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ffmpeg/libavcodec/mpc8.c
Michael Niedermayer 80d156d7fd Merge remote-tracking branch 'qatar/master' 
* qatar/master:
  qdm2: Use floating point synthesis filter.
  h264: correct border check.
  h264: fix loopfilter with threading at slice boundaries.
  Fix ff_mpa_synth_filter_fixed() prototype
  Rename costablegen.c ---> cos_tablegen.c.
  Collapse tableprint.c into tableprint.h.
  Simplify trig table rules
  Remove potentially unstable filenames from comments in generated files.
  Ignore generated tables and generated table generator programs.
  Simplify CLEANFILES make variable by using wildcards.
  Remove silly insults from avformat_version() Doxygen documentation.
  mpegaudiodsp: fix x86 and ppc makefiles
  configure: Adjust AVX assembler check.
  mpegaudio: remove unused version of SAME_HEADER_MASK
  mpegaudio: remove useless #undef at end of file
  asfdec: add missing #include for av_bswap32()
  mpegaudio: merge two #if CONFIG_FLOAT blocks
  mpegaudio: move some struct definitions from mpegaudio.h
  Move some mpegaudio functions to new mpegaudiodsp subsystem

Conflicts:
	libavcodec/h264.c
	libavcodec/x86/Makefile

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-05-20 05:48:22 +02:00

420 lines
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/*
* Musepack SV8 decoder
* Copyright (c) 2007 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
* MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
* divided into 32 subbands.
*/
#include "libavutil/lfg.h"
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
#include "mpegaudiodsp.h"
#include "libavutil/audioconvert.h"
#include "mpc.h"
#include "mpcdata.h"
#include "mpc8data.h"
#include "mpc8huff.h"
static VLC band_vlc, scfi_vlc[2], dscf_vlc[2], res_vlc[2];
static VLC q1_vlc, q2_vlc[2], q3_vlc[2], quant_vlc[4][2], q9up_vlc;
static const int q3_offsets[2] = { MPC8_Q3_OFFSET, MPC8_Q4_OFFSET };
static const int quant_offsets[6] = { MPC8_Q5_OFFSET, MPC8_Q6_OFFSET, MPC8_Q7_OFFSET, MPC8_Q8_OFFSET };
static inline int mpc8_dec_base(GetBitContext *gb, int k, int n)
{
int len = mpc8_cnk_len[k-1][n-1] - 1;
int code = len ? get_bits_long(gb, len) : 0;
if (code >= mpc8_cnk_lost[k-1][n-1])
code = ((code << 1) | get_bits1(gb)) - mpc8_cnk_lost[k-1][n-1];
return code;
}
static inline int mpc8_dec_enum(GetBitContext *gb, int k, int n)
{
int bits = 0;
const uint32_t * C = mpc8_cnk[k-1];
int code = mpc8_dec_base(gb, k, n);
do {
n--;
if (code >= C[n]) {
bits |= 1 << n;
code -= C[n];
C -= 32;
k--;
}
} while(k > 0);
return bits;
}
static inline int mpc8_get_mod_golomb(GetBitContext *gb, int m)
{
if(mpc8_cnk_len[0][m] < 1) return 0;
return mpc8_dec_base(gb, 1, m+1);
}
static int mpc8_get_mask(GetBitContext *gb, int size, int t)
{
int mask = 0;
if(t && t != size)
mask = mpc8_dec_enum(gb, FFMIN(t, size - t), size);
if((t << 1) > size) mask = ~mask;
return mask;
}
static const uint16_t vlc_offsets[13] = {
0, 640, 1184, 1748, 2298, 2426, 2554, 3066, 3578, 4106, 4618, 5196, 5708
};
static av_cold int mpc8_decode_init(AVCodecContext * avctx)
{
int i;
MPCContext *c = avctx->priv_data;
GetBitContext gb;
static int vlc_initialized = 0;
int channels;
static VLC_TYPE band_table[542][2];
static VLC_TYPE q1_table[520][2];
static VLC_TYPE q9up_table[524][2];
static VLC_TYPE scfi0_table[1 << MPC8_SCFI0_BITS][2];
static VLC_TYPE scfi1_table[1 << MPC8_SCFI1_BITS][2];
static VLC_TYPE dscf0_table[560][2];
static VLC_TYPE dscf1_table[598][2];
static VLC_TYPE q3_0_table[512][2];
static VLC_TYPE q3_1_table[516][2];
