Mercurial > libavformat.hg
annotate au.c @ 4783:fda46f120854 libavformat
allocate tracks instead of using MAX_STREAMS
author | bcoudurier |
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date | Sun, 22 Mar 2009 03:52:55 +0000 |
parents | 77e0c7511d41 |
children | 33a244b7ca65 |
rev | line source |
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885 | 1 /* |
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replace coder/decoder file description in libavformat by muxer/demuxer
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2 * AU muxer and demuxer |
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cosmetics: Remove pointless period after copyright statement non-sentences.
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3 * Copyright (c) 2001 Fabrice Bellard |
0 | 4 * |
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Change license headers to say 'FFmpeg' instead of 'this program/this library'
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5 * This file is part of FFmpeg. |
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Change license headers to say 'FFmpeg' instead of 'this program/this library'
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6 * |
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7 * FFmpeg is free software; you can redistribute it and/or |
0 | 8 * modify it under the terms of the GNU Lesser General Public |
9 * License as published by the Free Software Foundation; either | |
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Change license headers to say 'FFmpeg' instead of 'this program/this library'
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10 * version 2.1 of the License, or (at your option) any later version. |
0 | 11 * |
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12 * FFmpeg is distributed in the hope that it will be useful, |
0 | 13 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
15 * Lesser General Public License for more details. | |
16 * | |
17 * You should have received a copy of the GNU Lesser General Public | |
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Change license headers to say 'FFmpeg' instead of 'this program/this library'
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18 * License along with FFmpeg; if not, write to the Free Software |
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Update licensing information: The FSF changed postal address.
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
0 | 20 */ |
21 | |
22 /* | |
23 * First version by Francois Revol revol@free.fr | |
24 * | |
25 * Reference documents: | |
26 * http://www.opengroup.org/public/pubs/external/auformat.html | |
27 * http://www.goice.co.jp/member/mo/formats/au.html | |
28 */ | |
29 | |
30 #include "avformat.h" | |
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move unrelated functions declarations out of allformats.h
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31 #include "raw.h" |
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move common stuff from avienc.c and wav.c to new file riff.c
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32 #include "riff.h" |
0 | 33 |
34 /* if we don't know the size in advance */ | |
1885 | 35 #define AU_UNKNOWN_SIZE ((uint32_t)(~0)) |
0 | 36 |
37 /* The ffmpeg codecs we support, and the IDs they have in the file */ | |
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add codec_id <-> codec_tag tables to AVIn/OutputFormat
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38 static const AVCodecTag codec_au_tags[] = { |
0 | 39 { CODEC_ID_PCM_MULAW, 1 }, |
2933 | 40 { CODEC_ID_PCM_S8, 2 }, |
0 | 41 { CODEC_ID_PCM_S16BE, 3 }, |
3736
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Make AU demuxer handle S24BE, S32BE and F64BE PCM audio.
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42 { CODEC_ID_PCM_S24BE, 4 }, |
441e758e1a5e
Make AU demuxer handle S24BE, S32BE and F64BE PCM audio.
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43 { CODEC_ID_PCM_S32BE, 5 }, |
3599
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Support 32-bit floating point audio samples in Sun AU demuxer
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44 { CODEC_ID_PCM_F32BE, 6 }, |
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45 { CODEC_ID_PCM_F64BE, 7 }, |
0 | 46 { CODEC_ID_PCM_ALAW, 27 }, |
47 { 0, 0 }, | |
48 }; | |
49 | |
4206 | 50 #if CONFIG_AU_MUXER |
0 | 51 /* AUDIO_FILE header */ |
52 static int put_au_header(ByteIOContext *pb, AVCodecContext *enc) | |
53 { | |
196 | 54 if(!enc->codec_tag) |
0 | 55 return -1; |
56 put_tag(pb, ".snd"); /* magic number */ | |
57 put_be32(pb, 24); /* header size */ | |
1885 | 58 put_be32(pb, AU_UNKNOWN_SIZE); /* data size */ |
196 | 59 put_be32(pb, (uint32_t)enc->codec_tag); /* codec ID */ |
0 | 60 put_be32(pb, enc->sample_rate); |
65 | 61 put_be32(pb, (uint32_t)enc->channels); |
0 | 62 return 0; |
63 } | |
64 | |
65 static int au_write_header(AVFormatContext *s) | |
66 { | |
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Use dynamically allocated ByteIOContext in AVFormatContext
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67 ByteIOContext *pb = s->pb; |
0 | 68 |
69 s->priv_data = NULL; | |
70 | |
71 /* format header */ | |
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changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
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72 if (put_au_header(pb, s->streams[0]->codec) < 0) { |
0 | 73 return -1; |
74 } | |
75 | |
76 put_flush_packet(pb); | |
77 | |
78 return 0; | |
79 } | |
80 | |
468 | 81 static int au_write_packet(AVFormatContext *s, AVPacket *pkt) |
0 | 82 { |
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83 ByteIOContext *pb = s->pb; |
468 | 84 put_buffer(pb, pkt->data, pkt->size); |
0 | 85 return 0; |
86 } | |
87 | |
88 static int au_write_trailer(AVFormatContext *s) | |
89 { | |
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90 ByteIOContext *pb = s->pb; |
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Remove offset_t typedef and use int64_t directly instead.
