Mercurial > libavformat.hg
annotate wav.c @ 1017:47dc020f5215 libavformat
Reenable NUT and NUT regression tests as requested by Michael.
author | diego |
---|---|
date | Fri, 17 Mar 2006 23:21:18 +0000 |
parents | edbe5c3717f9 |
children | 3aeb079a7e74 |
rev | line source |
---|---|
885 | 1 /* |
0 | 2 * WAV encoder and decoder |
3 * Copyright (c) 2001, 2002 Fabrice Bellard. | |
4 * | |
5 * This library is free software; you can redistribute it and/or | |
6 * modify it under the terms of the GNU Lesser General Public | |
7 * License as published by the Free Software Foundation; either | |
8 * version 2 of the License, or (at your option) any later version. | |
9 * | |
10 * This library is distributed in the hope that it will be useful, | |
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
13 * Lesser General Public License for more details. | |
14 * | |
15 * You should have received a copy of the GNU Lesser General Public | |
16 * License along with this library; if not, write to the Free Software | |
896
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Update licensing information: The FSF changed postal address.
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17 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
0 | 18 */ |
19 #include "avformat.h" | |
20 #include "avi.h" | |
21 | |
22 const CodecTag codec_wav_tags[] = { | |
23 { CODEC_ID_MP2, 0x50 }, | |
232 | 24 { CODEC_ID_MP3, 0x55 }, |
0 | 25 { CODEC_ID_AC3, 0x2000 }, |
874
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DTS is 0x2001 according to: http://www1.mplayerhq.hu/MPlayer/DOCS/codecs-status.html
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26 { CODEC_ID_DTS, 0x2001 }, |
0 | 27 { CODEC_ID_PCM_S16LE, 0x01 }, |
28 { CODEC_ID_PCM_U8, 0x01 }, /* must come after s16le in this list */ | |
846
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29 { CODEC_ID_PCM_S24LE, 0x01 }, |
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30 { CODEC_ID_PCM_S32LE, 0x01 }, |
0 | 31 { CODEC_ID_PCM_ALAW, 0x06 }, |
32 { CODEC_ID_PCM_MULAW, 0x07 }, | |
33 { CODEC_ID_ADPCM_MS, 0x02 }, | |
34 { CODEC_ID_ADPCM_IMA_WAV, 0x11 }, | |
817
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Yamaha ADPCM in wav patch by (Vidar Madsen: vidarino, gmail com)
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35 { CODEC_ID_ADPCM_YAMAHA, 0x20 }, |
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* Initial implementation of the G.726 ADPCM audio codec.
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36 { CODEC_ID_ADPCM_G726, 0x45 }, |
226 | 37 { CODEC_ID_ADPCM_IMA_DK4, 0x61 }, /* rogue format number */ |
38 { CODEC_ID_ADPCM_IMA_DK3, 0x62 }, /* rogue format number */ | |
0 | 39 { CODEC_ID_WMAV1, 0x160 }, |
40 { CODEC_ID_WMAV2, 0x161 }, | |
550
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untested AAC in WAV/AVI patch by (Mns Rullgrd <mru at mru dot ath dot cx>)
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41 { CODEC_ID_AAC, 0x706d }, |
411
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some random id for vorbis so we can do some experiments with vorbis in various containers, anyone knows if vorbis in WAV/AVI has a official id?
