mirror of
https://github.com/FFmpeg/FFmpeg.git
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356 lines
12 KiB
C
356 lines
12 KiB
C
/*
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* Musepack decoder
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* Copyright (c) 2006 Konstantin Shishkov
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg 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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* FFmpeg 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 FFmpeg; 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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/**
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* @file mpc.c Musepack decoder
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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 "bitstream.h"
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#include "dsputil.h"
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#include "random.h"
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#ifdef CONFIG_MPEGAUDIO_HP
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#define USE_HIGHPRECISION
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#endif
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#include "mpegaudio.h"
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#include "mpcdata.h"
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#define BANDS 32
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#define SAMPLES_PER_BAND 36
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#define MPC_FRAME_SIZE (BANDS * SAMPLES_PER_BAND)
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static VLC scfi_vlc, dscf_vlc, hdr_vlc, quant_vlc[MPC7_QUANT_VLC_TABLES][2];
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static DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]);
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typedef struct {
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DSPContext dsp;
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int IS, MSS, gapless;
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int lastframelen, bands;
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int oldDSCF[2][BANDS];
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AVRandomState rnd;
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int frames_to_skip;
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/* for synthesis */
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DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512*2]);
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int synth_buf_offset[MPA_MAX_CHANNELS];
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DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][36][SBLIMIT]);
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} MPCContext;
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/** Subband structure - hold all variables for each subband */
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typedef struct {
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int msf; ///< mid-stereo flag
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int res[2];
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int scfi[2];
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int scf_idx[2][3];
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int Q[2];
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}Band;
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static int mpc7_decode_init(AVCodecContext * avctx)
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{
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int i, j;
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MPCContext *c = avctx->priv_data;
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GetBitContext gb;
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uint8_t buf[16];
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float f1=1.20050805774840750476 * 256;
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static int vlc_inited = 0;
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if(avctx->extradata_size < 16){
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av_log(avctx, AV_LOG_ERROR, "Too small extradata size (%i)!\n", avctx->extradata_size);
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return -1;
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}
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memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
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av_init_random(0xDEADBEEF, &c->rnd);
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dsputil_init(&c->dsp, avctx);
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c->dsp.bswap_buf(buf, avctx->extradata, 4);
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ff_mpa_synth_init(mpa_window);
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init_get_bits(&gb, buf, 128);
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c->IS = get_bits1(&gb);
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c->MSS = get_bits1(&gb);
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c->bands = get_bits(&gb, 6);
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if(c->bands >= BANDS){
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av_log(avctx, AV_LOG_ERROR, "Too many bands: %i\n", c->bands);
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return -1;
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}
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skip_bits(&gb, 88);
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c->gapless = get_bits1(&gb);
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c->lastframelen = get_bits(&gb, 11);
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av_log(avctx, AV_LOG_DEBUG, "IS: %d, MSS: %d, TG: %d, LFL: %d, bands: %d\n",
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c->IS, c->MSS, c->gapless, c->lastframelen, c->bands);
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c->frames_to_skip = 0;
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if(vlc_inited) return 0;
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av_log(avctx, AV_LOG_DEBUG, "Initing VLC\n");
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if(init_vlc(&scfi_vlc, MPC7_SCFI_BITS, MPC7_SCFI_SIZE,
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&mpc7_scfi[1], 2, 1,
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&mpc7_scfi[0], 2, 1, INIT_VLC_USE_STATIC)){
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av_log(avctx, AV_LOG_ERROR, "Cannot init SCFI VLC\n");
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return -1;
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}
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if(init_vlc(&dscf_vlc, MPC7_DSCF_BITS, MPC7_DSCF_SIZE,
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&mpc7_dscf[1], 2, 1,
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&mpc7_dscf[0], 2, 1, INIT_VLC_USE_STATIC)){
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av_log(avctx, AV_LOG_ERROR, "Cannot init DSCF VLC\n");
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return -1;
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}
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if(init_vlc(&hdr_vlc, MPC7_HDR_BITS, MPC7_HDR_SIZE,
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&mpc7_hdr[1], 2, 1,
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&mpc7_hdr[0], 2, 1, INIT_VLC_USE_STATIC)){
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av_log(avctx, AV_LOG_ERROR, "Cannot init HDR VLC\n");
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return -1;
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}
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for(i = 0; i < MPC7_QUANT_VLC_TABLES; i++){
