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 
ff_audio_interleave_close(AVFormatContext * s)29 void 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 
ff_audio_interleave_init(AVFormatContext * s,const int * samples_per_frame,AVRational time_base)41 int 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 
interleave_new_audio_packet(AVFormatContext * s,AVPacket * pkt,int stream_index,int flush)78 static 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 
ff_audio_rechunk_interleave(AVFormatContext * s,AVPacket * out,AVPacket * pkt,int flush,int (* get_packet)(AVFormatContext *,AVPacket *,AVPacket *,int),int (* compare_ts)(AVFormatContext *,AVPacket *,AVPacket *))104 int 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, ret;
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             // rewrite pts and dts to be decoded time line position
123             pkt->pts = pkt->dts = aic->dts;
124             aic->dts += pkt->duration;
125             if ((ret = ff_interleave_add_packet(s, pkt, compare_ts)) < 0)
126                 return ret;
127         }
128         pkt = NULL;
129     }
130 
131     for (i = 0; i < s->nb_streams; i++) {
132         AVStream *st = s->streams[i];
133         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
134             AVPacket new_pkt;
135             while ((ret = interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) {
136                 if ((ret = ff_interleave_add_packet(s, &new_pkt, compare_ts)) < 0)
137                     return ret;
138             }
139             if (ret < 0)
140                 return ret;
141         }
142     }
143 
144     return get_packet(s, out, NULL, flush);
145 }
146