Libav
westwood_aud.c
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1 /*
2  * Westwood Studios AUD Format Demuxer
3  * Copyright (c) 2003 The ffmpeg Project
4  *
5  * This file is part of Libav.
6  *
7  * Libav is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * Libav is distributed in the hope that it will be useful,
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
18  * License along with Libav; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
37 #include "libavutil/intreadwrite.h"
38 #include "avformat.h"
39 #include "internal.h"
40 
41 #define AUD_HEADER_SIZE 12
42 #define AUD_CHUNK_PREAMBLE_SIZE 8
43 #define AUD_CHUNK_SIGNATURE 0x0000DEAF
44 
45 static int wsaud_probe(AVProbeData *p)
46 {
47  int field;
48 
49  /* Probabilistic content detection strategy: There is no file signature
50  * so perform sanity checks on various header parameters:
51  * 8000 <= sample rate (16 bits) <= 48000 ==> 40001 acceptable numbers
52  * flags <= 0x03 (2 LSBs are used) ==> 4 acceptable numbers
53  * compression type (8 bits) = 1 or 99 ==> 2 acceptable numbers
54  * first audio chunk signature (32 bits) ==> 1 acceptable number
55  * The number space contains 2^64 numbers. There are 40001 * 4 * 2 * 1 =
56  * 320008 acceptable number combinations.
57  */
58 
60  return 0;
61 
62  /* check sample rate */
63  field = AV_RL16(&p->buf[0]);
64  if ((field < 8000) || (field > 48000))
65  return 0;
66 
67  /* enforce the rule that the top 6 bits of this flags field are reserved (0);
68  * this might not be true, but enforce it until deemed unnecessary */
69  if (p->buf[10] & 0xFC)
70  return 0;
71 
72  /* note: only check for WS IMA (type 99) right now since there is no
73  * support for type 1 */
74  if (p->buf[11] != 99 && p->buf[11] != 1)
75  return 0;
76 
77  /* read ahead to the first audio chunk and validate the first header signature */
78  if (AV_RL32(&p->buf[16]) != AUD_CHUNK_SIGNATURE)
79  return 0;
80 
81  /* return 1/2 certainty since this file check is a little sketchy */
83 }
84 
86 {
87  AVIOContext *pb = s->pb;
88  AVStream *st;
89  unsigned char header[AUD_HEADER_SIZE];
90  int sample_rate, channels, codec;
91 
92  if (avio_read(pb, header, AUD_HEADER_SIZE) != AUD_HEADER_SIZE)
93  return AVERROR(EIO);
94 
95  sample_rate = AV_RL16(&header[0]);
96  channels = (header[10] & 0x1) + 1;
97  codec = header[11];
98 
99  /* initialize the audio decoder stream */
100  st = avformat_new_stream(s, NULL);
101  if (!st)
102  return AVERROR(ENOMEM);
103 
104  switch (codec) {
105  case 1:
106  if (channels != 1) {
107  avpriv_request_sample(s, "Stereo WS-SND1");
108  return AVERROR_PATCHWELCOME;
109  }
111  break;
112  case 99:
114  st->codec->bits_per_coded_sample = 4;
115  st->codec->bit_rate = channels * sample_rate * 4;
116  break;
117  default:
118  avpriv_request_sample(s, "Unknown codec: %d", codec);
119  return AVERROR_PATCHWELCOME;
120  }
121  avpriv_set_pts_info(st, 64, 1, sample_rate);
123  st->codec->channels = channels;
124  st->codec->channel_layout = channels == 1 ? AV_CH_LAYOUT_MONO :
126  st->codec->sample_rate = sample_rate;
127 
128  return 0;
129 }
130 
132  AVPacket *pkt)
133 {
134  AVIOContext *pb = s->pb;
135  unsigned char preamble[AUD_CHUNK_PREAMBLE_SIZE];
136  unsigned int chunk_size;
137  int ret = 0;
138  AVStream *st = s->streams[0];
139 
140  if (avio_read(pb, preamble, AUD_CHUNK_PREAMBLE_SIZE) !=
142  return AVERROR(EIO);
143 
144  /* validate the chunk */
145  if (AV_RL32(&preamble[4]) != AUD_CHUNK_SIGNATURE)
146  return AVERROR_INVALIDDATA;
147 
148  chunk_size = AV_RL16(&preamble[0]);
149 
151  /* For Westwood SND1 audio we need to add the output size and input
152  size to the start of the packet to match what is in VQA.
153  Specifically, this is needed to signal when a packet should be
154  decoding as raw 8-bit pcm or variable-size ADPCM. */
155  int out_size = AV_RL16(&preamble[2]);
156  if ((ret = av_new_packet(pkt, chunk_size + 4)))
157  return ret;
158  if ((ret = avio_read(pb, &pkt->data[4], chunk_size)) != chunk_size)
159  return ret < 0 ? ret : AVERROR(EIO);
160  AV_WL16(&pkt->data[0], out_size);
161  AV_WL16(&pkt->data[2], chunk_size);
162 
163  pkt->duration = out_size;
164  } else {
165  ret = av_get_packet(pb, pkt, chunk_size);
166  if (ret != chunk_size)
167  return AVERROR(EIO);
168 
169  /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
170  pkt->duration = (chunk_size * 2) / st->codec->channels;
171  }
172  pkt->stream_index = st->index;
173 
174  return ret;
175 }
176 
178  .name = "wsaud",
179  .long_name = NULL_IF_CONFIG_SMALL("Westwood Studios audio"),
180  .read_probe = wsaud_probe,
181  .read_header = wsaud_read_header,
182  .read_packet = wsaud_read_packet,
183 };