Mercurial > libavformat.hg
annotate vocdec.c @ 5677:545f70d48943 libavformat
Make rtsp_close_streams() take a AVFormatContext instead of a RTSPState
argument, so we can use AVFormatContext->* here in the future.
Patch by Martin Storsj <$firstname $firstname st>.
author | rbultje |
---|---|
date | Fri, 19 Feb 2010 23:10:19 +0000 |
parents | 3528d151a57e |
children | 536e5527c1e0 |
rev | line source |
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937 | 1 /* |
2 * Creative Voice File demuxer. | |
3 * Copyright (c) 2006 Aurelien Jacobs <aurel@gnuage.org> | |
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. |
0899bfe4105c
Change license headers to say 'FFmpeg' instead of 'this program/this library'
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6 * |
0899bfe4105c
Change license headers to say 'FFmpeg' instead of 'this program/this library'
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7 * FFmpeg is free software; you can redistribute it and/or |
937 | 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. |
937 | 11 * |
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Change license headers to say 'FFmpeg' instead of 'this program/this library'
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12 * FFmpeg is distributed in the hope that it will be useful, |
937 | 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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18 * License along with FFmpeg; if not, write to the Free Software |
2217 | 19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
937 | 20 */ |
21 | |
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7d2f3f1b68d8
Fix build: Add intreadwrite.h and bswap.h #includes where necessary.
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22 #include "libavutil/intreadwrite.h" |
937 | 23 #include "voc.h" |
24 | |
25 | |
26 static int voc_probe(AVProbeData *p) | |
27 { | |
28 int version, check; | |
29 | |
3764 | 30 if (memcmp(p->buf, ff_voc_magic, sizeof(ff_voc_magic) - 1)) |
937 | 31 return 0; |
2184 | 32 version = AV_RL16(p->buf + 22); |
33 check = AV_RL16(p->buf + 24); | |
937 | 34 if (~version + 0x1234 != check) |
35 return 10; | |
36 | |
37 return AVPROBE_SCORE_MAX; | |
38 } | |
39 | |
40 static int voc_read_header(AVFormatContext *s, AVFormatParameters *ap) | |
41 { | |
4081 | 42 VocDecContext *voc = s->priv_data; |
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Use dynamically allocated ByteIOContext in AVFormatContext
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43 ByteIOContext *pb = s->pb; |
937 | 44 int header_size; |
45 AVStream *st; | |
46 | |
47 url_fskip(pb, 20); | |
48 header_size = get_le16(pb) - 22; | |
49 if (header_size != 4) { | |
1883 | 50 av_log(s, AV_LOG_ERROR, "unknown header size: %d\n", header_size); |
2276
3c1e2d519277
Replace all occurrences of AVERROR_NOTSUPP with AVERROR(ENOSYS).
takis
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51 return AVERROR(ENOSYS); |
937 | 52 } |
53 url_fskip(pb, header_size); | |
54 st = av_new_stream(s, 0); | |
55 if (!st) | |
2273
7eb456c4ed8a
Replace all occurrences of AVERROR_NOMEM with AVERROR(ENOMEM).
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56 return AVERROR(ENOMEM); |
937 | 57 st->codec->codec_type = CODEC_TYPE_AUDIO; |
58 | |
59 voc->remaining_size = 0; | |
60 return 0; | |
61 } | |
62 | |
63 int | |
64 voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size) | |
65 { | |
4081 | 66 VocDecContext *voc = s->priv_data; |
937 | 67 AVCodecContext *dec = st->codec; |
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68 ByteIOContext *pb = s->pb; |
4081 | 69 VocType type; |
937 | 70 int size; |
71 int sample_rate = 0; | |
72 int channels = 1; | |
73 | |
74 while (!voc->remaining_size) { | |
75 type = get_byte(pb); | |
76 if (type == VOC_TYPE_EOF) | |
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b21c2af60bc9
Replace all occurrences of AVERROR_IO with AVERROR(EIO).
