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
36typedef 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
47static 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
55AVFILTER_DEFINE_CLASS(asetnsamples);
56
57static 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
67static av_cold void uninit(AVFilterContext *ctx)
68{
69    ASNSContext *asns = ctx->priv;
70    av_audio_fifo_free(asns->fifo);
71}
72
73static 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
85static 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 += av_rescale_q(nb_out_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
120
121    ret = ff_filter_frame(outlink, outsamples);
122    if (ret < 0)
123        return ret;
124    return nb_out_samples;
125}
126
127static 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
154static 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
168static const AVFilterPad asetnsamples_inputs[] = {
169    {
170        .name         = "default",
171        .type         = AVMEDIA_TYPE_AUDIO,
172        .filter_frame = filter_frame,
173    },
174    { NULL }
175};
176
177static 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
187AVFilter ff_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};
197