_VALUE) {
202 av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
206 s->next_pts = av_rescale_q(buf->pts, inlink->time_base,
211 buf_out->audio->nb_samples = ret;
212 if (buf->pts != AV_NOPTS_VALUE) {
213 buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,
214 outlink->time_base) -
215 av_rescale(delay, outlink->sample_rate,
216 inlink->sample_rate);
218 buf_out->pts = s->next_pts;
220 s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
222 ret = ff_filter_samples(outlink, buf_out);
227 avfilter_unref_buffer(buf);
229 ret = ff_filter_samples(outlink, buf);
236 AVFilter avfilter_af_resample = {
238 .description = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
239 .priv_size = sizeof(ResampleContext),
242 .query_formats = query_formats,
244 .inputs = (const AVFilterPad[]) {{ .name = "default",
245 .type = AVMEDIA_TYPE_AUDIO,
246 .filter_samples = filter_samples,
247 .min_perms = AV_PERM_READ },
249 .outputs = (const AVFilterPad[]) {{ .name = "default",
250 .type = AVMEDIA_TYPE_AUDIO,
251 .config_props = config_output,
252 .request_frame = request_frame },