FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavcodec/vmdaudio.c
Date: 2021-09-16 08:47:15
Exec Total Coverage
Lines: 56 90 62.2%
Branches: 27 50 54.0%

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1 /*
2 * Sierra VMD audio decoder
3 * Copyright (c) 2004 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 * Sierra VMD audio decoder
25 * by Vladimir "VAG" Gneushev (vagsoft at mail.ru)
26 * for more information on the Sierra VMD format, visit:
27 * http://www.pcisys.net/~melanson/codecs/
28 *
29 * The audio decoder, expects each encoded data
30 * chunk to be prepended with the appropriate 16-byte frame information
31 * record from the VMD file. It does not require the 0x330-byte VMD file
32 * header, but it does need the audio setup parameters passed in through
33 * normal libavcodec API means.
34 */
35
36 #include <string.h>
37
38 #include "libavutil/avassert.h"
39 #include "libavutil/channel_layout.h"
40 #include "libavutil/common.h"
41 #include "libavutil/intreadwrite.h"
42
43 #include "avcodec.h"
44 #include "internal.h"
45
46 #define BLOCK_TYPE_AUDIO 1
47 #define BLOCK_TYPE_INITIAL 2
48 #define BLOCK_TYPE_SILENCE 3
49
50 typedef struct VmdAudioContext {
51 int out_bps;
52 int chunk_size;
53 } VmdAudioContext;
54
55 static const uint16_t vmdaudio_table[128] = {
56 0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080,
57 0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120,
58 0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0,
59 0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230,
60 0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280,
61 0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0,
62 0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320,
63 0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370,
64 0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0,
65 0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480,
66 0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700,
67 0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00,
68 0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000
69 };
70
71 4 static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)
72 {
73 4 VmdAudioContext *s = avctx->priv_data;
74
75
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4 if (avctx->channels < 1 || avctx->channels > 2) {
76 av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
77 return AVERROR(EINVAL);
78 }
79
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4 if (avctx->block_align < 1 || avctx->block_align % avctx->channels ||
80
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4 avctx->block_align > INT_MAX - avctx->channels
81 ) {
82 av_log(avctx, AV_LOG_ERROR, "invalid block align\n");
83 return AVERROR(EINVAL);
84 }
85
86
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4 avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :
87 AV_CH_LAYOUT_STEREO;
88
89
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4 if (avctx->bits_per_coded_sample == 16)
90 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
91 else
92 4 avctx->sample_fmt = AV_SAMPLE_FMT_U8;
93 4 s->out_bps = av_get_bytes_per_sample(avctx->sample_fmt);
94
95 4 s->chunk_size = avctx->block_align + avctx->channels * (s->out_bps == 2);
96
97 4 av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, "
98 "block align = %d, sample rate = %d\n",
99 avctx->channels, avctx->bits_per_coded_sample, avctx->block_align,
100 avctx->sample_rate);
101
102 4 return 0;
103 }
104
105 static void decode_audio_s16(int16_t *out, const uint8_t *buf, int buf_size,
106 int channels)
107 {
108 int ch;
109 const uint8_t *buf_end = buf + buf_size;
110 int predictor[2];
111 int st = channels - 1;
112
113 /* decode initial raw sample */
114 for (ch = 0; ch < channels; ch++) {
115 predictor[ch] = (int16_t)AV_RL16(buf);
116 buf += 2;
117 *out++ = predictor[ch];
118 }
119
120 /* decode DPCM samples */
121 ch = 0;
122 while (buf < buf_end) {
123 uint8_t b = *buf++;
124 if (b & 0x80)
125 predictor[ch] -= vmdaudio_table[b & 0x7F];
126 else
127 predictor[ch] += vmdaudio_table[b];
128 predictor[ch] = av_clip_int16(predictor[ch]);
129 *out++ = predictor[ch];
130 ch ^= st;
131 }
132 }
133
