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
35
38
40
42 {
43 0, 512, 1024, 1536, 2052, 2564, 3076, 3588, 4100, 4612, 5124,
44 5636, 6164, 6676, 7224
45 };
46
47
49 {
50 int i, j;
54 static int vlc_initialized = 0;
55
59 static VLC_TYPE quant_tables[7224][2];
60
61 /* Musepack SV7 is always stereo */
65 }
66
69 return -1;
70 }
78
84 return -1;
85 }
92
95
96 if(vlc_initialized) return 0;
98 scfi_vlc.
table = scfi_table;
104 return -1;
105 }
106 dscf_vlc.
table = dscf_table;
112 return -1;
113 }
114 hdr_vlc.
table = hdr_table;
120 return -1;
121 }
123 for(j = 0; j < 2; j++){
125 quant_vlc[i][j].
table_allocated = quant_offsets[i*2 + j + 1] - quant_offsets[i*2 + j];
130 return -1;
131 }
132 }
133 }
134 vlc_initialized = 1;
135
136 return 0;
137 }
138
139 /**
140 * Fill samples for given subband
141 */
143 {
144 int i, i1, t;
145 switch(idx){
146 case -1:
149 }
150 break;
151 case 1:
153 for(i = 0; i < SAMPLES_PER_BAND/3; i++){
158 }
159 break;
160 case 2:
162 for(i = 0; i < SAMPLES_PER_BAND/2; i++){
166 }
167 break;
168 case 3: case 4: case 5: case 6: case 7:
172 break;
173 case 8: case 9: case 10: case 11: case 12:
174 case 13: case 14: case 15: case 16: case 17:
175 t = (1 << (idx - 2)) - 1;
178 break;
179 default: // case 0 and -2..-17
180 return;
181 }
182 }
183
185 {
187 if (t == 8)
189 return ref + t;
190 }
191
193 int *got_frame_ptr,
AVPacket *avpkt)
194 {
197 int buf_size;
200 int i, ch;
203 int off, ret, last_frame, skip;
204 int bits_used, bits_avail;
205
206 memset(bands, 0,
sizeof(*bands) * (c->
maxbands + 1));
207
208 buf_size = avpkt->
size & ~3;
209 if (buf_size <= 0) {
213 }
214 if (buf_size != avpkt->
size) {
216 "extra bytes at the end will be skipped.\n");
217 }
218
219 skip = buf[0];
220 last_frame = buf[1];
221 buf += 4;
222 buf_size -= 4;
223
224 /* get output buffer */
227 return ret;
228
233 buf_size >> 2);
236
237 /* read subband indexes */
239 for(ch = 0; ch < 2; ch++){
240 int t = 4;
243 else bands[i].
res[ch] = bands[i-1].
res[ch] + t;
244 if (bands[i].res[ch] < -1 || bands[i].res[ch] > 17) {
247 }
248 }
249
250 if(bands[i].res[0] || bands[i].res[1]){
251 mb = i;
253 }
254 }
255 /* get scale indexes coding method */
256 for(i = 0; i <=
mb; i++)
257 for(ch = 0; ch < 2; ch++)
259 /* get scale indexes */
260 for(i = 0; i <=
mb; i++){
261 for(ch = 0; ch < 2; ch++){
262 if(bands[i].res[ch]){
265 switch(bands[i].scfi[ch]){
266 case 0:
269 break;
270 case 1:
273 break;
274 case 2:
277 break;
278 case 3:
280 break;
281 }
283 }
284 }
285 }
286 /* get quantizers */
287 memset(c->
Q, 0,
sizeof(c->
Q));
288 off = 0;
290 for(ch = 0; ch < 2; ch++)
292
294 if(last_frame)
296
298 bits_avail = buf_size * 8;
299 if (!last_frame && ((bits_avail < bits_used) || (bits_used + 32 <= bits_avail))) {
300 av_log(avctx,
AV_LOG_ERROR,
"Error decoding frame: used %i of %i bits\n", bits_used, bits_avail);
301 return -1;
302 }
305 *got_frame_ptr = 0;
307 }
308
309 *got_frame_ptr = 1;
310
312 }
313
315 {
317
320 }
321
323 {
327 return 0;
328 }
329
343 };
void(* bswap_buf)(uint32_t *dst, const uint32_t *src, int w)
void ff_mpc_dequantize_and_synth(MPCContext *c, int maxband, int16_t **out, int channels)
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
This structure describes decoded (raw) audio or video data.