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42 { CODEC_ID_VORBIS, ('V'<<8)+'o' }, //HACK/FIXME, does vorbis in WAV/AVI have an (in)official id? |
517 | 43 { CODEC_ID_SONIC, 0x2048 }, |
44 { CODEC_ID_SONIC_LS, 0x2048 }, | |
561
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Creative ADPCM decoder, format 0x200, courtesy of Konstantin Shishkov
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45 { CODEC_ID_ADPCM_CT, 0x200 }, |
685 | 46 { CODEC_ID_ADPCM_SWF, ('S'<<8)+'F' }, |
893
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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47 { CODEC_ID_TRUESPEECH, 0x22 }, |
0 | 48 { 0, 0 }, |
49 }; | |
50 | |
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51 #ifdef CONFIG_MUXERS |
0 | 52 /* WAVEFORMATEX header */ |
53 /* returns the size or -1 on error */ | |
54 int put_wav_header(ByteIOContext *pb, AVCodecContext *enc) | |
55 { | |
196 | 56 int bps, blkalign, bytespersec; |
0 | 57 int hdrsize = 18; |
58 | |
196 | 59 if(!enc->codec_tag) |
60 enc->codec_tag = codec_get_tag(codec_wav_tags, enc->codec_id); | |
61 if(!enc->codec_tag) | |
0 | 62 return -1; |
196 | 63 |
64 put_le16(pb, enc->codec_tag); | |
0 | 65 put_le16(pb, enc->channels); |
66 put_le32(pb, enc->sample_rate); | |
67 if (enc->codec_id == CODEC_ID_PCM_U8 || | |
68 enc->codec_id == CODEC_ID_PCM_ALAW || | |
69 enc->codec_id == CODEC_ID_PCM_MULAW) { | |
70 bps = 8; | |
232 | 71 } else if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3) { |
0 | 72 bps = 0; |
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Yamaha ADPCM in wav patch by (Vidar Madsen: vidarino, gmail com)
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73 } else if (enc->codec_id == CODEC_ID_ADPCM_IMA_WAV || enc->codec_id == CODEC_ID_ADPCM_MS || enc->codec_id == CODEC_ID_ADPCM_G726 || enc->codec_id == CODEC_ID_ADPCM_YAMAHA) { // |
0 | 74 bps = 4; |
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75 } else if (enc->codec_id == CODEC_ID_PCM_S24LE) { |
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reimar
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76 bps = 24; |
fd1c2109505e
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reimar
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77 } else if (enc->codec_id == CODEC_ID_PCM_S32LE) { |
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reimar
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78 bps = 32; |
0 | 79 } else { |
80 bps = 16; | |
81 } | |
885 | 82 |
232 | 83 if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3) { |
577 | 84 blkalign = enc->frame_size; //this is wrong, but seems many demuxers dont work if this is set correctly |
0 | 85 //blkalign = 144 * enc->bit_rate/enc->sample_rate; |
713
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28_fix_parameters_in_G726.patch by (Calcium | calcium nurs or jp)
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86 } else if (enc->codec_id == CODEC_ID_ADPCM_G726) { // |
3c8a21cd6296
28_fix_parameters_in_G726.patch by (Calcium | calcium nurs or jp)
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87 blkalign = 1; |
0 | 88 } else if (enc->block_align != 0) { /* specified by the codec */ |
89 blkalign = enc->block_align; | |
90 } else | |
91 blkalign = enc->channels*bps >> 3; | |
92 if (enc->codec_id == CODEC_ID_PCM_U8 || | |
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93 enc->codec_id == CODEC_ID_PCM_S24LE || |
fd1c2109505e
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reimar
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94 enc->codec_id == CODEC_ID_PCM_S32LE || |
0 | 95 enc->codec_id == CODEC_ID_PCM_S16LE) { |
96 bytespersec = enc->sample_rate * blkalign; | |
97 } else { | |
98 bytespersec = enc->bit_rate / 8; | |
99 } | |
100 put_le32(pb, bytespersec); /* bytes per second */ | |
101 put_le16(pb, blkalign); /* block align */ | |
102 put_le16(pb, bps); /* bits per sample */ | |
232 | 103 if (enc->codec_id == CODEC_ID_MP3) { |
0 | 104 put_le16(pb, 12); /* wav_extra_size */ |
105 hdrsize += 12; | |
106 put_le16(pb, 1); /* wID */ | |
107 put_le32(pb, 2); /* fdwFlags */ | |
108 put_le16(pb, 1152); /* nBlockSize */ | |
109 put_le16(pb, 1); /* nFramesPerBlock */ | |
110 put_le16(pb, 1393); /* nCodecDelay */ | |
111 } else if (enc->codec_id == CODEC_ID_MP2) { | |
112 put_le16(pb, 22); /* wav_extra_size */ | |
113 hdrsize += 22; | |
114 put_le16(pb, 2); /* fwHeadLayer */ | |
115 put_le32(pb, enc->bit_rate); /* dwHeadBitrate */ | |
116 put_le16(pb, enc->channels == 2 ? 1 : 8); /* fwHeadMode */ | |
117 put_le16(pb, 0); /* fwHeadModeExt */ | |
