1/*
2 * Musepack decoder core
3 * Copyright (c) 2006 Konstantin Shishkov
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 * Musepack decoder core
25 * MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
26 * divided into 32 subbands.
27 */
28
29#include "avcodec.h"
30#include "get_bits.h"
31#include "dsputil.h"
32#include "mpegaudiodsp.h"
33#include "mpegaudio.h"
34
35#include "mpc.h"
36#include "mpcdata.h"
37
38void ff_mpc_init(void)
39{
40    ff_mpa_synth_init_fixed(ff_mpa_synth_window_fixed);
41}
42
43/**
44 * Process decoded Musepack data and produce PCM
45 */
46static void mpc_synth(MPCContext *c, int16_t *out, int channels)
47{
48    int dither_state = 0;
49    int i, ch;
50    OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE], *samples_ptr;
51
52    for(ch = 0;  ch < channels; ch++){
53        samples_ptr = samples + ch;
54        for(i = 0; i < SAMPLES_PER_BAND; i++) {
55            ff_mpa_synth_filter_fixed(&c->mpadsp,
56                                c->synth_buf[ch], &(c->synth_buf_offset[ch]),
57                                ff_mpa_synth_window_fixed, &dither_state,
58                                samples_ptr, channels,
59                                c->sb_samples[ch][i]);
60            samples_ptr += 32 * channels;
61        }
62    }
63    for(i = 0; i < MPC_FRAME_SIZE*channels; i++)
64        *out++=samples[i];
65}
66
67void ff_mpc_dequantize_and_synth(MPCContext * c, int maxband, void *data, int channels)
68{
69    int i, j, ch;
70    Band *bands = c->bands;
71    int off;
72    float mul;
73
74    /* dequantize */
75    memset(c->sb_samples, 0, sizeof(c->sb_samples));
76    off = 0;
77    for(i = 0; i <= maxband; i++, off += SAMPLES_PER_BAND){
78        for(ch = 0; ch < 2; ch++){
79            if(bands[i].res[ch]){
80                j = 0;
81                mul = mpc_CC[bands[i].res[ch] + 1] * mpc_SCF[bands[i].scf_idx[ch][0]+6];
82                for(; j < 12; j++)
83                    c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
84                mul = mpc_CC[bands[i].res[ch] + 1] * mpc_SCF[bands[i].scf_idx[ch][1]+6];
85                for(; j < 24; j++)
86                    c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
87                mul = mpc_CC[bands[i].res[ch] + 1] * mpc_SCF[bands[i].scf_idx[ch][2]+6];
88                for(; j < 36; j++)
89                    c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
90            }
91        }
92        if(bands[i].msf){
93            int t1, t2;
94            for(j = 0; j < SAMPLES_PER_BAND; j++){
95                t1 = c->sb_samples[0][j][i];
96                t2 = c->sb_samples[1][j][i];
97                c->sb_samples[0][j][i] = t1 + t2;
98                c->sb_samples[1][j][i] = t1 - t2;
99            }
100        }
101    }
102
103    mpc_synth(c, data, channels);
104}
105