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
36 #include "libavutil/channel_layout.h"
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
wsaud_probe(const AVProbeData * p)45 static int wsaud_probe(const 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
59 if (p->buf_size < AUD_HEADER_SIZE + AUD_CHUNK_PREAMBLE_SIZE)
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_EXTENSION;
81 }
82
wsaud_read_header(AVFormatContext * s)83 static int wsaud_read_header(AVFormatContext *s)
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 avpriv_request_sample(s, "Stereo WS-SND1");
106 return AVERROR_PATCHWELCOME;
107 }
108 st->codecpar->codec_id = AV_CODEC_ID_WESTWOOD_SND1;
109 break;
110 case 99:
111 st->codecpar->codec_id = AV_CODEC_ID_ADPCM_IMA_WS;
112 st->codecpar->bits_per_coded_sample = 4;
113 st->codecpar->bit_rate = channels * sample_rate * 4;
114 break;
115 default:
116 avpriv_request_sample(s, "Unknown codec: %d", codec);
117 return AVERROR_PATCHWELCOME;
118 }
119 avpriv_set_pts_info(st, 64, 1, sample_rate);
120 st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
121 av_channel_layout_default(&st->codecpar->ch_layout, channels);
122 st->codecpar->sample_rate = sample_rate;
123
124 return 0;
125 }
126
wsaud_read_packet(AVFormatContext * s,AVPacket * pkt)127 static int wsaud_read_packet(AVFormatContext *s,
128 AVPacket *pkt)
129 {
130 AVIOContext *pb = s->pb;
131 unsigned char preamble[AUD_CHUNK_PREAMBLE_SIZE];
132 unsigned int chunk_size;
133 int ret = 0;
134 AVStream *st = s->streams[0];
135
136 if (avio_read(pb, preamble, AUD_CHUNK_PREAMBLE_SIZE) !=
137 AUD_CHUNK_PREAMBLE_SIZE)
138 return AVERROR(EIO);
139
140 /* validate the chunk */
141 if (AV_RL32(&preamble[4]) != AUD_CHUNK_SIGNATURE)
142 return AVERROR_INVALIDDATA;
143
144 chunk_size = AV_RL16(&preamble[0]);
145
146 if (st->codecpar->codec_id == AV_CODEC_ID_WESTWOOD_SND1) {
147 /* For Westwood SND1 audio we need to add the output size and input
148 size to the start of the packet to match what is in VQA.
149 Specifically, this is needed to signal when a packet should be
150 decoding as raw 8-bit pcm or variable-size ADPCM. */
151 int out_size = AV_RL16(&preamble[2]);
152 if ((ret = av_new_packet(pkt, chunk_size + 4)) < 0)
153 return ret;
154 if ((ret = avio_read(pb, &pkt->data[4], chunk_size)) != chunk_size)
155 return ret < 0 ? ret : AVERROR(EIO);
156 AV_WL16(&pkt->data[0], out_size);
157 AV_WL16(&pkt->data[2], chunk_size);
158
159 pkt->duration = out_size;
160 } else {
161 ret = av_get_packet(pb, pkt, chunk_size);
162 if (ret != chunk_size)
163 return AVERROR(EIO);
164
165 if (st->codecpar->ch_layout.nb_channels <= 0) {
166 av_log(s, AV_LOG_ERROR, "invalid number of channels %d\n",
167 st->codecpar->ch_layout.nb_channels);
168 return AVERROR_INVALIDDATA;
169 }
170
171 /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
172 pkt->duration = (chunk_size * 2) / st->codecpar->ch_layout.nb_channels;
173 }
174 pkt->stream_index = st->index;
175
176 return ret;
177 }
178
179 const AVInputFormat ff_wsaud_demuxer = {
180 .name = "wsaud",
181 .long_name = NULL_IF_CONFIG_SMALL("Westwood Studios audio"),
182 .read_probe = wsaud_probe,
183 .read_header = wsaud_read_header,
184 .read_packet = wsaud_read_packet,
185 };
186