134 215 static int vmdaudio_decode_frame(AVCodecContext *avctx, void *data,
135 int *got_frame_ptr, AVPacket *avpkt)
136 {
137 215 AVFrame *frame = data;
138 215 const uint8_t *buf = avpkt->data;
139 const uint8_t *buf_end;
140 215 int buf_size = avpkt->size;
141 215 VmdAudioContext *s = avctx->priv_data;
142 int block_type, silent_chunks, audio_chunks;
143 int ret;
144 uint8_t *output_samples_u8;
145 int16_t *output_samples_s16;
146
147
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215 if (buf_size < 16) {
148 av_log(avctx, AV_LOG_WARNING, "skipping small junk packet\n");
149 *got_frame_ptr = 0;
150 return buf_size;
151 }
152
153 215 block_type = buf[6];
154
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215 if (block_type < BLOCK_TYPE_AUDIO || block_type > BLOCK_TYPE_SILENCE) {
155 av_log(avctx, AV_LOG_ERROR, "unknown block type: %d\n", block_type);
156 return AVERROR(EINVAL);
157 }
158 215 buf += 16;
159 215 buf_size -= 16;
160
161 /* get number of silent chunks */
162 215 silent_chunks = 0;
163
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215 if (block_type == BLOCK_TYPE_INITIAL) {
164 uint32_t flags;
165
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1 if (buf_size < 4) {
166 av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
167 return AVERROR(EINVAL);
168 }
169 1 flags = AV_RB32(buf);
170 1 silent_chunks = av_popcount(flags);
171 1 buf += 4;
172 1 buf_size -= 4;
173
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214 } else if (block_type == BLOCK_TYPE_SILENCE) {
174 29 silent_chunks = 1;
175 29 buf_size = 0; // should already be zero but set it just to be sure
176 }
177
178 /* ensure output buffer is large enough */
179 215 audio_chunks = buf_size / s->chunk_size;
180
181 /* drop incomplete chunks */
182 215 buf_size = audio_chunks * s->chunk_size;
183
184
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215 if (silent_chunks + audio_chunks >= INT_MAX / avctx->block_align)
185 return AVERROR_INVALIDDATA;
186
187 /* get output buffer */
188 215 frame->nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) /
189 215 avctx->channels;
190
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215 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
191 return ret;
192 215 output_samples_u8 = frame->data[0];
193 215 output_samples_s16 = (int16_t *)frame->data[0];
194
195 /* decode silent chunks */
196
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215 if (silent_chunks > 0) {
197 29 int silent_size = avctx->block_align * silent_chunks;
198
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29 av_assert0(avctx->block_align * silent_chunks <= frame->nb_samples * avctx->channels);
199
200
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29 if (s->out_bps == 2) {
201 memset(output_samples_s16, 0x00, silent_size * 2);
202 output_samples_s16 += silent_size;
203 } else {
204 29 memset(output_samples_u8, 0x80, silent_size);
205 29 output_samples_u8 += silent_size;
206 }
207 }
208
209 /* decode audio chunks */
210
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215 if (audio_chunks > 0) {
211 186 buf_end = buf + buf_size;
212
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186 av_assert0((buf_size & (avctx->channels > 1)) == 0);
213
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399 while (buf_end - buf >= s->chunk_size) {
214
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213 if (s->out_bps == 2) {
215 decode_audio_s16(output_samples_s16, buf, s->chunk_size,
216 avctx->channels);
217 output_samples_s16 += avctx->block_align;
218 } else {
219 213 memcpy(output_samples_u8, buf, s->chunk_size);
220 213 output_samples_u8 += avctx->block_align;
221 }
222 213 buf += s->chunk_size;
223 }
224 }
225
226 215 *got_frame_ptr = 1;
227
228 215 return avpkt->size;
229 }
230
231 const AVCodec ff_vmdaudio_decoder = {
232 .name = "vmdaudio",
233 .long_name = NULL_IF_CONFIG_SMALL("Sierra VMD audio"),
234 .type = AVMEDIA_TYPE_AUDIO,
235 .id = AV_CODEC_ID_VMDAUDIO,
236 .priv_data_size = sizeof(VmdAudioContext),
237 .init = vmdaudio_decode_init,
238 .decode = vmdaudio_decode_frame,
239 .capabilities = AV_CODEC_CAP_DR1,
240 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
241 };
242