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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"
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_BOOL, {.i64=1}, 0, 1, FLAGS },
51  { "p", "pad last frame with zeros", OFFSET(pad), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, FLAGS },
52  { NULL }
53 };
54 
55 AVFILTER_DEFINE_CLASS(asetnsamples);
56 
58 {
59  ASNSContext *asns = ctx->priv;
60 
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 
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 
81  return 0;
82 }
83 
84 static int push_samples(AVFilterLink *outlink)
85 {
86  ASNSContext *asns = outlink->src->priv;
87  AVFrame *outsamples = NULL;
88  int ret, nb_out_samples, nb_pad_samples;
89 
90  if (asns->pad) {
91  nb_out_samples = av_audio_fifo_size(asns->fifo) ? asns->nb_out_samples : 0;
92  nb_pad_samples = nb_out_samples - FFMIN(nb_out_samples, av_audio_fifo_size(asns->fifo));
93  } else {
94  nb_out_samples = FFMIN(asns->nb_out_samples, av_audio_fifo_size(asns->fifo));
95  nb_pad_samples = 0;
96  }
97 
98  if (!nb_out_samples)
99  return 0;
100 
101  outsamples = ff_get_audio_buffer(outlink, nb_out_samples);
102  if (!outsamples)
103  return AVERROR(ENOMEM);
104 
105  av_audio_fifo_read(asns->fifo,
106  (void **)outsamples->extended_data, nb_out_samples);
107 
108  if (nb_pad_samples)
109  av_samples_set_silence(outsamples->extended_data, nb_out_samples - nb_pad_samples,
110  nb_pad_samples, outlink->channels,
111  outlink->format);
112  outsamples->nb_samples = nb_out_samples;
113  outsamples->channel_layout = outlink->channel_layout;
114  outsamples->sample_rate = outlink->sample_rate;
115  outsamples->pts = asns->next_out_pts;
116 
117  if (asns->next_out_pts != AV_NOPTS_VALUE)
118  asns->next_out_pts += av_rescale_q(nb_out_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
119 
120  ret = ff_filter_frame(outlink, outsamples);
121  if (ret < 0)
122  return ret;
123  return nb_out_samples;
124 }
125 
126 static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
127 {
128  AVFilterContext *ctx = inlink->dst;
129  ASNSContext *asns = ctx->priv;
130  AVFilterLink *outlink = ctx->outputs[0];
131  int ret;
132  int nb_samples = insamples->nb_samples;
133 
134  if (av_audio_fifo_space(asns->fifo) < nb_samples) {
135  av_log(ctx, AV_LOG_DEBUG, "No space for %d samples, stretching audio fifo\n", nb_samples);
136  ret = av_audio_fifo_realloc(asns->fifo, av_audio_fifo_size(asns->fifo) + nb_samples);
137  if (ret < 0) {
138  av_log(ctx, AV_LOG_ERROR,
139  "Stretching audio fifo failed, discarded %d samples\n", nb_samples);
140  return -1;
141  }
142  }
143  av_audio_fifo_write(asns->fifo, (void **)insamples->extended_data, nb_samples);
144  if (asns->next_out_pts == AV_NOPTS_VALUE)
145  asns->next_out_pts = insamples->pts;
146  av_frame_free(&insamples);
147 
148  while (av_audio_fifo_size(asns->fifo) >= asns->nb_out_samples)
149  push_samples(outlink);
150  return 0;
151 }
152 
153 static int request_frame(AVFilterLink *outlink)
154 {
155  AVFilterLink *inlink = outlink->src->inputs[0];
156  int ret;
157 
158  ret = ff_request_frame(inlink);
159  if (ret == AVERROR_EOF) {
160  ret = push_samples(outlink);
161  return ret < 0 ? ret : ret > 0 ? 0 : AVERROR_EOF;
162  }
163 
164  return ret;
165 }
166 
168  {
169  .name = "default",
170  .type = AVMEDIA_TYPE_AUDIO,
171  .filter_frame = filter_frame,
172  },
173  { NULL }
174 };
175 
177  {
178  .name = "default",
179  .type = AVMEDIA_TYPE_AUDIO,
180  .request_frame = request_frame,
181  .config_props = config_props_output,
182  },
183  { NULL }
184 };
185 
187  .name = "asetnsamples",
188  .description = NULL_IF_CONFIG_SMALL("Set the number of samples for each output audio frames."),
189  .priv_size = sizeof(ASNSContext),
190  .priv_class = &asetnsamples_class,
191  .init = init,
192  .uninit = uninit,
193  .inputs = asetnsamples_inputs,
194  .outputs = asetnsamples_outputs,
195 };
#define NULL
Definition: coverity.c:32
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
Definition: audio_fifo.c:60
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
Read data from an AVAudioFifo.
Definition: audio_fifo.c:182
This structure describes decoded (raw) audio or video data.
Definition: frame.h:184
static av_cold int init(AVFilterContext *ctx)
static int config_props_output(AVFilterLink *outlink)
AVOption.
Definition: opt.h:245
static const AVOption asetnsamples_options[]
AVFilter ff_af_asetnsamples
Main libavfilter public API header.
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:45
static av_cold void uninit(AVFilterContext *ctx)
const char * name
Pad name.
Definition: internal.h:59
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:315
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1189
#define av_cold
Definition: attributes.h:82
AVOptions.
static const AVFilterPad asetnsamples_outputs[]
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:268
#define AVERROR_EOF
End of file.
Definition: error.h:55
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:192
#define av_log(a,...)
static const AVFilterPad asetnsamples_inputs[]
A filter pad used for either input or output.
Definition: internal.h:53
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:142
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
int av_audio_fifo_space(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for writing.
Definition: audio_fifo.c:234
int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt)
Fill an audio buffer with silence.
Definition: samplefmt.c:237
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:65
#define AVERROR(e)
Definition: error.h:43
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:154
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:176
void * priv
private data for use by the filter
Definition: avfilter.h:322
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
#define OFFSET(x)
simple assert() macros that are a bit more flexible than ISO C assert().
static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
#define FLAGS
Context for an Audio FIFO Buffer.
Definition: audio_fifo.c:34
uint64_t channel_layout
Channel layout of the audio data.
Definition: frame.h:349
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
Definition: audio_fifo.c:229
audio channel layout utility functions
int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples)
Reallocate an AVAudioFifo.
Definition: audio_fifo.c:97
#define FFMIN(a, b)
Definition: common.h:96
int64_t next_out_pts
AVFormatContext * ctx
Definition: movenc.c:48
static int request_frame(AVFilterLink *outlink)
static const AVFilterPad outputs[]
Definition: af_afftfilt.c:386
static const AVFilterPad inputs[]
Definition: af_afftfilt.c:376
Describe the class of an AVClass context structure.
Definition: log.h:67
int sample_rate
Sample rate of the audio data.
Definition: frame.h:344
Filter definition.
Definition: avfilter.h:144
Rational number (pair of numerator and denominator).
Definition: rational.h:58
int nb_out_samples
how many samples to output
AVAudioFifo * fifo
samples are queued here
const char * name
Filter name.
Definition: avfilter.h:148
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:319
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:113
static int push_samples(AVFilterLink *outlink)
AVFILTER_DEFINE_CLASS(asetnsamples)
Audio FIFO Buffer.
An instance of a filter.
Definition: avfilter.h:307
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
Definition: avfilter.c:369
internal API functions
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:231
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:241
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:242