FFmpeg: libavfilter/af_asetnsamples.c Source File

FFmpeg
af_asetnsamples.c
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1 /*
2  * Copyright (c) 2012 Andrey Utkin
3  * Copyright (c) 2012 Stefano Sabatini
4  *
5  * This file is part of FFmpeg.
6  *
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.
11  *
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.
16  *
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
20  */
21 
22 /**
23  * @file
24  * Filter that changes number of samples on single output operation
25  */
26 
27 #include "libavutil/audio_fifo.h"
28 #include "libavutil/avassert.h"
29 #include "libavutil/channel_layout.h"
30 #include "libavutil/opt.h"
31 #include "avfilter.h"
32 #include "audio.h"
33 #include "internal.h"
34 #include "formats.h"
35 
36  typedef struct {
37   const AVClass *class;
38   int nb_out_samples; ///< how many samples to output
39   AVAudioFifo *fifo; ///< samples are queued here
40   int64_t next_out_pts;
41   int pad;
42 } ASNSContext;
43 
44  #define OFFSET(x) offsetof(ASNSContext, x)
45  #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
46 
47  static const AVOption asetnsamples_options[] = {
48  { "nb_out_samples", "set the number of per-frame output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, FLAGS },
49  { "n", "set the number of per-frame output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, FLAGS },
50  { "pad", "pad last frame with zeros", OFFSET(pad), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, FLAGS },
51  { "p", "pad last frame with zeros", OFFSET(pad), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, FLAGS },
52  { NULL }
53 };
54 
55 AVFILTER_DEFINE_CLASS(asetnsamples);
56 
57  static av_cold int init(AVFilterContext *ctx)
58 {
59  ASNSContext *asns = ctx->priv;
60 
61  asns->next_out_pts = AV_NOPTS_VALUE;
62  av_log(ctx, AV_LOG_VERBOSE, "nb_out_samples:%d pad:%d\n", asns->nb_out_samples, asns->pad);
63 
64  return 0;
65 }
66 
67  static av_cold void uninit(AVFilterContext *ctx)
68 {
69  ASNSContext *asns = ctx->priv;
70  av_audio_fifo_free(asns->fifo);
71 }
72 
73  static int config_props_output(AVFilterLink *outlink)
74 {
75  ASNSContext *asns = outlink->src->priv;
76 
77  asns->fifo = av_audio_fifo_alloc(outlink->format, outlink->channels, asns->nb_out_samples);
78  if (!asns->fifo)
79  return AVERROR(ENOMEM);
80  outlink->flags |= FF_LINK_FLAG_REQUEST_LOOP;
81 
82  return 0;
83 }
84 
85  static int push_samples(AVFilterLink *outlink)
86 {
87  ASNSContext *asns = outlink->src->priv;
88  AVFrame *outsamples = NULL;
89  int ret, nb_out_samples, nb_pad_samples;
90 
91  if (asns->pad) {
92  nb_out_samples = av_audio_fifo_size(asns->fifo) ? asns->nb_out_samples : 0;
93  nb_pad_samples = nb_out_samples - FFMIN(nb_out_samples, av_audio_fifo_size(asns->fifo));
94  } else {
95  nb_out_samples = FFMIN(asns->nb_out_samples, av_audio_fifo_size(asns->fifo));
96  nb_pad_samples = 0;
97  }
98 
99  if (!nb_out_samples)
100  return 0;
101 
102  outsamples = ff_get_audio_buffer(outlink, nb_out_samples);
103  if (!outsamples)
104  return AVERROR(ENOMEM);
105 
106  av_audio_fifo_read(asns->fifo,
107  (void **)outsamples->extended_data, nb_out_samples);
108 
109  if (nb_pad_samples)
110  av_samples_set_silence(outsamples->extended_data, nb_out_samples - nb_pad_samples,
111  nb_pad_samples, outlink->channels,
112  outlink->format);
113  outsamples->nb_samples = nb_out_samples;
114  outsamples->channel_layout = outlink->channel_layout;
115  outsamples->sample_rate = outlink->sample_rate;
116  outsamples->pts = asns->next_out_pts;
117 
118  if (asns->next_out_pts != AV_NOPTS_VALUE)
119  asns->next_out_pts += nb_out_samples;
120 
121  ret = ff_filter_frame(outlink, outsamples);
122  if (ret < 0)
123  return ret;
124  return nb_out_samples;
125 }
126 
127  static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
128 {
129  AVFilterContext *ctx = inlink->dst;
130  ASNSContext *asns = ctx->priv;
131  AVFilterLink *outlink = ctx->outputs[0];
132  int ret;
133  int nb_samples = insamples->nb_samples;
134 
135  if (av_audio_fifo_space(asns->fifo) < nb_samples) {
136  av_log(ctx, AV_LOG_DEBUG, "No space for %d samples, stretching audio fifo\n", nb_samples);
137  ret = av_audio_fifo_realloc(asns->fifo, av_audio_fifo_size(asns->fifo) + nb_samples);
138  if (ret < 0) {
139  av_log(ctx, AV_LOG_ERROR,
140  "Stretching audio fifo failed, discarded %d samples\n", nb_samples);
141  return -1;
142  }
143  }
144  av_audio_fifo_write(asns->fifo, (void **)insamples->extended_data, nb_samples);
145  if (asns->next_out_pts == AV_NOPTS_VALUE)
146  asns->next_out_pts = insamples->pts;
147  av_frame_free(&insamples);
148 
149  while (av_audio_fifo_size(asns->fifo) >= asns->nb_out_samples)
150  push_samples(outlink);
151  return 0;
152 }
153 
154  static int request_frame(AVFilterLink *outlink)
155 {
156  AVFilterLink *inlink = outlink->src->inputs[0];
157  int ret;
158 
159  ret = ff_request_frame(inlink);
160  if (ret == AVERROR_EOF) {
161  ret = push_samples(outlink);
162  return ret < 0 ? ret : ret > 0 ? 0 : AVERROR_EOF;
163  }
164 
165  return ret;
166 }
167 
168  static const AVFilterPad asetnsamples_inputs[] = {
169  {
170  .name = "default",
171  .type = AVMEDIA_TYPE_AUDIO,
172  .filter_frame = filter_frame,
173  },
174  { NULL }
175 };
176 
177  static const AVFilterPad asetnsamples_outputs[] = {
178  {
179  .name = "default",
180  .type = AVMEDIA_TYPE_AUDIO,
181  .request_frame = request_frame,
182  .config_props = config_props_output,
183  },
184  { NULL }
185 };
186 
187  AVFilter avfilter_af_asetnsamples = {
188  .name = "asetnsamples",
189  .description = NULL_IF_CONFIG_SMALL("Set the number of samples for each output audio frames."),
190  .priv_size = sizeof(ASNSContext),
191  .priv_class = &asetnsamples_class,
192  .init = init,
193  .uninit = uninit,
194  .inputs = asetnsamples_inputs,
195  .outputs = asetnsamples_outputs,
196 };
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