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https://github.com/nyanmisaka/ffmpeg-rockchip.git
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d7eabd5042
MPC8 allows indices of mpc_CC up to -1, and mpc_SCF up to -6, thus pad the tables by that much on the left end. Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org
105 lines
3.3 KiB
C
105 lines
3.3 KiB
C
/*
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* Musepack decoder core
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* Copyright (c) 2006 Konstantin Shishkov
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*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Musepack decoder core
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* MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
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* divided into 32 subbands.
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*/
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#include "avcodec.h"
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#include "get_bits.h"
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#include "dsputil.h"
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#include "mpegaudiodsp.h"
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#include "mpegaudio.h"
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#include "mpc.h"
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#include "mpcdata.h"
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void ff_mpc_init(void)
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{
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ff_mpa_synth_init_fixed(ff_mpa_synth_window_fixed);
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}
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/**
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* Process decoded Musepack data and produce PCM
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*/
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static void mpc_synth(MPCContext *c, int16_t *out, int channels)
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{
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int dither_state = 0;
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int i, ch;
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OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE], *samples_ptr;
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for(ch = 0; ch < channels; ch++){
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samples_ptr = samples + ch;
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for(i = 0; i < SAMPLES_PER_BAND; i++) {
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ff_mpa_synth_filter_fixed(&c->mpadsp,
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c->synth_buf[ch], &(c->synth_buf_offset[ch]),
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ff_mpa_synth_window_fixed, &dither_state,
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samples_ptr, channels,
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c->sb_samples[ch][i]);
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samples_ptr += 32 * channels;
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}
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}
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for(i = 0; i < MPC_FRAME_SIZE*channels; i++)
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*out++=samples[i];
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}
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void ff_mpc_dequantize_and_synth(MPCContext * c, int maxband, void *data, int channels)
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{
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int i, j, ch;
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Band *bands = c->bands;
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int off;
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float mul;
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/* dequantize */
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memset(c->sb_samples, 0, sizeof(c->sb_samples));
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off = 0;
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for(i = 0; i <= maxband; i++, off += SAMPLES_PER_BAND){
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for(ch = 0; ch < 2; ch++){
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if(bands[i].res[ch]){
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j = 0;
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mul = mpc_CC[bands[i].res[ch] + 1] * mpc_SCF[bands[i].scf_idx[ch][0]+6];
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for(; j < 12; j++)
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c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
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mul = mpc_CC[bands[i].res[ch] + 1] * mpc_SCF[bands[i].scf_idx[ch][1]+6];
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for(; j < 24; j++)
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c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
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mul = mpc_CC[bands[i].res[ch] + 1] * mpc_SCF[bands[i].scf_idx[ch][2]+6];
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for(; j < 36; j++)
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c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
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}
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}
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if(bands[i].msf){
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int t1, t2;
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for(j = 0; j < SAMPLES_PER_BAND; j++){
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t1 = c->sb_samples[0][j][i];
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t2 = c->sb_samples[1][j][i];
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c->sb_samples[0][j][i] = t1 + t2;
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c->sb_samples[1][j][i] = t1 - t2;
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}
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}
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}
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mpc_synth(c, data, channels);
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}
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