2 * This file is part of FFmpeg.
4 * FFmpeg is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Lesser General Public
6 * License as published by the Free Software Foundation; either
7 * version 2.1 of the License, or (at your option) any later version.
9 * FFmpeg is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Lesser General Public License for more details.
14 * You should have received a copy of the GNU Lesser General Public
15 * License along with FFmpeg; if not, write to the Free Software
16 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 * sample format and channel layout conversion audio filter
24 #include "libavutil/avassert.h"
25 #include "libavutil/avstring.h"
26 #include "libavutil/common.h"
27 #include "libavutil/dict.h"
28 #include "libavutil/mathematics.h"
29 #include "libavutil/opt.h"
31 #include "libavresample/avresample.h"
38 typedef struct ResampleContext
{
40 AVAudioResampleContext
*avr
;
41 AVDictionary
*options
;
46 /* set by filter_frame() to signal an output frame to request_frame() */
50 static av_cold
int init(AVFilterContext
*ctx
, AVDictionary
**opts
)
52 ResampleContext
*s
= ctx
->priv
;
53 const AVClass
*avr_class
= avresample_get_class();
54 AVDictionaryEntry
*e
= NULL
;
56 while ((e
= av_dict_get(*opts
, "", e
, AV_DICT_IGNORE_SUFFIX
))) {
57 if (av_opt_find(&avr_class
, e
->key
, NULL
, 0,
58 AV_OPT_SEARCH_FAKE_OBJ
| AV_OPT_SEARCH_CHILDREN
))
59 av_dict_set(&s
->options
, e
->key
, e
->value
, 0);
63 while ((e
= av_dict_get(s
->options
, "", e
, AV_DICT_IGNORE_SUFFIX
)))
64 av_dict_set(opts
, e
->key
, NULL
, 0);
66 /* do not allow the user to override basic format options */
67 av_dict_set(&s
->options
, "in_channel_layout", NULL
, 0);
68 av_dict_set(&s
->options
, "out_channel_layout", NULL
, 0);
69 av_dict_set(&s
->options
, "in_sample_fmt", NULL
, 0);
70 av_dict_set(&s
->options
, "out_sample_fmt", NULL
, 0);
71 av_dict_set(&s
->options
, "in_sample_rate", NULL
, 0);
72 av_dict_set(&s
->options
, "out_sample_rate", NULL
, 0);
77 static av_cold
void uninit(AVFilterContext
*ctx
)
79 ResampleContext
*s
= ctx
->priv
;
82 avresample_close(s
->avr
);
83 avresample_free(&s
->avr
);
85 av_dict_free(&s
->options
);
88 static int query_formats(AVFilterContext
*ctx
)
90 AVFilterLink
*inlink
= ctx
->inputs
[0];
91 AVFilterLink
*outlink
= ctx
->outputs
[0];
93 AVFilterFormats
*in_formats
= ff_all_formats(AVMEDIA_TYPE_AUDIO
);
94 AVFilterFormats
*out_formats
= ff_all_formats(AVMEDIA_TYPE_AUDIO
);
95 AVFilterFormats
*in_samplerates
= ff_all_samplerates();
96 AVFilterFormats
*out_samplerates
= ff_all_samplerates();
97 AVFilterChannelLayouts
*in_layouts
= ff_all_channel_layouts();
98 AVFilterChannelLayouts
*out_layouts
= ff_all_channel_layouts();
100 ff_formats_ref(in_formats
, &inlink
->out_formats
);
101 ff_formats_ref(out_formats
, &outlink
->in_formats
);
103 ff_formats_ref(in_samplerates
, &inlink
->out_samplerates
);
104 ff_formats_ref(out_samplerates
, &outlink
->in_samplerates
);
106 ff_channel_layouts_ref(in_layouts
, &inlink
->out_channel_layouts
);
107 ff_channel_layouts_ref(out_layouts
, &outlink
->in_channel_layouts
);
112 static int config_output(AVFilterLink
*outlink
)
114 AVFilterContext
*ctx
= outlink
->src
;
115 AVFilterLink
*inlink
= ctx
->inputs
[0];
116 ResampleContext
*s
= ctx
->priv
;
117 char buf1
[64], buf2
[64];
121 avresample_close(s
->avr
);
122 avresample_free(&s
->avr
);
125 if (inlink
->channel_layout
== outlink
->channel_layout
&&
126 inlink
->sample_rate
== outlink
->sample_rate
&&
127 (inlink
->format
== outlink
->format
||
128 (av_get_channel_layout_nb_channels(inlink
->channel_layout
) == 1 &&
129 av_get_channel_layout_nb_channels(outlink
->channel_layout
) == 1 &&
130 av_get_planar_sample_fmt(inlink
->format
) ==
131 av_get_planar_sample_fmt(outlink
->format
))))
134 if (!(s
->avr
= avresample_alloc_context()))
135 return AVERROR(ENOMEM
);
138 AVDictionaryEntry
*e
= NULL
;
139 while ((e
= av_dict_get(s
->options
, "", e
, AV_DICT_IGNORE_SUFFIX
)))
140 av_log(ctx
, AV_LOG_VERBOSE
, "lavr option: %s=%s\n", e
->key
, e
->value
);
142 av_opt_set_dict(s
->avr
, &s
->options
);
145 av_opt_set_int(s
->avr
, "in_channel_layout", inlink
->channel_layout
, 0);
146 av_opt_set_int(s
->avr
, "out_channel_layout", outlink
->channel_layout
, 0);
147 av_opt_set_int(s
->avr
, "in_sample_fmt", inlink
->format
, 0);
148 av_opt_set_int(s
->avr
