blob: 6d4954befed710b3d7a5f072b589d8d7235e78b4
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->codecpar->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->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { |
59 | aic->sample_size = (st->codecpar->channels * |
60 | av_get_bits_per_sample(st->codecpar->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 | int ret; |
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 | ret = av_new_packet(pkt, size); |
89 | if (ret < 0) |
90 | return ret; |
91 | av_fifo_generic_read(aic->fifo, pkt->data, size, NULL); |
92 | |
93 | pkt->dts = pkt->pts = aic->dts; |
94 | pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base); |
95 | pkt->stream_index = stream_index; |
96 | aic->dts += pkt->duration; |
97 | |
98 | aic->samples++; |
99 | if (!*aic->samples) |
100 | aic->samples = aic->samples_per_frame; |
101 | |
102 | return size; |
103 | } |
104 | |
105 | int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush, |
106 | int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int), |
107 | int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *)) |
108 | { |
109 | int i, ret; |
110 | |
111 | if (pkt) { |
112 | AVStream *st = s->streams[pkt->stream_index]; |
113 | AudioInterleaveContext *aic = st->priv_data; |
114 | if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { |
115 | unsigned new_size = av_fifo_size(aic->fifo) + pkt->size; |
116 | if (new_size > aic->fifo_size) { |
117 | if (av_fifo_realloc2(aic->fifo, new_size) < 0) |
118 | return AVERROR(ENOMEM); |
119 | aic->fifo_size = new_size; |
120 | } |
121 | av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL); |
122 | } else { |
123 | // rewrite pts and dts to be decoded time line position |
124 | pkt->pts = pkt->dts = aic->dts; |
125 | aic->dts += pkt->duration; |
126 | if ((ret = ff_interleave_add_packet(s, pkt, compare_ts)) < 0) |
127 | return ret; |
128 | } |
129 | pkt = NULL; |
130 | } |
131 | |
132 | for (i = 0; i < s->nb_streams; i++) { |
133 | AVStream *st = s->streams[i]; |
134 | if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { |
135 | AVPacket new_pkt = { 0 }; |
136 | while ((ret = interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) { |
137 | if ((ret = ff_interleave_add_packet(s, &new_pkt, compare_ts)) < 0) |
138 | return ret; |
139 | } |
140 | if (ret < 0) |
141 | return ret; |
142 | } |
143 | } |
144 | |
145 | return get_packet(s, out, NULL, flush); |
146 | } |
147 |