fate-run: make avconv() properly deal with multiple inputs.
[libav.git] / libavfilter / af_resample.c
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1/*
2 *
3 * This file is part of Libav.
4 *
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.
9 *
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.
14 *
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
18 */
19
20/**
21 * @file
22 * sample format and channel layout conversion audio filter
23 */
24
25#include "libavutil/avassert.h"
26#include "libavutil/avstring.h"
27#include "libavutil/mathematics.h"
28#include "libavutil/opt.h"
29
30#include "libavresample/avresample.h"
31
32#include "audio.h"
33#include "avfilter.h"
ff1f51a8 34#include "formats.h"
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35#include "internal.h"
36
37typedef struct ResampleContext {
38 AVAudioResampleContext *avr;
39
40 int64_t next_pts;
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41
42 /* set by filter_samples() to signal an output frame to request_frame() */
43 int got_output;
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44} ResampleContext;
45
46static av_cold void uninit(AVFilterContext *ctx)
47{
48 ResampleContext *s = ctx->priv;
49
50 if (s->avr) {
51 avresample_close(s->avr);
52 avresample_free(&s->avr);
53 }
54}
55
56static int query_formats(AVFilterContext *ctx)
57{
58 AVFilterLink *inlink = ctx->inputs[0];
59 AVFilterLink *outlink = ctx->outputs[0];
60
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61 AVFilterFormats *in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
62 AVFilterFormats *out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
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63 AVFilterFormats *in_samplerates = ff_all_samplerates();
64 AVFilterFormats *out_samplerates = ff_all_samplerates();
65 AVFilterChannelLayouts *in_layouts = ff_all_channel_layouts();
66 AVFilterChannelLayouts *out_layouts = ff_all_channel_layouts();
d371e7b9 67
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68 ff_formats_ref(in_formats, &inlink->out_formats);
69 ff_formats_ref(out_formats, &outlink->in_formats);
d371e7b9 70
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71 ff_formats_ref(in_samplerates, &inlink->out_samplerates);
72 ff_formats_ref(out_samplerates, &outlink->in_samplerates);
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73
74 ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
75 ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
76
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77 return 0;
78}
79
80static int config_output(AVFilterLink *outlink)
81{
82 AVFilterContext *ctx = outlink->src;
83 AVFilterLink *inlink = ctx->inputs[0];
84 ResampleContext *s = ctx->priv;
85 char buf1[64], buf2[64];
86 int ret;
87
88 if (s->avr) {
89 avresample_close(s->avr);
90 avresample_free(&s->avr);
91 }
92
93 if (inlink->channel_layout == outlink->channel_layout &&
94 inlink->sample_rate == outlink->sample_rate &&
95 inlink->format == outlink->format)
96 return 0;
97
98 if (!(s->avr = avresample_alloc_context()))
99 return AVERROR(ENOMEM);
100
101 av_opt_set_int(s->avr, "in_channel_layout", inlink ->channel_layout, 0);
102 av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
103 av_opt_set_int(s->avr, "in_sample_fmt", inlink ->format, 0);
104 av_opt_set_int(s->avr, "out_sample_fmt", outlink->format, 0);
105 av_opt_set_int(s->avr, "in_sample_rate", inlink ->sample_rate, 0);
106 av_opt_set_int(s->avr, "out_sample_rate", outlink->sample_rate, 0);
107
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108 if ((ret = avresample_open(s->avr)) < 0)
109 return ret;
110
111 outlink->time_base = (AVRational){ 1, outlink->sample_rate };
112 s->next_pts = AV_NOPTS_VALUE;
113
114 av_get_channel_layout_string(buf1, sizeof(buf1),
115 -1, inlink ->channel_layout);
116 av_get_channel_layout_string(buf2, sizeof(buf2),
117 -1, outlink->channel_layout);
118 av_log(ctx, AV_LOG_VERBOSE,
e0d8427d 119 "fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
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120 av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
121 av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
122
123 return 0;
124}
125
126static int request_frame(AVFilterLink *outlink)
127{
128 AVFilterContext *ctx = outlink->src;
129 ResampleContext *s = ctx->priv;
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130 int ret = 0;
131
132 s->got_output = 0;
133 while (ret >= 0 && !s->got_output)
134 ret = ff_request_frame(ctx->inputs[0]);
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135
136 /* flush the lavr delay buffer */
137 if (ret == AVERROR_EOF && s->avr) {
138 AVFilterBufferRef *buf;
139 int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
140 outlink->sample_rate,
141 ctx->inputs[0]->sample_rate,
142 AV_ROUND_UP);
143
144 if (!nb_samples)
145 return ret;
146
147 buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
148 if (!buf)
149 return AVERROR(ENOMEM);
150
151 ret = avresample_convert(s->avr, (void**)buf->extended_data,
152 buf->linesize[0], nb_samples,
153 NULL, 0, 0);
154 if (ret <= 0) {
155 avfilter_unref_buffer(buf);
156 return (ret == 0) ? AVERROR_EOF : ret;
157 }
158
159 buf->pts = s->next_pts;
160 ff_filter_samples(outlink, buf);
161 return 0;
162 }
163 return ret;
164}
165
166static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
167{
168 AVFilterContext *ctx = inlink->dst;
169 ResampleContext *s = ctx->priv;
170 AVFilterLink *outlink = ctx->outputs[0];
171
172 if (s->avr) {
173 AVFilterBufferRef *buf_out;
174 int delay, nb_samples, ret;
175
176 /* maximum possible samples lavr can output */
177 delay = avresample_get_delay(s->avr);
178 nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,
179 outlink->sample_rate, inlink->sample_rate,
180 AV_ROUND_UP);
181
182 buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
183 ret = avresample_convert(s->avr, (void**)buf_out->extended_data,
184 buf_out->linesize[0], nb_samples,
185 (void**)buf->extended_data, buf->linesize[0],
186 buf->audio->nb_samples);
187
188 av_assert0(!avresample_available(s->avr));
189
190 if (s->next_pts == AV_NOPTS_VALUE) {
191 if (buf->pts == AV_NOPTS_VALUE) {
192 av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
193 "assuming 0.\n");
194 s->next_pts = 0;
195 } else
196 s->next_pts = av_rescale_q(buf->pts, inlink->time_base,
197 outlink->time_base);
198 }
199
200 if (ret > 0) {
201 buf_out->audio->nb_samples = ret;
202 if (buf->pts != AV_NOPTS_VALUE) {
203 buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,
204 outlink->time_base) -
205 av_rescale(delay, outlink->sample_rate,
206 inlink->sample_rate);
207 } else
208 buf_out->pts = s->next_pts;
209
210 s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
211
212 ff_filter_samples(outlink, buf_out);
1ffb6456 213 s->got_output = 1;
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214 }
215 avfilter_unref_buffer(buf);
1ffb6456 216 } else {
d371e7b9 217 ff_filter_samples(outlink, buf);
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218 s->got_output = 1;
219 }
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220}
221
222AVFilter avfilter_af_resample = {
223 .name = "resample",
224 .description = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
225 .priv_size = sizeof(ResampleContext),
226
227 .uninit = uninit,
228 .query_formats = query_formats,
229
230 .inputs = (const AVFilterPad[]) {{ .name = "default",
231 .type = AVMEDIA_TYPE_AUDIO,
232 .filter_samples = filter_samples,
233 .min_perms = AV_PERM_READ },
234 { .name = NULL}},
235 .outputs = (const AVFilterPad[]) {{ .name = "default",
236 .type = AVMEDIA_TYPE_AUDIO,
237 .config_props = config_output,
238 .request_frame = request_frame },
239 { .name = NULL}},
240};