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91 int64_t file_size; |
0 | 92 |
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93 if (!url_is_streamed(s->pb)) { |
0 | 94 |
95 /* update file size */ | |
96 file_size = url_ftell(pb); | |
97 url_fseek(pb, 8, SEEK_SET); | |
65 | 98 put_be32(pb, (uint32_t)(file_size - 24)); |
0 | 99 url_fseek(pb, file_size, SEEK_SET); |
100 | |
101 put_flush_packet(pb); | |
102 } | |
103 | |
104 return 0; | |
105 } | |
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Replace generic CONFIG_MUXERS preprocessor conditionals by more specific
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106 #endif /* CONFIG_AU_MUXER */ |
0 | 107 |
108 static int au_probe(AVProbeData *p) | |
109 { | |
110 /* check file header */ | |
111 if (p->buf[0] == '.' && p->buf[1] == 's' && | |
112 p->buf[2] == 'n' && p->buf[3] == 'd') | |
113 return AVPROBE_SCORE_MAX; | |
114 else | |
115 return 0; | |
116 } | |
117 | |
118 /* au input */ | |
119 static int au_read_header(AVFormatContext *s, | |
306 | 120 AVFormatParameters *ap) |
0 | 121 { |
122 int size; | |
123 unsigned int tag; | |
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124 ByteIOContext *pb = s->pb; |
3967 | 125 unsigned int id, channels, rate; |
126 enum CodecID codec; | |
0 | 127 AVStream *st; |
128 | |
129 /* check ".snd" header */ | |
130 tag = get_le32(pb); | |
131 if (tag != MKTAG('.', 's', 'n', 'd')) | |
132 return -1; | |
133 size = get_be32(pb); /* header size */ | |
134 get_be32(pb); /* data size */ | |
885 | 135 |
0 | 136 id = get_be32(pb); |
137 rate = get_be32(pb); | |
138 channels = get_be32(pb); | |
885 | 139 |
0 | 140 codec = codec_get_id(codec_au_tags, id); |
141 | |
142 if (size >= 24) { | |
143 /* skip unused data */ | |
144 url_fseek(pb, size - 24, SEEK_CUR); | |
145 } | |
146 | |
147 /* now we are ready: build format streams */ | |
187 | 148 st = av_new_stream(s, 0); |
0 | 149 if (!st) |
150 return -1; | |
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151 st->codec->codec_type = CODEC_TYPE_AUDIO; |
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152 st->codec->codec_tag = id; |
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153 st->codec->codec_id = codec; |
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changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility
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154 st->codec->channels = channels; |
feca73904e67
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155 st->codec->sample_rate = rate; |
567 | 156 av_set_pts_info(st, 64, 1, rate); |
0 | 157 return 0; |
158 } | |
159 | |
160 #define MAX_SIZE 4096 | |
161 | |
162 static int au_read_packet(AVFormatContext *s, | |
163 AVPacket *pkt) | |
164 { | |
165 int ret; | |
166 | |
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167 if (url_feof(s->pb)) |
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Replace all occurrences of AVERROR_IO with AVERROR(EIO).
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168 return AVERROR(EIO); |
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169 ret= av_get_packet(s->pb, pkt, MAX_SIZE); |
775 | 170 if (ret < 0) |
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171 return AVERROR(EIO); |
0 | 172 pkt->stream_index = 0; |
173 | |
174 /* note: we need to modify the packet size here to handle the last | |
175 packet */ | |
176 pkt->size = ret; | |
177 return 0; | |
178 } | |
179 | |
4206 | 180 #if CONFIG_AU_DEMUXER |
1169 | 181 AVInputFormat au_demuxer = { |
0 | 182 "au", |
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183 NULL_IF_CONFIG_SMALL("SUN AU format"), |
0 | 184 0, |
185 au_probe, | |
186 au_read_header, | |
187 au_read_packet, | |
3484 | 188 NULL, |
306 | 189 pcm_read_seek, |
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190 .codec_tag= (const AVCodecTag* const []){codec_au_tags, 0}, |
0 | 191 }; |
1169 | 192 #endif |
0 | 193 |
4206 | 194 #if CONFIG_AU_MUXER |
1169 | 195 AVOutputFormat au_muxer = { |
0 | 196 "au", |
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197 NULL_IF_CONFIG_SMALL("SUN AU format"), |
0 | 198 "audio/basic", |
199 "au", | |
200 0, | |
201 CODEC_ID_PCM_S16BE, | |
202 CODEC_ID_NONE, | |
203 au_write_header, | |
204 au_write_packet, | |
205 au_write_trailer, | |
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206 .codec_tag= (const AVCodecTag* const []){codec_au_tags, 0}, |
0 | 207 }; |
1169 | 208 #endif //CONFIG_AU_MUXER |