Mercurial > libavcodec.hg
view mpc.c @ 4478:ca8c6efd00d3 libavcodec
cosmetic: indent sensibly so code can be read at all
author | mru |
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date | Mon, 05 Feb 2007 19:35:36 +0000 |
parents | a188a94e1b61 |
children | e3b224087a85 |
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/* * Musepack decoder * Copyright (c) 2006 Konstantin Shishkov * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA * */ /** * @file mpc.c Musepack decoder * MPEG Audio Layer 1/2 -like codec with frames of 1152 samples * divided into 32 subbands. */ #include "avcodec.h" #include "bitstream.h" #include "dsputil.h" #ifdef CONFIG_MPEGAUDIO_HP #define USE_HIGHPRECISION #endif #include "mpegaudio.h" #include "mpcdata.h" #define BANDS 32 #define SAMPLES_PER_BAND 36 #define MPC_FRAME_SIZE (BANDS * SAMPLES_PER_BAND) static VLC scfi_vlc, dscf_vlc, hdr_vlc, quant_vlc[MPC7_QUANT_VLC_TABLES][2]; static DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]); typedef struct { DSPContext dsp; int IS, MSS, gapless; int lastframelen, bands; int oldDSCF[2][BANDS]; int rnd; int frames_to_skip; /* for synthesis */ DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512*2]); int synth_buf_offset[MPA_MAX_CHANNELS]; DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][36][SBLIMIT]); } MPCContext; /** Subband structure - hold all variables for each subband */ typedef struct { int msf; ///< mid-stereo flag int res[2]; int scfi[2]; int scf_idx[2][3]; int Q[2]; }Band; static int mpc7_decode_init(AVCodecContext * avctx) { int i, j; MPCContext *c = avctx->priv_data; GetBitContext gb; uint8_t buf[16]; float f1=1.20050805774840750476 * 256; static int vlc_inited = 0; if(avctx->extradata_size < 16){ av_log(avctx, AV_LOG_ERROR, "Too small extradata size (%i)!\n", avctx->extradata_size); return -1; } memset(c->oldDSCF, 0, sizeof(c->oldDSCF)); c->rnd = 0xDEADBEEF; dsputil_init(&c->dsp, avctx); c->dsp.bswap_buf(buf, avctx->extradata, 4); ff_mpa_synth_init(mpa_window); init_get_bits(&gb, buf, 128); c->IS = get_bits1(&gb); c->MSS = get_bits1(&gb); c->bands = get_bits(&gb, 6); if(c->bands >= BANDS){ av_log(avctx, AV_LOG_ERROR, "Too many bands: %i\n", c->bands); return -1; } skip_bits(&gb, 88); c->gapless = get_bits1(&gb); c->lastframelen = get_bits(&gb, 11); av_log(avctx, AV_LOG_DEBUG, "IS: %d, MSS: %d, TG: %d, LFL: %d, bands: %d\n", c->IS, c->MSS, c->gapless, c->lastframelen, c->bands); c->frames_to_skip = 0; if(vlc_inited) return 0; av_log(avctx, AV_LOG_DEBUG, "Initing VLC\n"); if(init_vlc(&scfi_vlc, MPC7_SCFI_BITS, MPC7_SCFI_SIZE, &mpc7_scfi[1], 2, 1, &mpc7_scfi[0], 2, 1, INIT_VLC_USE_STATIC)){ av_log(avctx, AV_LOG_ERROR, "Cannot init SCFI VLC\n"); return -1; } if(init_vlc(&dscf_vlc, MPC7_DSCF_BITS, MPC7_DSCF_SIZE, &mpc7_dscf[1], 2, 1, &mpc7_dscf[0], 2, 1, INIT_VLC_USE_STATIC)){ av_log(avctx, AV_LOG_ERROR, "Cannot init DSCF VLC\n"); return -1; } if(init_vlc(&hdr_vlc, MPC7_HDR_BITS, MPC7_HDR_SIZE, &mpc7_hdr[1], 2, 1, &mpc7_hdr[0], 2, 1, INIT_VLC_USE_STATIC)){ av_log(avctx, AV_LOG_ERROR, "Cannot init HDR VLC\n"); return -1; } for(i = 0; i < MPC7_QUANT_VLC_TABLES; i++){ for(j = 0; j < 2; j++){ if(init_vlc(&quant_vlc[i][j], 9, mpc7_quant_vlc_sizes[i], &mpc7_quant_vlc[i][j][1], 4, 2, &mpc7_quant_vlc[i][j][0], 4, 2, INIT_VLC_USE_STATIC)){ av_log(avctx, AV_LOG_ERROR, "Cannot init QUANT VLC %i,%i\n",i,j); return -1; } } } vlc_inited = 1; return 0; } // XXX replace with something better static int av_always_inline mpc_rnd(MPCContext *c) { c->rnd = c->rnd * 27 + 17; return c->rnd; } /** * Process decoded Musepack data and produce PCM * @todo make it available for MPC8 and MPC6 */ static void mpc_synth(MPCContext *c, int16_t *out) { int dither_state = 0; int i, ch; OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE], *samples_ptr; for(ch = 0; ch < 2; ch++){ samples_ptr = samples + ch; for(i = 0; i < SAMPLES_PER_BAND; i++) { ff_mpa_synth_filter(c->synth_buf[ch], &(c->synth_buf_offset[ch]), mpa_window, &dither_state, samples_ptr, 2, c->sb_samples[ch][i]); samples_ptr += 64; } } for(i = 0; i < MPC_FRAME_SIZE*2; i++) *out++=samples[i]; } /** * Fill samples for given subband */ static void inline idx_to_quant(MPCContext *c, GetBitContext *gb, int idx, int *dst) { int i, i1, t; switch(idx){ case -1: for(i = 0; i < SAMPLES_PER_BAND; i++){ t = mpc_rnd(c); *dst++ = ((t>>24)& 0xFF) + ((t>>16) & 0xFF) + ((t>>8) & 0xFF) + (t & 0xFF) - 510; } case 1: i1 = get_bits1(gb); for(i = 0; i < SAMPLES_PER_BAND/3; i++){ t = get_vlc2(gb, quant_vlc[0][i1].table, 9, 2); *dst++ = mpc_idx30[t]; *dst++ = mpc_idx31[t]; *dst++ = mpc_idx32[t]; } break; case 2: i1 = get_bits1(gb); for(i = 0; i < SAMPLES_PER_BAND/2; i++){ t = get_vlc2(gb, quant_vlc[1][i1].table, 9, 2); *dst++ = mpc_idx50[t]; *dst++ = mpc_idx51[t]; } break; case 3: case 4: case 5: case 6: case 7: i1 = get_bits1(gb); for(i = 0; i < SAMPLES_PER_BAND; i++) *dst++ = get_vlc2(gb, quant_vlc[idx-1][i1].table, 9, 2) - mpc7_quant_vlc_off[idx-1]; break; case 8: case 9: case 10: case 11: case 12: case 13: case 14: case 15: case 16: case 17: t = (1 << (idx - 2)) - 1; for(i = 0; i < SAMPLES_PER_BAND; i++) *dst++ = get_bits(gb, idx - 1) - t; break; default: // case 0 and -2..-17 return; } } static int mpc7_decode_frame(AVCodecContext * avctx, void *data, int *data_size, uint8_t * buf, int buf_size) { MPCContext *c = avctx->priv_data; GetBitContext gb; uint8_t *bits; int i, j, ch, t; int mb = -1; Band bands[BANDS]; int Q[2][MPC_FRAME_SIZE]; int off; float mul; int bits_used, bits_avail; memset(bands, 0, sizeof(bands)); if(buf_size <= 4){ av_log(avctx, AV_LOG_ERROR, "Too small buffer passed (%i bytes)\n", buf_size); } bits = av_malloc(((buf_size - 1) & ~3) + FF_INPUT_BUFFER_PADDING_SIZE); c->dsp.bswap_buf(bits, buf + 4, (buf_size - 4) >> 2); init_get_bits(&gb, bits, (buf_size - 4)* 8); skip_bits(&gb, buf[0]); /* read subband indexes */ for(i = 0; i <= c->bands; i++){ for(ch = 0; ch < 2; ch++){ if(i) t = get_vlc2(&gb, hdr_vlc.table, MPC7_HDR_BITS, 1) - 5; if(!i || (t == 4)) bands[i].res[ch] = get_bits(&gb, 4); else bands[i].res[ch] = bands[i-1].res[ch] + t; } if(bands[i].res[0] || bands[i].res[1]){ mb = i; if(c->MSS) bands[i].msf = get_bits1(&gb); } } /* get scale indexes coding method */ for(i = 0; i <= mb; i++) for(ch = 0; ch < 2; ch++) if(bands[i].res[ch]) bands[i].scfi[ch] = get_vlc2(&gb, scfi_vlc.table, MPC7_SCFI_BITS, 1); /* get scale indexes */ for(i = 0; i <= mb; i++){ for(ch = 0; ch < 2; ch++){ if(bands[i].res[ch]){ bands[i].scf_idx[ch][2] = c->oldDSCF[ch][i]; t = get_vlc2(&gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7; bands[i].scf_idx[ch][0] = (t == 8) ? get_bits(&gb, 6) : (bands[i].scf_idx[ch][2] + t); switch(bands[i].scfi[ch]){ case 0: t = get_vlc2(&gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7; bands[i].scf_idx[ch][1] = (t == 8) ? get_bits(&gb, 6) : (bands[i].scf_idx[ch][0] + t); t = get_vlc2(&gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7; bands[i].scf_idx[ch][2] = (t == 8) ? get_bits(&gb, 6) : (bands[i].scf_idx[ch][1] + t); break; case 1: t = get_vlc2(&gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7; bands[i].scf_idx[ch][1] = (t == 8) ? get_bits(&gb, 6) : (bands[i].scf_idx[ch][0] + t); bands[i].scf_idx[ch][2] = bands[i].scf_idx[ch][1]; break; case 2: bands[i].scf_idx[ch][1] = bands[i].scf_idx[ch][0]; t = get_vlc2(&gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7; bands[i].scf_idx[ch][2] = (t == 8) ? get_bits(&gb, 6) : (bands[i].scf_idx[ch][1] + t); break; case 3: bands[i].scf_idx[ch][2] = bands[i].scf_idx[ch][1] = bands[i].scf_idx[ch][0]; break; } c->oldDSCF[ch][i] = bands[i].scf_idx[ch][2]; } } } /* get quantizers */ memset(Q, 0, sizeof(Q)); off = 0; for(i = 0; i < BANDS; i++, off += SAMPLES_PER_BAND) for(ch = 0; ch < 2; ch++) idx_to_quant(c, &gb, bands[i].res[ch], Q[ch] + off); /* dequantize */ memset(c->sb_samples, 0, sizeof(c->sb_samples)); off = 0; for(i = 0; i <= mb; i++, off += SAMPLES_PER_BAND){ for(ch = 0; ch < 2; ch++){ if(bands[i].res[ch]){ j = 0; mul = mpc_CC[bands[i].res[ch]] * mpc7_SCF[bands[i].scf_idx[ch][0]]; for(; j < 12; j++) c->sb_samples[ch][j][i] = mul * Q[ch][j + off]; mul = mpc_CC[bands[i].res[ch]] * mpc7_SCF[bands[i].scf_idx[ch][1]]; for(; j < 24; j++) c->sb_samples[ch][j][i] = mul * Q[ch][j + off]; mul = mpc_CC[bands[i].res[ch]] * mpc7_SCF[bands[i].scf_idx[ch][2]]; for(; j < 36; j++) c->sb_samples[ch][j][i] = mul * Q[ch][j + off]; } } if(bands[i].msf){ int t1, t2; for(j = 0; j < SAMPLES_PER_BAND; j++){ t1 = c->sb_samples[0][j][i]; t2 = c->sb_samples[1][j][i]; c->sb_samples[0][j][i] = t1 + t2; c->sb_samples[1][j][i] = t1 - t2; } } } mpc_synth(c, data); av_free(bits); bits_used = get_bits_count(&gb); bits_avail = (buf_size - 4) * 8; if(!buf[1] && ((bits_avail < bits_used) || (bits_used + 32 <= bits_avail))){ av_log(NULL,0, "Error decoding frame: used %i of %i bits\n", bits_used, bits_avail); return -1; } if(c->frames_to_skip){ c->frames_to_skip--; *data_size = 0; return buf_size; } *data_size = (buf[1] ? c->lastframelen : MPC_FRAME_SIZE) * 4; return buf_size; } static void mpc7_decode_flush(AVCodecContext *avctx) { MPCContext *c = avctx->priv_data; memset(c->oldDSCF, 0, sizeof(c->oldDSCF)); c->frames_to_skip = 32; } AVCodec mpc7_decoder = { "mpc sv7", CODEC_TYPE_AUDIO, CODEC_ID_MUSEPACK7, sizeof(MPCContext), mpc7_decode_init, NULL, NULL, mpc7_decode_frame, .flush = mpc7_decode_flush, };