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for(j = 0; j < 2; j++){
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if(init_vlc(&quant_vlc[i][j], 9, mpc7_quant_vlc_sizes[i],
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&mpc7_quant_vlc[i][j][1], 4, 2,
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&mpc7_quant_vlc[i][j][0], 4, 2, INIT_VLC_USE_STATIC)){
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av_log(avctx, AV_LOG_ERROR, "Cannot init QUANT VLC %i,%i\n",i,j);
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return -1;
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}
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}
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}
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vlc_inited = 1;
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return 0;
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}
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/**
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* Process decoded Musepack data and produce PCM
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* @todo make it available for MPC8 and MPC6
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*/
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static void mpc_synth(MPCContext *c, int16_t *out)
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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 < 2; 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(c->synth_buf[ch], &(c->synth_buf_offset[ch]),
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mpa_window, &dither_state,
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samples_ptr, 2,
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c->sb_samples[ch][i]);
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samples_ptr += 64;
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}
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}
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for(i = 0; i < MPC_FRAME_SIZE*2; i++)
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*out++=samples[i];
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}
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/**
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* Fill samples for given subband
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*/
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static inline void idx_to_quant(MPCContext *c, GetBitContext *gb, int idx, int *dst)
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{
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int i, i1, t;
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switch(idx){
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case -1:
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for(i = 0; i < SAMPLES_PER_BAND; i++){
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*dst++ = (av_random(&c->rnd) & 0x3FC) - 510;
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}
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break;
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case 1:
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i1 = get_bits1(gb);
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for(i = 0; i < SAMPLES_PER_BAND/3; i++){
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t = get_vlc2(gb, quant_vlc[0][i1].table, 9, 2);
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*dst++ = mpc_idx30[t];
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*dst++ = mpc_idx31[t];
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*dst++ = mpc_idx32[t];
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}
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break;
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case 2:
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i1 = get_bits1(gb);
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for(i = 0; i < SAMPLES_PER_BAND/2; i++){
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t = get_vlc2(gb, quant_vlc[1][i1].table, 9, 2);
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*dst++ = mpc_idx50[t];
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*dst++ = mpc_idx51[t];
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}
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break;
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case 3: case 4: case 5: case 6: case 7:
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i1 = get_bits1(gb);
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for(i = 0; i < SAMPLES_PER_BAND; i++)
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*dst++ = get_vlc2(gb, quant_vlc[idx-1][i1].table, 9, 2) - mpc7_quant_vlc_off[idx-1];
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break;
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case 8: case 9: case 10: case 11: case 12:
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case 13: case 14: case 15: case 16: case 17:
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t = (1 << (idx - 2)) - 1;
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for(i = 0; i < SAMPLES_PER_BAND; i++)
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*dst++ = get_bits(gb, idx - 1) - t;
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break;
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default: // case 0 and -2..-17
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return;
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}
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}
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static int mpc7_decode_frame(AVCodecContext * avctx,
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void *data, int *data_size,
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uint8_t * buf, int buf_size)
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{
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MPCContext *c = avctx->priv_data;
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GetBitContext gb;
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uint8_t *bits;
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int i, j, ch, t;
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int mb = -1;
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Band bands[BANDS];
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int Q[2][MPC_FRAME_SIZE];
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int off;
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float mul;
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int bits_used, bits_avail;
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memset(bands, 0, sizeof(bands));
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if(buf_size <= 4){
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av_log(avctx, AV_LOG_ERROR, "Too small buffer passed (%i bytes)\n", buf_size);
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}
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bits = av_malloc(((buf_size - 1) & ~3) + FF_INPUT_BUFFER_PADDING_SIZE);
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c->dsp.bswap_buf(bits, buf + 4, (buf_size - 4) >> 2);
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init_get_bits(&gb, bits, (buf_size - 4)* 8);
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skip_bits(&gb, buf[0]);
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/* read subband indexes */
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for(i = 0; i <= c->bands; i++){
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for(ch = 0; ch < 2; ch++){
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if(i) t = get_vlc2(&gb, hdr_vlc.table, MPC7_HDR_BITS, 1) - 5;
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if(!i || (t == 4)) bands[i].res[ch] = get_bits(&gb, 4);
