1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
|
/*
* Copyright (c) 2012 Andrey Utkin
* Copyright (c) 2012 Stefano Sabatini
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Filter that changes number of samples on single output operation
*/
#include "libavutil/audio_fifo.h"
#include "libavutil/audioconvert.h"
#include "libavutil/avassert.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "audio.h"
#include "internal.h"
#include "formats.h"
typedef struct {
const AVClass *class;
int nb_out_samples; ///< how many samples to output
AVAudioFifo *fifo; ///< samples are queued here
int64_t next_out_pts;
int req_fullfilled;
int pad;
} ASNSContext;
#define OFFSET(x) offsetof(ASNSContext, x)
static const AVOption asetnsamples_options[] = {
{ "pad", "pad last frame with zeros", OFFSET(pad), AV_OPT_TYPE_INT, {.dbl=1}, 0, 1 },
{ "p", "pad last frame with zeros", OFFSET(pad), AV_OPT_TYPE_INT, {.dbl=1}, 0, 1 },
{ "nb_out_samples", "set the number of per-frame output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.dbl=1024}, 1, INT_MAX },
{ "n", "set the number of per-frame output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.dbl=1024}, 1, INT_MAX },
{ NULL }
};
AVFILTER_DEFINE_CLASS(asetnsamples);
static av_cold int init(AVFilterContext *ctx, const char *args)
{
ASNSContext *asns = ctx->priv;
int err;
asns->class = &asetnsamples_class;
av_opt_set_defaults(asns);
if ((err = av_set_options_string(asns, args, "=", ":")) < 0) {
av_log(ctx, AV_LOG_ERROR, "Error parsing options string: '%s'\n", args);
return err;
}
asns->next_out_pts = AV_NOPTS_VALUE;
av_log(ctx, AV_LOG_INFO, "nb_out_samples:%d pad:%d\n", asns->nb_out_samples, asns->pad);
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
ASNSContext *asns = ctx->priv;
av_audio_fifo_free(asns->fifo);
}
static int config_props_output(AVFilterLink *outlink)
{
ASNSContext *asns = outlink->src->priv;
int nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout);
asns->fifo = av_audio_fifo_alloc(outlink->format, nb_channels, asns->nb_out_samples);
if (!asns->fifo)
return AVERROR(ENOMEM);
return 0;
}
static int push_samples(AVFilterLink *outlink)
{
ASNSContext *asns = outlink->src->priv;
AVFilterBufferRef *outsamples = NULL;
int nb_out_samples, nb_pad_samples;
if (asns->pad) {
nb_out_samples = av_audio_fifo_size(asns->fifo) ? asns->nb_out_samples : 0;
nb_pad_samples = nb_out_samples - FFMIN(nb_out_samples, av_audio_fifo_size(asns->fifo));
} else {
nb_out_samples = FFMIN(asns->nb_out_samples, av_audio_fifo_size(asns->fifo));
nb_pad_samples = 0;
}
if (!nb_out_samples)
return 0;
outsamples = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_out_samples);
av_assert0(outsamples);
av_audio_fifo_read(asns->fifo,
(void **)outsamples->extended_data, nb_out_samples);
if (nb_pad_samples)
av_samples_set_silence(outsamples->extended_data, nb_out_samples - nb_pad_samples,
nb_pad_samples, av_get_channel_layout_nb_channels(outlink->channel_layout),
outlink->format);
outsamples->audio->nb_samples = nb_out_samples;
outsamples->audio->channel_layout = outlink->channel_layout;
outsamples->audio->sample_rate = outlink->sample_rate;
outsamples->pts = asns->next_out_pts;
if (asns->next_out_pts != AV_NOPTS_VALUE)
asns->next_out_pts += nb_out_samples;
ff_filter_samples(outlink, outsamples);
asns->req_fullfilled = 1;
return nb_out_samples;
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
ASNSContext *asns = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
int ret;
int nb_samples = insamples->audio->nb_samples;
if (av_audio_fifo_space(asns->fifo) < nb_samples) {
av_log(ctx, AV_LOG_DEBUG, "No space for %d samples, stretching audio fifo\n", nb_samples);
ret = av_audio_fifo_realloc(asns->fifo, av_audio_fifo_size(asns->fifo) + nb_samples);
if (ret < 0) {
av_log(ctx, AV_LOG_ERROR,
"Stretching audio fifo failed, discarded %d samples\n", nb_samples);
return -1;
}
}
av_audio_fifo_write(asns->fifo, (void **)insamples->extended_data, nb_samples);
if (asns->next_out_pts == AV_NOPTS_VALUE)
asns->next_out_pts = insamples->pts;
avfilter_unref_buffer(insamples);
if (av_audio_fifo_size(asns->fifo) >= asns->nb_out_samples)
push_samples(outlink);
return 0;
}
static int request_frame(AVFilterLink *outlink)
{
ASNSContext *asns = outlink->src->priv;
AVFilterLink *inlink = outlink->src->inputs[0];
int ret;
asns->req_fullfilled = 0;
do {
ret = ff_request_frame(inlink);
} while (!asns->req_fullfilled && ret >= 0);
if (ret == AVERROR_EOF)
while (push_samples(outlink))
;
return ret;
}
AVFilter avfilter_af_asetnsamples = {
.name = "asetnsamples",
.description = NULL_IF_CONFIG_SMALL("Set the number of samples for each output audio frames."),
.priv_size = sizeof(ASNSContext),
.init = init,
.uninit = uninit,
.inputs = (const AVFilterPad[]) {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ|AV_PERM_WRITE
},
{ .name = NULL }
},
.outputs = (const AVFilterPad[]) {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.request_frame = request_frame,
.config_props = config_props_output,
},
{ .name = NULL }
},
};
|
