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 "avformat.h"
25#include "audiointerleave.h"
26#include "internal.h"
27
28void ff_audio_interleave_close(AVFormatContext *s)
29{
30    int i;
31    for (i = 0; i < s->nb_streams; i++) {
32        AVStream *st = s->streams[i];
33        AudioInterleaveContext *aic = st->priv_data;
34
35        if (st->codec->codec_type == CODEC_TYPE_AUDIO)
36            av_fifo_free(&aic->fifo);
37    }
38}
39
40int ff_audio_interleave_init(AVFormatContext *s,
41                             const int *samples_per_frame,
42                             AVRational time_base)
43{
44    int i;
45
46    if (!samples_per_frame)
47        return -1;
48
49    for (i = 0; i < s->nb_streams; i++) {
50        AVStream *st = s->streams[i];
51        AudioInterleaveContext *aic = st->priv_data;
52
53        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
54            aic->sample_size = (st->codec->channels *
55                                av_get_bits_per_sample(st->codec->codec_id)) / 8;
56            if (!aic->sample_size) {
57                av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
58                return -1;
59            }
60            aic->samples_per_frame = samples_per_frame;
61            aic->samples = aic->samples_per_frame;
62            aic->time_base = time_base;
63
64            aic->fifo_size = 100* *aic->samples;
65            av_fifo_init(&aic->fifo, 100 * *aic->samples);
66        }
67    }
68
69    return 0;
70}
71
72static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
73                                   int stream_index, int flush)
74{
75    AVStream *st = s->streams[stream_index];
76    AudioInterleaveContext *aic = st->priv_data;
77
78    int size = FFMIN(av_fifo_size(&aic->fifo), *aic->samples * aic->sample_size);
79    if (!size || (!flush && size == av_fifo_size(&aic->fifo)))
80        return 0;
81
82    av_new_packet(pkt, size);
83    av_fifo_read(&aic->fifo, pkt->data, size);
84
85    pkt->dts = pkt->pts = aic->dts;
86    pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
87    pkt->stream_index = stream_index;
88    aic->dts += pkt->duration;
89
90    aic->samples++;
91    if (!*aic->samples)
92        aic->samples = aic->samples_per_frame;
93
94    return size;
95}
96
97int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
98                        int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
99                        int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
100{
101    int i;
102
103    if (pkt) {
104        AVStream *st = s->streams[pkt->stream_index];
105        AudioInterleaveContext *aic = st->priv_data;
106        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
107            unsigned new_size = av_fifo_size(&aic->fifo) + pkt->size;
108            if (new_size > aic->fifo_size) {
109                if (av_fifo_realloc2(&aic->fifo, new_size) < 0)
110                    return -1;
111                aic->fifo_size = new_size;
112            }
113            av_fifo_generic_write(&aic->fifo, pkt->data, pkt->size, NULL);
114        } else {
115            // rewrite pts and dts to be decoded time line position
116            pkt->pts = pkt->dts = aic->dts;
117            aic->dts += pkt->duration;
118            ff_interleave_add_packet(s, pkt, compare_ts);
119        }
120        pkt = NULL;
121    }
122
123    for (i = 0; i < s->nb_streams; i++) {
124        AVStream *st = s->streams[i];
125        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
126            AVPacket new_pkt;
127            while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush))
128                ff_interleave_add_packet(s, &new_pkt, compare_ts);
129        }
130    }
131
132    return get_packet(s, out, pkt, flush);
133}
134