1 /*
2 * Musepack SV7 decoder
3 * Copyright (c) 2006 Konstantin Shishkov
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 * MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
25 * divided into 32 subbands.
26 */
27
28 #include "libavutil/channel_layout.h"
29 #include "libavutil/internal.h"
30 #include "libavutil/lfg.h"
31 #include "libavutil/mem_internal.h"
32 #include "libavutil/thread.h"
33
34 #include "avcodec.h"
35 #include "get_bits.h"
36 #include "internal.h"
37 #include "mpegaudiodsp.h"
38
39 #include "mpc.h"
40 #include "mpc7data.h"
41
42 static VLC scfi_vlc, dscf_vlc, hdr_vlc, quant_vlc[MPC7_QUANT_VLC_TABLES][2];
43
mpc7_init_static(void)44 static av_cold void mpc7_init_static(void)
45 {
46 static VLC_TYPE quant_tables[7224][2];
47 const uint8_t *raw_quant_table = mpc7_quant_vlcs;
48
49 INIT_VLC_STATIC_FROM_LENGTHS(&scfi_vlc, MPC7_SCFI_BITS, MPC7_SCFI_SIZE,
50 &mpc7_scfi[1], 2,
51 &mpc7_scfi[0], 2, 1, 0, 0, 1 << MPC7_SCFI_BITS);
52 INIT_VLC_STATIC_FROM_LENGTHS(&dscf_vlc, MPC7_DSCF_BITS, MPC7_DSCF_SIZE,
53 &mpc7_dscf[1], 2,
54 &mpc7_dscf[0], 2, 1, -7, 0, 1 << MPC7_DSCF_BITS);
55 INIT_VLC_STATIC_FROM_LENGTHS(&hdr_vlc, MPC7_HDR_BITS, MPC7_HDR_SIZE,
56 &mpc7_hdr[1], 2,
57 &mpc7_hdr[0], 2, 1, -5, 0, 1 << MPC7_HDR_BITS);
58 for (unsigned i = 0, offset = 0; i < MPC7_QUANT_VLC_TABLES; i++){
59 for (int j = 0; j < 2; j++) {
60 quant_vlc[i][j].table = &quant_tables[offset];
61 quant_vlc[i][j].table_allocated = FF_ARRAY_ELEMS(quant_tables) - offset;
62 ff_init_vlc_from_lengths(&quant_vlc[i][j], 9, mpc7_quant_vlc_sizes[i],
63 &raw_quant_table[1], 2,
64 &raw_quant_table[0], 2, 1,
65 mpc7_quant_vlc_off[i],
66 INIT_VLC_STATIC_OVERLONG, NULL);
67 raw_quant_table += 2 * mpc7_quant_vlc_sizes[i];
68 offset += quant_vlc[i][j].table_size;
69 }
70 }
71 ff_mpa_synth_init_fixed();
72 }
73
mpc7_decode_init(AVCodecContext * avctx)74 static av_cold int mpc7_decode_init(AVCodecContext * avctx)
75 {
76 static AVOnce init_static_once = AV_ONCE_INIT;
77 MPCContext *c = avctx->priv_data;
78 GetBitContext gb;
79 LOCAL_ALIGNED_16(uint8_t, buf, [16]);
80
81 /* Musepack SV7 is always stereo */
82 if (avctx->channels != 2) {
83 avpriv_request_sample(avctx, "%d channels", avctx->channels);
84 return AVERROR_PATCHWELCOME;
85 }
86
87 if(avctx->extradata_size < 16){
88 av_log(avctx, AV_LOG_ERROR, "Too small extradata size (%i)!\n", avctx->extradata_size);
89 return AVERROR_INVALIDDATA;
90 }
91 memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
92 av_lfg_init(&c->rnd, 0xDEADBEEF);
93 ff_bswapdsp_init(&c->bdsp);
94 ff_mpadsp_init(&c->mpadsp);
95 c->bdsp.bswap_buf((uint32_t *) buf, (const uint32_t *) avctx->extradata, 4);
96 init_get_bits(&gb, buf, 128);
97
98 c->IS = get_bits1(&gb);
99 c->MSS = get_bits1(&gb);
100 c->maxbands = get_bits(&gb, 6);
101 if(c->maxbands >= BANDS){
102 av_log(avctx, AV_LOG_ERROR, "Too many bands: %i\n", c->maxbands);
103 return AVERROR_INVALIDDATA;
104 }
105 skip_bits_long(&gb, 88);
106 c->gapless = get_bits1(&gb);
107 c->lastframelen = get_bits(&gb, 11);
108 av_log(avctx, AV_LOG_DEBUG, "IS: %d, MSS: %d, TG: %d, LFL: %d, bands: %d\n",
109 c->IS, c->MSS, c->gapless, c->lastframelen, c->maxbands);
110 c->frames_to_skip = 0;
111
112 avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
113 avctx->channel_layout = AV_CH_LAYOUT_STEREO;