118 put_le16(pb, 1); /* wHeadEmphasis */ | |
119 put_le16(pb, 16); /* fwHeadFlags */ | |
120 put_le32(pb, 0); /* dwPTSLow */ | |
121 put_le32(pb, 0); /* dwPTSHigh */ | |
122 } else if (enc->codec_id == CODEC_ID_ADPCM_IMA_WAV) { | |
123 put_le16(pb, 2); /* wav_extra_size */ | |
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Correcting the header size for ADPCM_IMA_WAV files patch by (Brian Becker <brian dot becker at palmone dot com>)
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124 hdrsize += 2; |
0 | 125 put_le16(pb, ((enc->block_align - 4 * enc->channels) / (4 * enc->channels)) * 8 + 1); /* wSamplesPerBlock */ |
529 | 126 } else if(enc->extradata_size){ |
412
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127 put_le16(pb, enc->extradata_size); |
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128 put_buffer(pb, enc->extradata, enc->extradata_size); |
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129 hdrsize += enc->extradata_size; |
413 | 130 if(hdrsize&1){ |
131 hdrsize++; | |
132 put_byte(pb, 0); | |
133 } | |
529 | 134 } else { |
135 hdrsize -= 2; | |
412
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136 } |
0 | 137 |
138 return hdrsize; | |
139 } | |
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140 #endif //CONFIG_MUXERS |
0 | 141 |
84
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142 /* We could be given one of the three possible structures here: |
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143 * WAVEFORMAT, PCMWAVEFORMAT or WAVEFORMATEX. Each structure |
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144 * is an expansion of the previous one with the fields added |
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145 * at the bottom. PCMWAVEFORMAT adds 'WORD wBitsPerSample' and |
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146 * WAVEFORMATEX adds 'WORD cbSize' and basically makes itself |
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147 * an openended structure. |
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148 */ |
885 | 149 void get_wav_header(ByteIOContext *pb, AVCodecContext *codec, int size) |
0 | 150 { |
151 int id; | |
152 | |
153 id = get_le16(pb); | |
154 codec->codec_type = CODEC_TYPE_AUDIO; | |
155 codec->codec_tag = id; | |
156 codec->channels = get_le16(pb); | |
157 codec->sample_rate = get_le32(pb); | |
158 codec->bit_rate = get_le32(pb) * 8; | |
159 codec->block_align = get_le16(pb); | |
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160 if (size == 14) { /* We're dealing with plain vanilla WAVEFORMAT */ |
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161 codec->bits_per_sample = 8; |
87 | 162 }else |
163 codec->bits_per_sample = get_le16(pb); | |
164 codec->codec_id = wav_codec_get_id(id, codec->bits_per_sample); | |
885 | 165 |
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166 if (size > 16) { /* We're obviously dealing with WAVEFORMATEX */ |
887 | 167 codec->extradata_size = get_le16(pb); |
168 if (codec->extradata_size > 0) { | |
169 if (codec->extradata_size > size - 18) | |
170 codec->extradata_size = size - 18; | |
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allocate a few bytes more for extradata so the bitstream reader if its used by the codec for parsing extardata, doesnt read over the end
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171 codec->extradata = av_mallocz(codec->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE); |
0 | 172 get_buffer(pb, codec->extradata, codec->extradata_size); |
84
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173 } else |
887 | 174 codec->extradata_size = 0; |
885 | 175 |
887 | 176 /* It is possible for the chunk to contain garbage at the end */ |
177 if (size - codec->extradata_size - 18 > 0) | |
178 url_fskip(pb, size - codec->extradata_size - 18); | |
0 | 179 } |
180 } | |
181 | |
182 | |
183 int wav_codec_get_id(unsigned int tag, int bps) | |
184 { | |
185 int id; | |
186 id = codec_get_id(codec_wav_tags, tag); | |
187 if (id <= 0) | |
188 return id; | |
189 /* handle specific u8 codec */ | |
190 if (id == CODEC_ID_PCM_S16LE && bps == 8) | |
191 id = CODEC_ID_PCM_U8; | |
846
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Support de-/encoding of 24 and 32 bit PCM (from and to internal 16 bit).
reimar
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820
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192 if (id == CODEC_ID_PCM_S16LE && bps == 24) |
fd1c2109505e
Support de-/encoding of 24 and 32 bit PCM (from and to internal 16 bit).
