Move some mpegaudio functions to new mpegaudiodsp subsystem
[libav.git] / libavcodec / mpc.c
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
38 void 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 */
46 static 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
67 void 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]] * mpc_SCF[bands[i].scf_idx[ch][0]];
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]] * mpc_SCF[bands[i].scf_idx[ch][1]];
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]] * mpc_SCF[bands[i].scf_idx[ch][2]];
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 }