114
115 ff_thread_once(&init_static_once, mpc7_init_static);
116
117 return 0;
118 }
119
120 /**
121 * Fill samples for given subband
122 */
idx_to_quant(MPCContext * c,GetBitContext * gb,int idx,int * dst)123 static inline void idx_to_quant(MPCContext *c, GetBitContext *gb, int idx, int *dst)
124 {
125 int i, i1, t;
126 switch(idx){
127 case -1:
128 for(i = 0; i < SAMPLES_PER_BAND; i++){
129 *dst++ = (av_lfg_get(&c->rnd) & 0x3FC) - 510;
130 }
131 break;
132 case 1:
133 i1 = get_bits1(gb);
134 for(i = 0; i < SAMPLES_PER_BAND/3; i++){
135 t = get_vlc2(gb, quant_vlc[0][i1].table, 9, 2);
136 *dst++ = mpc7_idx30[t];
137 *dst++ = mpc7_idx31[t];
138 *dst++ = mpc7_idx32[t];
139 }
140 break;
141 case 2:
142 i1 = get_bits1(gb);
143 for(i = 0; i < SAMPLES_PER_BAND/2; i++){
144 t = get_vlc2(gb, quant_vlc[1][i1].table, 9, 2);
145 *dst++ = mpc7_idx50[t];
146 *dst++ = mpc7_idx51[t];
147 }
148 break;
149 case 3: case 4: case 5: case 6: case 7:
150 i1 = get_bits1(gb);
151 for(i = 0; i < SAMPLES_PER_BAND; i++)
152 *dst++ = get_vlc2(gb, quant_vlc[idx-1][i1].table, 9, 2);
153 break;
154 case 8: case 9: case 10: case 11: case 12:
155 case 13: case 14: case 15: case 16: case 17:
156 t = (1 << (idx - 2)) - 1;
157 for(i = 0; i < SAMPLES_PER_BAND; i++)
158 *dst++ = get_bits(gb, idx - 1) - t;
159 break;
160 default: // case 0 and -2..-17
161 return;
162 }
163 }
164
get_scale_idx(GetBitContext * gb,int ref)165 static int get_scale_idx(GetBitContext *gb, int ref)
166 {
167 int t = get_vlc2(gb, dscf_vlc.table, MPC7_DSCF_BITS, 1);
168 if (t == 8)
169 return get_bits(gb, 6);
170 return ref + t;
171 }
172
mpc7_decode_frame(AVCodecContext * avctx,void * data,int * got_frame_ptr,AVPacket * avpkt)173 static int mpc7_decode_frame(AVCodecContext * avctx, void *data,
174 int *got_frame_ptr, AVPacket *avpkt)
175 {
176 AVFrame *frame = data;
177 const uint8_t *buf = avpkt->data;
178 int buf_size;
179 MPCContext *c = avctx->priv_data;
180 GetBitContext gb;
181 int i, ch;
182 int mb = -1;
183 Band *bands = c->bands;
184 int off, ret, last_frame, skip;
185 int bits_used, bits_avail;
186
187 memset(bands, 0, sizeof(*bands) * (c->maxbands + 1));
188
189 buf_size = avpkt->size & ~3;
190 if (buf_size <= 0) {
191 av_log(avctx, AV_LOG_ERROR, "packet size is too small (%i bytes)\n",
192 avpkt->size);
193 return AVERROR_INVALIDDATA;
194 }
195 if (buf_size != avpkt->size) {
196 av_log(avctx, AV_LOG_WARNING, "packet size is not a multiple of 4. "
197 "extra bytes at the end will be skipped.\n");
198 }
199
200 skip = buf[0];
201 last_frame = buf[1];
202 buf += 4;
203 buf_size -= 4;
204
205 /* get output buffer */
206 frame->nb_samples = MPC_FRAME_SIZE;
207 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
208 return ret;
209
210 av_fast_padded_malloc(&c->bits, &c->buf_size, buf_size);
211 if (!c->bits)
212 return AVERROR(ENOMEM);
213 c->bdsp.bswap_buf((uint32_t *) c->bits, (const uint32_t *) buf,
214 buf_size >> 2);
215 if ((ret = init_get_bits8(&gb, c->bits, buf_size)) < 0)
216 return ret;
217 skip_bits_long(&gb, skip);
218
219 /* read subband indexes */
220 for(i = 0; i <= c->maxbands; i++){
221 for(ch = 0; ch < 2; ch++){
222 int t = i ? get_vlc2(&gb, hdr_vlc.table, MPC7_HDR_BITS, 1) : 4;
223 if(t == 4) bands[i].res[ch] = get_bits(&gb, 4);
