1/*
2 * Audio Interleaving functions
3 *
4 * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
5 *
6 * This file is part of FFmpeg.
7 *
8 * FFmpeg is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU Lesser General Public
10 * License as published by the Free Software Foundation; either
11 * version 2.1 of the License, or (at your option) any later version.
12 *
13 * FFmpeg is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
16 * Lesser General Public License for more details.
17 *
18 * You should have received a copy of the GNU Lesser General Public
19 * License along with FFmpeg; if not, write to the Free Software
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 */
22
23#include "libavutil/fifo.h"
24#include "libavutil/mathematics.h"
25#include "avformat.h"
26#include "audiointerleave.h"
27#include "internal.h"
28
29void ff_audio_interleave_close(AVFormatContext *s)
30{
31    int i;
32    for (i = 0; i < s->nb_streams; i++) {
33        AVStream *st = s->streams[i];
34        AudioInterleaveContext *aic = st->priv_data;
35
36        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
37            av_fifo_freep(&aic->fifo);
38    }
39}
40
41int ff_audio_interleave_init(AVFormatContext *s,
42                             const int *samples_per_frame,
43                             AVRational time_base)
44{
45    int i;
46
47    if (!samples_per_frame)
48        return AVERROR(EINVAL);
49
50    if (!time_base.num) {
51        av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n");
52        return AVERROR(EINVAL);
53    }
54    for (i = 0; i < s->nb_streams; i++) {
55        AVStream *st = s->streams[i];
56        AudioInterleaveContext *aic = st->priv_data;
57
58        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
59            aic->sample_size = (st->codec->channels *
60                                av_get_bits_per_sample(st->codec->codec_id)) / 8;
61            if (!aic->sample_size) {
62                av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
63                return AVERROR(EINVAL);
64            }
65            aic->samples_per_frame = samples_per_frame;
66            aic->samples = aic->samples_per_frame;
67            aic->time_base = time_base;
68
69            aic->fifo_size = 100* *aic->samples;
70            if (!(aic->fifo= av_fifo_alloc_array(100, *aic->samples)))
71                return AVERROR(ENOMEM);
72        }
73    }
74
75    return 0;
76}
77
78static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
79                                       int stream_index, int flush)
80{
81    AVStream *st = s->streams[stream_index];
82    AudioInterleaveContext *aic = st->priv_data;
83
84    int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size);
85    if (!size || (!flush && size == av_fifo_size(aic->fifo)))
86        return 0;
87
88    if (av_new_packet(pkt, size) < 0)
89        return AVERROR(ENOMEM);
90    av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
91
92    pkt->dts = pkt->pts = aic->dts;
93    pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
94    pkt->stream_index = stream_index;
95    aic->dts += pkt->duration;
96
97    aic->samples++;
98    if (!*aic->samples)
99        aic->samples = aic->samples_per_frame;
100
101    return size;
102}
103
104int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
105                        int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
106                        int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
107{
108    int i;
109
110    if (pkt) {
111        AVStream *st = s->streams[pkt->stream_index];
112        AudioInterleaveContext *aic = st->priv_data;
113        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
114            unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
115            if (new_size > aic->fifo_size) {
116                if (av_fifo_realloc2(aic->fifo, new_size) < 0)
117                    return AVERROR(ENOMEM);
118                aic->fifo_size = new_size;
119            }
120            av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
121        } else {
122            int ret;
123            // rewrite pts and dts to be decoded time line position
124            pkt->pts = pkt->dts = aic->dts;
125            aic->dts += pkt->duration;
126            ret = ff_interleave_add_packet(s, pkt, compare_ts);
127            if (ret < 0)
128                return ret;
129        }
130        pkt = NULL;
131    }
132
133    for (i = 0; i < s->nb_streams; i++) {
134        AVStream *st = s->streams[i];
135        if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
136            AVPacket new_pkt;
137            int ret;
138            while ((ret = interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) {
139                ret = ff_interleave_add_packet(s, &new_pkt, compare_ts);
140                if (ret < 0)
141                    return ret;
142            }
143            if (ret < 0)
144                return ret;
145        }
146    }
147
148    return get_packet(s, out, NULL, flush);
149}
150