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author | Kostya Shishkov <kostya.shishkov@gmail.com> | 2006-12-24 04:51:43 +0000 |
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committer | Kostya Shishkov <kostya.shishkov@gmail.com> | 2006-12-24 04:51:43 +0000 |
commit | 185c7b6b2663ca22b5ae85e51b6ff5deacf5c038 (patch) | |
tree | 1f5550581c04ecfaff75909d37a362d60e925eac /libavcodec/mpc.c | |
parent | f8aa696f9fde35ea7d5a42ef3fb65a5771fe02cc (diff) | |
download | ffmpeg-185c7b6b2663ca22b5ae85e51b6ff5deacf5c038.tar.gz |
Musepack SV7 decoding support
Originally committed as revision 7375 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/mpc.c')
-rw-r--r-- | libavcodec/mpc.c | 346 |
1 files changed, 346 insertions, 0 deletions
diff --git a/libavcodec/mpc.c b/libavcodec/mpc.c new file mode 100644 index 0000000000..dcecb0ee5a --- /dev/null +++ b/libavcodec/mpc.c @@ -0,0 +1,346 @@ +/* + * 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; + /* 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); + + 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); + 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; + } + *data_size = (buf[1] ? c->lastframelen : MPC_FRAME_SIZE) * 4; + + return buf_size; +} + + +AVCodec mpc7_decoder = { + "mpc sv7", + CODEC_TYPE_AUDIO, + CODEC_ID_MUSEPACK7, + sizeof(MPCContext), + mpc7_decode_init, + NULL, + NULL, + mpc7_decode_frame, +}; |