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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 FFmpeg.
6  *
7  * FFmpeg 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  * FFmpeg 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 FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * Westwood Studios AUD file demuxer
25  * by Mike Melanson (melanson@pcisys.net)
26  * for more information on the Westwood file formats, visit:
27  * http://www.pcisys.net/~melanson/codecs/
28  * http://www.geocities.com/SiliconValley/8682/aud3.txt
29  *
30  * Implementation note: There is no definite file signature for AUD files.
31  * The demuxer uses a probabilistic strategy for content detection. This
32  * entails performing sanity checks on certain header values in order to
33  * qualify a file. Refer to wsaud_probe() for the precise parameters.
34  */
35 
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  if (p->buf[11] != 99 && p->buf[11] != 1)
73  return 0;
74 
75  /* read ahead to the first audio chunk and validate the first header signature */
76  if (AV_RL32(&p->buf[16]) != AUD_CHUNK_SIGNATURE)
77  return 0;
78 
79  /* return 1/2 certainty since this file check is a little sketchy */
80  return AVPROBE_SCORE_MAX / 2;
81 }
82 
84 {
85  AVIOContext *pb = s->pb;
86  AVStream *st;
87  unsigned char header[AUD_HEADER_SIZE];
88  int sample_rate, channels, codec;
89 
90  if (avio_read(pb, header, AUD_HEADER_SIZE) != AUD_HEADER_SIZE)
91  return AVERROR(EIO);
92 
93  sample_rate = AV_RL16(&header[0]);
94  channels = (header[10] & 0x1) + 1;
95  codec = header[11];
96 
97  /* initialize the audio decoder stream */
98  st = avformat_new_stream(s, NULL);
99  if (!st)
100  return AVERROR(ENOMEM);
101 
102  switch (codec) {
103  case 1:
104  if (channels != 1) {
105  av_log_ask_for_sample(s, "Stereo WS-SND1 is not supported.\n");
106  return AVERROR_PATCHWELCOME;
107  }
109  break;
110  case 99:
112  st->codec->bits_per_coded_sample = 4;
113  st->codec->bit_rate = channels * sample_rate * 4;
114  break;
115  default:
116  av_log_ask_for_sample(s, "Unknown codec: %d\n", codec);
117  return AVERROR_PATCHWELCOME;
118  }
119  avpriv_set_pts_info(st, 64, 1, sample_rate);
121  st->codec->channels = channels;
122  st->codec->channel_layout = channels == 1 ? AV_CH_LAYOUT_MONO :
125 
126  return 0;
127 }
128 
130  AVPacket *pkt)
131 {
132  AVIOContext *pb = s->pb;
133  unsigned char preamble[AUD_CHUNK_PREAMBLE_SIZE];
134  unsigned int chunk_size;
135  int ret = 0;
136  AVStream *st = s->streams[0];
137 
138  if (avio_read(pb, preamble, AUD_CHUNK_PREAMBLE_SIZE) !=
140  return AVERROR(EIO);
141 
142  /* validate the chunk */
143  if (AV_RL32(&preamble[4]) != AUD_CHUNK_SIGNATURE)
144  return AVERROR_INVALIDDATA;
145 
146  chunk_size = AV_RL16(&preamble[0]);
147 
149  /* For Westwood SND1 audio we need to add the output size and input
150  size to the start of the packet to match what is in VQA.
151  Specifically, this is needed to signal when a packet should be
152  decoding as raw 8-bit pcm or variable-size ADPCM. */
153  int out_size = AV_RL16(&preamble[2]);
154  if ((ret = av_new_packet(pkt, chunk_size + 4)))
155  return ret;
156  if ((ret = avio_read(pb, &pkt->data[4], chunk_size)) != chunk_size)
157  return ret < 0 ? ret : AVERROR(EIO);
158  AV_WL16(&pkt->data[0], out_size);
159  AV_WL16(&pkt->data[2], chunk_size);
160 
161  pkt->duration = out_size;
162  } else {
163  ret = av_get_packet(pb, pkt, chunk_size);
164  if (ret != chunk_size)
165  return AVERROR(EIO);
166 
167  /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
168  pkt->duration = (chunk_size * 2) / st->codec->channels;
169  }
170  pkt->stream_index = st->index;
171 
172  return ret;
173 }
174 
176  .name = "wsaud",
177  .long_name = NULL_IF_CONFIG_SMALL("Westwood Studios audio"),
178  .read_probe = wsaud_probe,
179  .read_header = wsaud_read_header,
180  .read_packet = wsaud_read_packet,
181 };