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else bands[i].res[ch] = bands[i-1].res[ch] + t;
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}
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if(bands[i].res[0] || bands[i].res[1]){
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mb = i;
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if(c->MSS) bands[i].msf = get_bits1(&gb);
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}
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}
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/* get scale indexes coding method */
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for(i = 0; i <= mb; i++)
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for(ch = 0; ch < 2; ch++)
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if(bands[i].res[ch]) bands[i].scfi[ch] = get_vlc2(&gb, scfi_vlc.table, MPC7_SCFI_BITS, 1);
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/* get scale indexes */
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for(i = 0; i <= mb; i++){
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for(ch = 0; ch < 2; ch++){
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if(bands[i].res[ch]){
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bands[i].scf_idx[ch][2] = c->oldDSCF[ch][i];
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t = get_vlc2(&gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7;
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bands[i].scf_idx[ch][0] = (t == 8) ? get_bits(&gb, 6) : (bands[i].scf_idx[ch][2] + t);
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switch(bands[i].scfi[ch]){
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case 0:
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t = get_vlc2(&gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7;
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bands[i].scf_idx[ch][1] = (t == 8) ? get_bits(&gb, 6) : (bands[i].scf_idx[ch][0] + t);
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t = get_vlc2(&gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7;
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bands[i].scf_idx[ch][2] = (t == 8) ? get_bits(&gb, 6) : (bands[i].scf_idx[ch][1] + t);
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break;
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case 1:
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t = get_vlc2(&gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7;
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bands[i].scf_idx[ch][1] = (t == 8) ? get_bits(&gb, 6) : (bands[i].scf_idx[ch][0] + t);
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bands[i].scf_idx[ch][2] = bands[i].scf_idx[ch][1];
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break;
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case 2:
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bands[i].scf_idx[ch][1] = bands[i].scf_idx[ch][0];
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t = get_vlc2(&gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7;
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bands[i].scf_idx[ch][2] = (t == 8) ? get_bits(&gb, 6) : (bands[i].scf_idx[ch][1] + t);
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break;
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case 3:
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bands[i].scf_idx[ch][2] = bands[i].scf_idx[ch][1] = bands[i].scf_idx[ch][0];
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break;
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}
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c->oldDSCF[ch][i] = bands[i].scf_idx[ch][2];
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}
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}
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}
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/* get quantizers */
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memset(Q, 0, sizeof(Q));
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off = 0;
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for(i = 0; i < BANDS; i++, off += SAMPLES_PER_BAND)
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for(ch = 0; ch < 2; ch++)
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idx_to_quant(c, &gb, bands[i].res[ch], Q[ch] + off);
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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 <= mb; 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]] * mpc7_SCF[bands[i].scf_idx[ch][0]];
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for(; j < 12; j++)
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c->sb_samples[ch][j][i] = mul * Q[ch][j + off];
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mul = mpc_CC[bands[i].res[ch]] * mpc7_SCF[bands[i].scf_idx[ch][1]];
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for(; j < 24; j++)
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c->sb_samples[ch][j][i] = mul * Q[ch][j + off];
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mul = mpc_CC[bands[i].res[ch]] * mpc7_SCF[bands[i].scf_idx[ch][2]];
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for(; j < 36; j++)
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c->sb_samples[ch][j][i] = mul * 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);
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av_free(bits);
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bits_used = get_bits_count(&gb);
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bits_avail = (buf_size - 4) * 8;
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if(!buf[1] && ((bits_avail < bits_used) || (bits_used + 32 <= bits_avail))){
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av_log(NULL,0, "Error decoding frame: used %i of %i bits\n", bits_used, bits_avail);
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return -1;
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}
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if(c->frames_to_skip){
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c->frames_to_skip--;
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*data_size = 0;
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return buf_size;
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}
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*data_size = (buf[1] ? c->lastframelen : MPC_FRAME_SIZE) * 4;
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return buf_size;
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}
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static void mpc7_decode_flush(AVCodecContext *avctx)
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{
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MPCContext *c = avctx->priv_data;
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memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
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c->frames_to_skip = 32;
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}
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AVCodec mpc7_decoder = {
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"mpc sv7",
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CODEC_TYPE_AUDIO,
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CODEC_ID_MUSEPACK7,
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sizeof(MPCContext),
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mpc7_decode_init,
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NULL,
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NULL,
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mpc7_decode_frame,
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.flush = mpc7_decode_flush,
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};
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