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77 return AVERROR(EIO); |
937 | 78 voc->remaining_size = get_le24(pb); |
5384
3528d151a57e
Determine VOC block length when length field is zero, fixes issue1538.
pross
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79 if (!voc->remaining_size) { |
3528d151a57e
Determine VOC block length when length field is zero, fixes issue1538.
pross
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80 if (url_is_streamed(s->pb)) |
3528d151a57e
Determine VOC block length when length field is zero, fixes issue1538.
pross
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81 return AVERROR(EIO); |
3528d151a57e
Determine VOC block length when length field is zero, fixes issue1538.
pross
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82 voc->remaining_size = url_fsize(pb) - url_ftell(pb); |
3528d151a57e
Determine VOC block length when length field is zero, fixes issue1538.
pross
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83 } |
937 | 84 max_size -= 4; |
85 | |
86 switch (type) { | |
87 case VOC_TYPE_VOICE_DATA: | |
88 dec->sample_rate = 1000000 / (256 - get_byte(pb)); | |
89 if (sample_rate) | |
90 dec->sample_rate = sample_rate; | |
91 dec->channels = channels; | |
5058
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Add ff_ prefixes to exported symbols in libavformat/riff.h.
diego
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92 dec->codec_id = ff_codec_get_id(ff_voc_codec_tags, get_byte(pb)); |
3908
1d3d17de20ba
Bump Major version, this commit is almost just renaming bits_per_sample to
michael
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93 dec->bits_per_coded_sample = av_get_bits_per_sample(dec->codec_id); |
937 | 94 voc->remaining_size -= 2; |
95 max_size -= 2; | |
96 channels = 1; | |
97 break; | |
98 | |
99 case VOC_TYPE_VOICE_DATA_CONT: | |
100 break; | |
101 | |
102 case VOC_TYPE_EXTENDED: | |
103 sample_rate = get_le16(pb); | |
104 get_byte(pb); | |
105 channels = get_byte(pb) + 1; | |
106 sample_rate = 256000000 / (channels * (65536 - sample_rate)); | |
107 voc->remaining_size = 0; | |
108 max_size -= 4; | |
109 break; | |
110 | |
111 case VOC_TYPE_NEW_VOICE_DATA: | |
112 dec->sample_rate = get_le32(pb); | |
3908
1d3d17de20ba
Bump Major version, this commit is almost just renaming bits_per_sample to
michael
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113 dec->bits_per_coded_sample = get_byte(pb); |
937 | 114 dec->channels = get_byte(pb); |
5058
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Add ff_ prefixes to exported symbols in libavformat/riff.h.
diego
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115 dec->codec_id = ff_codec_get_id(ff_voc_codec_tags, get_le16(pb)); |
937 | 116 url_fskip(pb, 4); |
117 voc->remaining_size -= 12; | |
118 max_size -= 12; | |
119 break; | |
120 | |
121 default: | |
122 url_fskip(pb, voc->remaining_size); | |
123 max_size -= voc->remaining_size; | |
124 voc->remaining_size = 0; | |
125 break; | |
126 } | |
127 } | |
128 | |
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Bump Major version, this commit is almost just renaming bits_per_sample to
michael
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129 dec->bit_rate = dec->sample_rate * dec->bits_per_coded_sample; |
937 | 130 |
131 if (max_size <= 0) | |
1884 | 132 max_size = 2048; |
937 | 133 size = FFMIN(voc->remaining_size, max_size); |
134 voc->remaining_size -= size; | |
135 return av_get_packet(pb, pkt, size); | |
136 } | |
137 | |
138 static int voc_read_packet(AVFormatContext *s, AVPacket *pkt) | |
139 { | |
140 return voc_get_packet(s, pkt, s->streams[0], 0); | |
141 } | |
142 | |
1169 | 143 AVInputFormat voc_demuxer = { |
937 | 144 "voc", |
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7a0230981402
Make long_names in lavf/lavdev optional depending on CONFIG_SMALL.
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145 NULL_IF_CONFIG_SMALL("Creative Voice file format"), |
4081 | 146 sizeof(VocDecContext), |
937 | 147 voc_probe, |
148 voc_read_header, | |
149 voc_read_packet, | |
3766
f062deeedb8d
Change codec_tag type from const struct AVCodecTag ** to const struct AVCodecTag * const *
reimar
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150 .codec_tag=(const AVCodecTag* const []){ff_voc_codec_tags, 0}, |
937 | 151 }; |