79f219874553077c268b9964011903a1873a455d
[libav.git] / libavformat / westwood_aud.c
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
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/intreadwrite.h"
37 #include "avformat.h"
38 #include "internal.h"
39
40 #define AUD_HEADER_SIZE 12
41 #define AUD_CHUNK_PREAMBLE_SIZE 8
42 #define AUD_CHUNK_SIGNATURE 0x0000DEAF
43
44 typedef struct WsAudDemuxContext {
45 int audio_channels;
46 int audio_samplerate;
47 int audio_stream_index;
48 int64_t audio_frame_counter;
49 } WsAudDemuxContext;
50
51 static int wsaud_probe(AVProbeData *p)
52 {
53 int field;
54
55 /* Probabilistic content detection strategy: There is no file signature
56 * so perform sanity checks on various header parameters:
57 * 8000 <= sample rate (16 bits) <= 48000 ==> 40001 acceptable numbers
58 * flags <= 0x03 (2 LSBs are used) ==> 4 acceptable numbers
59 * compression type (8 bits) = 1 or 99 ==> 2 acceptable numbers
60 * first audio chunk signature (32 bits) ==> 1 acceptable number
61 * The number space contains 2^64 numbers. There are 40001 * 4 * 2 * 1 =
62 * 320008 acceptable number combinations.
63 */
64
65 if (p->buf_size < AUD_HEADER_SIZE + AUD_CHUNK_PREAMBLE_SIZE)
66 return 0;
67
68 /* check sample rate */
69 field = AV_RL16(&p->buf[0]);
70 if ((field < 8000) || (field > 48000))
71 return 0;
72
73 /* enforce the rule that the top 6 bits of this flags field are reserved (0);
74 * this might not be true, but enforce it until deemed unnecessary */
75 if (p->buf[10] & 0xFC)
76 return 0;
77
78 /* note: only check for WS IMA (type 99) right now since there is no
79 * support for type 1 */
80 if (p->buf[11] != 99 && p->buf[11] != 1)
81 return 0;
82
83 /* read ahead to the first audio chunk and validate the first header signature */
84 if (AV_RL32(&p->buf[16]) != AUD_CHUNK_SIGNATURE)
85 return 0;
86
87 /* return 1/2 certainty since this file check is a little sketchy */
88 return AVPROBE_SCORE_MAX / 2;
89 }
90
91 static int wsaud_read_header(AVFormatContext *s,
92 AVFormatParameters *ap)
93 {
94 WsAudDemuxContext *wsaud = s->priv_data;
95 AVIOContext *pb = s->pb;
96 AVStream *st;
97 unsigned char header[AUD_HEADER_SIZE];
98 int sample_rate, channels, codec;
99
100 if (avio_read(pb, header, AUD_HEADER_SIZE) != AUD_HEADER_SIZE)
101 return AVERROR(EIO);
102
103 sample_rate = AV_RL16(&header[0]);
104 channels = (header[10] & 0x1) + 1;
105 codec = header[11];
106
107 /* initialize the audio decoder stream */
108 st = avformat_new_stream(s, NULL);
109 if (!st)
110 return AVERROR(ENOMEM);
111
112 switch (codec) {
113 case 1:
114 if (channels != 1) {
115 av_log_ask_for_sample(s, "Stereo WS-SND1 is not supported.\n");
116 return AVERROR_PATCHWELCOME;
117 }
118 st->codec->codec_id = CODEC_ID_WESTWOOD_SND1;
119 break;
120 case 99:
121 st->codec->codec_id = CODEC_ID_ADPCM_IMA_WS;
122 st->codec->bits_per_coded_sample = 4;
123 st->codec->bit_rate = channels * sample_rate * 4;
124 break;
125 default:
126 av_log_ask_for_sample(s, "Unknown codec: %d\n", codec);
127 return AVERROR_PATCHWELCOME;
128 }
129 avpriv_set_pts_info(st, 64, 1, sample_rate);
130 st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
131 st->codec->channels = channels;
132 st->codec->sample_rate = sample_rate;
133
134 wsaud->audio_channels = channels;
135 wsaud->audio_samplerate = sample_rate;
136 wsaud->audio_stream_index = st->index;
137 wsaud->audio_frame_counter = 0;
138
139 return 0;
140 }
141
142 static int wsaud_read_packet(AVFormatContext *s,
143 AVPacket *pkt)
144 {
145 WsAudDemuxContext *wsaud = s->priv_data;
146 AVIOContext *pb = s->pb;
147 unsigned char preamble[AUD_CHUNK_PREAMBLE_SIZE];
148 unsigned int chunk_size;
149 int ret = 0;
150 AVStream *st = s->streams[wsaud->audio_stream_index];
151
152 if (avio_read(pb, preamble, AUD_CHUNK_PREAMBLE_SIZE) !=
153 AUD_CHUNK_PREAMBLE_SIZE)
154 return AVERROR(EIO);
155
156 /* validate the chunk */
157 if (AV_RL32(&preamble[4]) != AUD_CHUNK_SIGNATURE)
158 return AVERROR_INVALIDDATA;
159
160 chunk_size = AV_RL16(&preamble[0]);
161
162 if (st->codec->codec_id == CODEC_ID_WESTWOOD_SND1) {
163 /* For Westwood SND1 audio we need to add the output size and input
164 size to the start of the packet to match what is in VQA.
165 Specifically, this is needed to signal when a packet should be
166 decoding as raw 8-bit pcm or variable-size ADPCM. */
167 int out_size = AV_RL16(&preamble[2]);
168 if ((ret = av_new_packet(pkt, chunk_size + 4)))
169 return ret;
170 if ((ret = avio_read(pb, &pkt->data[4], chunk_size)) != chunk_size)
171 return ret < 0 ? ret : AVERROR(EIO);
172 AV_WL16(&pkt->data[0], out_size);
173 AV_WL16(&pkt->data[2], chunk_size);
174
175 pkt->duration = out_size;
176 } else {
177 ret = av_get_packet(pb, pkt, chunk_size);
178 if (ret != chunk_size)
179 return AVERROR(EIO);
180 pkt->pts = wsaud->audio_frame_counter;
181 pkt->pts /= wsaud->audio_samplerate;
182
183 /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
184 wsaud->audio_frame_counter += (chunk_size * 2) / wsaud->audio_channels;
185 }
186 pkt->stream_index = st->index;
187
188 return ret;
189 }
190
191 AVInputFormat ff_wsaud_demuxer = {
192 .name = "wsaud",
193 .long_name = NULL_IF_CONFIG_SMALL("Westwood Studios audio format"),
194 .priv_data_size = sizeof(WsAudDemuxContext),
195 .read_probe = wsaud_probe,
196 .read_header = wsaud_read_header,
197 .read_packet = wsaud_read_packet,
198 };