224 else bands[i].res[ch] = bands[i-1].res[ch] + t;
225 if (bands[i].res[ch] < -1 || bands[i].res[ch] > 17) {
226 av_log(avctx, AV_LOG_ERROR, "subband index invalid\n");
227 return AVERROR_INVALIDDATA;
228 }
229 }
230
231 if(bands[i].res[0] || bands[i].res[1]){
232 mb = i;
233 if(c->MSS) bands[i].msf = get_bits1(&gb);
234 }
235 }
236 /* get scale indexes coding method */
237 for(i = 0; i <= mb; i++)
238 for(ch = 0; ch < 2; ch++)
239 if(bands[i].res[ch]) bands[i].scfi[ch] = get_vlc2(&gb, scfi_vlc.table, MPC7_SCFI_BITS, 1);
240 /* get scale indexes */
241 for(i = 0; i <= mb; i++){
242 for(ch = 0; ch < 2; ch++){
243 if(bands[i].res[ch]){
244 bands[i].scf_idx[ch][2] = c->oldDSCF[ch][i];
245 bands[i].scf_idx[ch][0] = get_scale_idx(&gb, bands[i].scf_idx[ch][2]);
246 switch(bands[i].scfi[ch]){
247 case 0:
248 bands[i].scf_idx[ch][1] = get_scale_idx(&gb, bands[i].scf_idx[ch][0]);
249 bands[i].scf_idx[ch][2] = get_scale_idx(&gb, bands[i].scf_idx[ch][1]);
250 break;
251 case 1:
252 bands[i].scf_idx[ch][1] = get_scale_idx(&gb, bands[i].scf_idx[ch][0]);
253 bands[i].scf_idx[ch][2] = bands[i].scf_idx[ch][1];
254 break;
255 case 2:
256 bands[i].scf_idx[ch][1] = bands[i].scf_idx[ch][0];
257 bands[i].scf_idx[ch][2] = get_scale_idx(&gb, bands[i].scf_idx[ch][1]);
258 break;
259 case 3:
260 bands[i].scf_idx[ch][2] = bands[i].scf_idx[ch][1] = bands[i].scf_idx[ch][0];
261 break;
262 }
263 c->oldDSCF[ch][i] = bands[i].scf_idx[ch][2];
264 }
265 }
266 }
267 /* get quantizers */
268 memset(c->Q, 0, sizeof(c->Q));
269 off = 0;
270 for(i = 0; i < BANDS; i++, off += SAMPLES_PER_BAND)
271 for(ch = 0; ch < 2; ch++)
272 idx_to_quant(c, &gb, bands[i].res[ch], c->Q[ch] + off);
273
274 ff_mpc_dequantize_and_synth(c, mb, (int16_t **)frame->extended_data, 2);
275 if(last_frame)
276 frame->nb_samples = c->lastframelen;
277
278 bits_used = get_bits_count(&gb);
279 bits_avail = buf_size * 8;
280 if (!last_frame && ((bits_avail < bits_used) || (bits_used + 32 <= bits_avail))) {
281 av_log(avctx, AV_LOG_ERROR, "Error decoding frame: used %i of %i bits\n", bits_used, bits_avail);
282 return AVERROR_INVALIDDATA;
283 }
284 if(c->frames_to_skip){
285 c->frames_to_skip--;
286 *got_frame_ptr = 0;
287 return avpkt->size;
288 }
289
290 *got_frame_ptr = 1;
291
292 return avpkt->size;
293 }
294
mpc7_decode_flush(AVCodecContext * avctx)295 static void mpc7_decode_flush(AVCodecContext *avctx)
296 {
297 MPCContext *c = avctx->priv_data;
298
299 memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
300 c->frames_to_skip = 32;
301 }
302
mpc7_decode_close(AVCodecContext * avctx)303 static av_cold int mpc7_decode_close(AVCodecContext *avctx)
304 {
305 MPCContext *c = avctx->priv_data;
306 av_freep(&c->bits);
307 c->buf_size = 0;
308 return 0;
309 }
310
311 AVCodec ff_mpc7_decoder = {
312 .name = "mpc7",
313 .long_name = NULL_IF_CONFIG_SMALL("Musepack SV7"),
314 .type = AVMEDIA_TYPE_AUDIO,
315 .id = AV_CODEC_ID_MUSEPACK7,
316 .priv_data_size = sizeof(MPCContext),
317 .init = mpc7_decode_init,
318 .close = mpc7_decode_close,
319 .decode = mpc7_decode_frame,
320 .flush = mpc7_decode_flush,
321 .capabilities = AV_CODEC_CAP_DR1,
322 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
323 AV_SAMPLE_FMT_NONE },
324 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
325 };
326