reimar
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193 id = CODEC_ID_PCM_S24LE; |
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reimar
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194 if (id == CODEC_ID_PCM_S16LE && bps == 32) |
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reimar
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820
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195 id = CODEC_ID_PCM_S32LE; |
0 | 196 return id; |
197 } | |
198 | |
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199 #ifdef CONFIG_MUXERS |
0 | 200 typedef struct { |
201 offset_t data; | |
202 } WAVContext; | |
203 | |
204 static int wav_write_header(AVFormatContext *s) | |
205 { | |
206 WAVContext *wav = s->priv_data; | |
207 ByteIOContext *pb = &s->pb; | |
208 offset_t fmt; | |
209 | |
210 put_tag(pb, "RIFF"); | |
211 put_le32(pb, 0); /* file length */ | |
212 put_tag(pb, "WAVE"); | |
213 | |
214 /* format header */ | |
215 fmt = start_tag(pb, "fmt "); | |
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216 if (put_wav_header(pb, s->streams[0]->codec) < 0) { |
0 | 217 av_free(wav); |
218 return -1; | |
219 } | |
220 end_tag(pb, fmt); | |
221 | |
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222 av_set_pts_info(s->streams[0], 64, 1, s->streams[0]->codec->sample_rate); |
645
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223 |
0 | 224 /* data header */ |
225 wav->data = start_tag(pb, "data"); | |
885 | 226 |
0 | 227 put_flush_packet(pb); |
228 | |
229 return 0; | |
230 } | |
231 | |
468 | 232 static int wav_write_packet(AVFormatContext *s, AVPacket *pkt) |
0 | 233 { |
234 ByteIOContext *pb = &s->pb; | |
468 | 235 put_buffer(pb, pkt->data, pkt->size); |
0 | 236 return 0; |
237 } | |
238 | |
239 static int wav_write_trailer(AVFormatContext *s) | |
240 { | |
241 ByteIOContext *pb = &s->pb; | |
242 WAVContext *wav = s->priv_data; | |
243 offset_t file_size; | |
244 | |
245 if (!url_is_streamed(&s->pb)) { | |
246 end_tag(pb, wav->data); | |
247 | |
248 /* update file size */ | |
249 file_size = url_ftell(pb); | |
250 url_fseek(pb, 4, SEEK_SET); | |
65 | 251 put_le32(pb, (uint32_t)(file_size - 8)); |
0 | 252 url_fseek(pb, file_size, SEEK_SET); |
253 | |
254 put_flush_packet(pb); | |
255 } | |
256 return 0; | |
257 } | |
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258 #endif //CONFIG_MUXERS |
0 | 259 |
260 /* return the size of the found tag */ | |
261 /* XXX: > 2GB ? */ | |
65 | 262 static int find_tag(ByteIOContext *pb, uint32_t tag1) |
0 | 263 { |
264 unsigned int tag; | |
265 int size; | |
266 | |
267 for(;;) { | |
268 if (url_feof(pb)) | |
269 return -1; | |
270 tag = get_le32(pb); | |
271 size = get_le32(pb); | |
272 if (tag == tag1) | |
273 break; | |
274 url_fseek(pb, size, SEEK_CUR); | |
275 } | |
276 if (size < 0) | |
277 size = 0x7fffffff; | |
278 return size; | |
279 } | |
280 | |
281 static int wav_probe(AVProbeData *p) | |
282 { | |
283 /* check file header */ | |
284 if (p->buf_size <= 32) | |
285 return 0; | |
286 if (p->buf[0] == 'R' && p->buf[1] == 'I' && | |
287 p->buf[2] == 'F' && p->buf[3] == 'F' && | |
288 p->buf[8] == 'W' && p->buf[9] == 'A' && | |
289 p->buf[10] == 'V' && p->buf[11] == 'E') | |
290 return AVPROBE_SCORE_MAX; | |
291 else | |
292 return 0; | |
293 } | |
294 | |
295 /* wav input */ | |
296 static int wav_read_header(AVFormatContext *s, | |
297 AVFormatParameters *ap) | |
298 { | |
299 int size; | |
300 unsigned int tag; | |
301 ByteIOContext *pb = &s->pb; | |
302 AVStream *st; | |
303 | |
304 /* check RIFF header */ | |
305 tag = get_le32(pb); | |
306 | |
307 if (tag != MKTAG('R', 'I', 'F', 'F')) | |
308 return -1; | |
309 get_le32(pb); /* file size */ | |