, "out_sample_fmt", outlink
->format
, 0);
149 av_opt_set_int(s
->avr
, "in_sample_rate", inlink
->sample_rate
, 0);
150 av_opt_set_int(s
->avr
, "out_sample_rate", outlink
->sample_rate
, 0);
152 if ((ret
= avresample_open(s
->avr
)) < 0)
155 outlink
->time_base
= (AVRational
){ 1, outlink
->sample_rate
};
156 s
->next_pts
= AV_NOPTS_VALUE
;
157 s
->next_in_pts
= AV_NOPTS_VALUE
;
159 av_get_channel_layout_string(buf1
, sizeof(buf1
),
160 -1, inlink
->channel_layout
);
161 av_get_channel_layout_string(buf2
, sizeof(buf2
),
162 -1, outlink
->channel_layout
);
163 av_log(ctx
, AV_LOG_VERBOSE
,
164 "fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
165 av_get_sample_fmt_name(inlink
->format
), inlink
->sample_rate
, buf1
,
166 av_get_sample_fmt_name(outlink
->format
), outlink
->sample_rate
, buf2
);
171 static int request_frame(AVFilterLink
*outlink
)
173 AVFilterContext
*ctx
= outlink
->src
;
174 ResampleContext
*s
= ctx
->priv
;
178 while (ret
>= 0 && !s
->got_output
)
179 ret
= ff_request_frame(ctx
->inputs
[0]);
181 /* flush the lavr delay buffer */
182 if (ret
== AVERROR_EOF
&& s
->avr
) {
184 int nb_samples
= avresample_get_out_samples(s
->avr
, 0);
189 frame
= ff_get_audio_buffer(outlink
, nb_samples
);
191 return AVERROR(ENOMEM
);
193 ret
= avresample_convert(s
->avr
, frame
->extended_data
,
194 frame
->linesize
[0], nb_samples
,
197 av_frame_free(&frame
);
198 return (ret
== 0) ? AVERROR_EOF
: ret
;
201 frame
->pts
= s
->next_pts
;
202 return ff_filter_frame(outlink
, frame
);
207 static int filter_frame(AVFilterLink
*inlink
, AVFrame
*in
)
209 AVFilterContext
*ctx
= inlink
->dst
;
210 ResampleContext
*s
= ctx
->priv
;
211 AVFilterLink
*outlink
= ctx
->outputs
[0];
216 int delay
, nb_samples
;
218 /* maximum possible samples lavr can output */
219 delay
= avresample_get_delay(s
->avr
);
220 nb_samples
= avresample_get_out_samples(s
->avr
, in
->nb_samples
);
222 out
= ff_get_audio_buffer(outlink
, nb_samples
);
224 ret
= AVERROR(ENOMEM
);
228 ret
= avresample_convert(s
->avr
, out
->extended_data
, out
->linesize
[0],
229 nb_samples
, in
->extended_data
, in
->linesize
[0],
237 av_assert0(!avresample_available(s
->avr
));
239 if (s
->next_pts
== AV_NOPTS_VALUE
) {
240 if (in
->pts
== AV_NOPTS_VALUE
) {
241 av_log(ctx
, AV_LOG_WARNING
, "First timestamp is missing, "
245 s
->next_pts
= av_rescale_q(in
->pts
, inlink
->time_base
,
250 out
->nb_samples
= ret
;
252 ret
= av_frame_copy_props(out
, in
);
258 out
->sample_rate
= outlink
->sample_rate
;
259 /* Only convert in->pts if there is a discontinuous jump.
260 This ensures that out->pts tracks the number of samples actually
261 output by the resampler in the absence of such a jump.
262 Otherwise, the rounding in av_rescale_q() and av_rescale()
263 causes off-by-1 errors. */
264 if (in
->pts
!= AV_NOPTS_VALUE
&& in
->pts
!= s
->next_in_pts
) {
265 out
->pts
= av_rescale_q(in
->pts
, inlink
->time_base
,
266 outlink
->time_base
) -
267 av_rescale(delay
, outlink
->sample_rate
,
268 inlink
->sample_rate
);
270 out
->pts
= s
->next_pts
;
272 s
->next_pts
= out
->pts
+ out
->nb_samples
;
273 s
->next_in_pts
= in
->pts
+ in
->nb_samples
;
275 ret
= ff_filter_frame(outlink
, out
);
282 in
->format
= outlink
->format
;
283 ret
= ff_filter_frame(outlink
, in
);
290 static const AVClass
*resample_child_class_next(const AVClass
*prev
)
292 return prev
? NULL
: avresample_get_class();
295 static void *resample_child_next(void *obj
, void *prev
)
297 ResampleContext
*s
= obj
;
298 return prev
? NULL
: s
->avr
;
301 static const AVClass resample_class
= {
302 .class_name
= "resample",
303 .item_name
= av_default_item_name
,
304 .version
= LIBAVUTIL_VERSION_INT
,
305 .child_class_next
= resample_child_class_next
,
306 .child_next
= resample_child_next
,
309 static const AVFilterPad avfilter_af_resample_inputs
[] = {
312 .type
= AVMEDIA_TYPE_AUDIO
,
313 .filter_frame
= filter_frame
,
318 static const AVFilterPad avfilter_af_resample_outputs
[] = {
321 .type
= AVMEDIA_TYPE_AUDIO
,
322 .config_props
= config_output
,
323 .request_frame
= request_frame
328 AVFilter ff_af_resample
= {
330 .description
= NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
331 .priv_size
= sizeof(ResampleContext
),
332 .priv_class
= &resample_class
,
335 .query_formats
= query_formats
,
336 .inputs
= avfilter_af_resample_inputs
,
337 .outputs
= avfilter_af_resample_outputs
,