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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 | ||
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 | ||
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 | ||
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 | ||
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 | } |