310 tag = get_le32(pb); | |
311 if (tag != MKTAG('W', 'A', 'V', 'E')) | |
312 return -1; | |
885 | 313 |
0 | 314 /* parse fmt header */ |
315 size = find_tag(pb, MKTAG('f', 'm', 't', ' ')); | |
316 if (size < 0) | |
317 return -1; | |
318 st = av_new_stream(s, 0); | |
319 if (!st) | |
320 return AVERROR_NOMEM; | |
321 | |
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322 get_wav_header(pb, st->codec, size); |
309 | 323 st->need_parsing = 1; |
567 | 324 |
820
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325 av_set_pts_info(st, 64, 1, st->codec->sample_rate); |
567 | 326 |
0 | 327 size = find_tag(pb, MKTAG('d', 'a', 't', 'a')); |
328 if (size < 0) | |
329 return -1; | |
330 return 0; | |
331 } | |
332 | |
333 #define MAX_SIZE 4096 | |
334 | |
335 static int wav_read_packet(AVFormatContext *s, | |
336 AVPacket *pkt) | |
337 { | |
309 | 338 int ret, size; |
339 AVStream *st; | |
0 | 340 |
341 if (url_feof(&s->pb)) | |
482 | 342 return AVERROR_IO; |
309 | 343 st = s->streams[0]; |
344 | |
345 size = MAX_SIZE; | |
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346 if (st->codec->block_align > 1) { |
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347 if (size < st->codec->block_align) |
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348 size = st->codec->block_align; |
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349 size = (size / st->codec->block_align) * st->codec->block_align; |
309 | 350 } |
351 if (av_new_packet(pkt, size)) | |
482 | 352 return AVERROR_IO; |
0 | 353 pkt->stream_index = 0; |
354 | |
355 ret = get_buffer(&s->pb, pkt->data, pkt->size); | |
356 if (ret < 0) | |
357 av_free_packet(pkt); | |
358 /* note: we need to modify the packet size here to handle the last | |
359 packet */ | |
360 pkt->size = ret; | |
361 return ret; | |
362 } | |
363 | |
364 static int wav_read_close(AVFormatContext *s) | |
365 { | |
366 return 0; | |
367 } | |
368 | |
885 | 369 static int wav_read_seek(AVFormatContext *s, |
558 | 370 int stream_index, int64_t timestamp, int flags) |
309 | 371 { |
372 AVStream *st; | |
373 | |
374 st = s->streams[0]; | |
820
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375 switch(st->codec->codec_id) { |
309 | 376 case CODEC_ID_MP2: |
377 case CODEC_ID_MP3: | |
378 case CODEC_ID_AC3: | |
496
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379 case CODEC_ID_DTS: |
309 | 380 /* use generic seeking with dynamically generated indexes */ |
381 return -1; | |
382 default: | |
383 break; | |
384 } | |
558 | 385 return pcm_read_seek(s, stream_index, timestamp, flags); |
309 | 386 } |
387 | |
388 | |
0 | 389 static AVInputFormat wav_iformat = { |
390 "wav", | |
391 "wav format", | |
392 0, | |
393 wav_probe, | |
394 wav_read_header, | |
395 wav_read_packet, | |
396 wav_read_close, | |
309 | 397 wav_read_seek, |
0 | 398 }; |
399 | |
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400 #ifdef CONFIG_MUXERS |
0 | 401 static AVOutputFormat wav_oformat = { |
402 "wav", | |
403 "wav format", | |
404 "audio/x-wav", | |
405 "wav", | |
406 sizeof(WAVContext), | |
407 CODEC_ID_PCM_S16LE, | |
408 CODEC_ID_NONE, | |
409 wav_write_header, | |
410 wav_write_packet, | |
411 wav_write_trailer, | |
412 }; | |
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413 #endif //CONFIG_MUXERS |
0 | 414 |
384
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415 int ff_wav_init(void) |
0 | 416 { |
417 av_register_input_format(&wav_iformat); | |
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418 #ifdef CONFIG_MUXERS |
0 | 419 av_register_output_format(&wav_oformat); |
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420 #endif //CONFIG_MUXERS |
0 | 421 return 0; |
422 } |