static const int8_t mpc7_idx32[]
ptrdiff_t const GLvoid * data
static void flush(AVCodecContext *avctx)
static const uint16_t mpc7_quant_vlc[MPC7_QUANT_VLC_TABLES][2][64 *2]
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
#define AV_LOG_WARNING
Something somehow does not look correct.
static void skip_bits_long(GetBitContext *s, int n)
static av_cold int init(AVCodecContext *avctx)
void av_fast_padded_malloc(void *ptr, unsigned int *size, size_t min_size)
Same behaviour av_fast_malloc but the buffer has additional AV_INPUT_BUFFER_PADDING_SIZE at the end w...
#define AV_CH_LAYOUT_STEREO
static const uint8_t mpc7_quant_vlc_sizes[MPC7_QUANT_VLC_TABLES *2]
static int mpc7_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
enum AVSampleFormat sample_fmt
audio sample format
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
static int get_bits_count(const GetBitContext *s)
bitstream reader API header.
static const int8_t mpc7_idx50[]
static const int8_t mpc7_idx31[]
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static const uint8_t mpc7_hdr[MPC7_HDR_SIZE *2]
#define MPC7_QUANT_VLC_TABLES
static const struct endianess table[]
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
static const uint8_t mpc7_quant_vlc_off[MPC7_QUANT_VLC_TABLES]
const char * name
Name of the codec implementation.
static const uint8_t mpc7_scfi[MPC7_SCFI_SIZE *2]
uint64_t channel_layout
Audio channel layout.
static av_cold int mpc7_decode_init(AVCodecContext *avctx)
common internal API header
audio channel layout utility functions
static void mpc7_decode_flush(AVCodecContext *avctx)
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
#define INIT_VLC_USE_NEW_STATIC
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Libavcodec external API header.
Musepack decoder MPEG Audio Layer 1/2 -like codec with frames of 1152 samples divided into 32 subband...
AVSampleFormat
Audio sample formats.
static const int8_t mpc7_idx51[]
av_cold void ff_mpc_init(void)
main external API structure.
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
static void idx_to_quant(MPCContext *c, GetBitContext *gb, int idx, int *dst)
Fill samples for given subband.
#define init_vlc(vlc, nb_bits, nb_codes,bits, bits_wrap, bits_size,codes, codes_wrap, codes_size,flags)
static unsigned int av_lfg_get(AVLFG *c)
Get the next random unsigned 32-bit number using an ALFG.
static unsigned int get_bits1(GetBitContext *s)
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
av_cold void av_lfg_init(AVLFG *c, unsigned int seed)
static av_cold int mpc7_decode_close(AVCodecContext *avctx)
static int decode(AVCodecContext *avctx, void *data, int *got_sub, AVPacket *avpkt)
common internal api header.
static const uint16_t quant_offsets[MPC7_QUANT_VLC_TABLES *2+1]
av_cold void ff_bswapdsp_init(BswapDSPContext *c)
static const int8_t mpc7_idx30[]
int channels
number of audio channels
VLC_TYPE(* table)[2]
code, bits
static enum AVSampleFormat sample_fmts[]
#define LOCAL_ALIGNED_16(t, v,...)
uint8_t ** extended_data
pointers to the data planes/channels.
static const uint8_t mpc7_dscf[MPC7_DSCF_SIZE *2]
static int get_scale_idx(GetBitContext *gb, int ref)
Subband structure - hold all variables for each subband.
This structure stores compressed data.
av_cold void ff_mpadsp_init(MPADSPContext *s)
int nb_samples
number of audio samples (per channel) described by this frame
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
static VLC quant_vlc[MPC7_QUANT_VLC_TABLES][2]