static VLC_TYPE codes_table[5708][2];
if(avctx->extradata_size < 2){
av_log(avctx, AV_LOG_ERROR, "Too small extradata size (%i)!\n", avctx->extradata_size);
return -1;
}
memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
av_lfg_init(&c->rnd, 0xDEADBEEF);
dsputil_init(&c->dsp, avctx);
ff_mpadsp_init(&c->mpadsp);
ff_mpc_init();
init_get_bits(&gb, avctx->extradata, 16);
skip_bits(&gb, 3);//sample rate
c->maxbands = get_bits(&gb, 5) + 1;
channels = get_bits(&gb, 4) + 1;
if (channels > 2) {
av_log_missing_feature(avctx, "Multichannel MPC SV8", 1);
return -1;
}
c->MSS = get_bits1(&gb);
c->frames = 1 << (get_bits(&gb, 3) * 2);
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
if(vlc_initialized) return 0;
av_log(avctx, AV_LOG_DEBUG, "Initing VLC\n");
band_vlc.table = band_table;
band_vlc.table_allocated = 542;
init_vlc(&band_vlc, MPC8_BANDS_BITS, MPC8_BANDS_SIZE,
mpc8_bands_bits, 1, 1,
mpc8_bands_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
q1_vlc.table = q1_table;
q1_vlc.table_allocated = 520;
init_vlc(&q1_vlc, MPC8_Q1_BITS, MPC8_Q1_SIZE,
mpc8_q1_bits, 1, 1,
mpc8_q1_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
q9up_vlc.table = q9up_table;
q9up_vlc.table_allocated = 524;
init_vlc(&q9up_vlc, MPC8_Q9UP_BITS, MPC8_Q9UP_SIZE,
mpc8_q9up_bits, 1, 1,
mpc8_q9up_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
scfi_vlc[0].table = scfi0_table;
scfi_vlc[0].table_allocated = 1 << MPC8_SCFI0_BITS;
init_vlc(&scfi_vlc[0], MPC8_SCFI0_BITS, MPC8_SCFI0_SIZE,
mpc8_scfi0_bits, 1, 1,
mpc8_scfi0_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
scfi_vlc[1].table = scfi1_table;
scfi_vlc[1].table_allocated = 1 << MPC8_SCFI1_BITS;
init_vlc(&scfi_vlc[1], MPC8_SCFI1_BITS, MPC8_SCFI1_SIZE,
mpc8_scfi1_bits, 1, 1,
mpc8_scfi1_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
dscf_vlc[0].table = dscf0_table;
dscf_vlc[0].table_allocated = 560;
init_vlc(&dscf_vlc[0], MPC8_DSCF0_BITS, MPC8_DSCF0_SIZE,
mpc8_dscf0_bits, 1, 1,
mpc8_dscf0_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
dscf_vlc[1].table = dscf1_table;
dscf_vlc[1].table_allocated = 598;
init_vlc(&dscf_vlc[1], MPC8_DSCF1_BITS, MPC8_DSCF1_SIZE,
mpc8_dscf1_bits, 1, 1,
mpc8_dscf1_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
q3_vlc[0].table = q3_0_table;
q3_vlc[0].table_allocated = 512;
init_vlc_sparse(&q3_vlc[0], MPC8_Q3_BITS, MPC8_Q3_SIZE,
mpc8_q3_bits, 1, 1,
mpc8_q3_codes, 1, 1,
mpc8_q3_syms, 1, 1, INIT_VLC_USE_NEW_STATIC);
q3_vlc[1].table = q3_1_table;
q3_vlc[1].table_allocated = 516;
init_vlc_sparse(&q3_vlc[1], MPC8_Q4_BITS, MPC8_Q4_SIZE,
mpc8_q4_bits, 1, 1,
mpc8_q4_codes, 1, 1,
mpc8_q4_syms, 1, 1, INIT_VLC_USE_NEW_STATIC);
for(i = 0; i < 2; i++){
res_vlc[i].table = &codes_table[vlc_offsets[0+i]];
res_vlc[i].table_allocated = vlc_offsets[1+i] - vlc_offsets[0+i];
init_vlc(&res_vlc[i], MPC8_RES_BITS, MPC8_RES_SIZE,
&mpc8_res_bits[i], 1, 1,
&mpc8_res_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
q2_vlc[i].table = &codes_table[vlc_offsets[2+i]];
q2_vlc[i].table_allocated = vlc_offsets[3+i] - vlc_offsets[2+i];
init_vlc(&q2_vlc[i], MPC8_Q2_BITS, MPC8_Q2_SIZE,
&mpc8_q2_bits[i], 1, 1,
&mpc8_q2_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
quant_vlc[0][i].table = &codes_table[vlc_offsets[4+i]];
quant_vlc[0][i].table_allocated = vlc_offsets[5+i] - vlc_offsets[4+i];
init_vlc(&quant_vlc[0][i], MPC8_Q5_BITS, MPC8_Q5_SIZE,
&mpc8_q5_bits[i], 1, 1,
&mpc8_q5_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
quant_vlc[1][i].table = &codes_table[vlc_offsets[6+i]];
quant_vlc[1][i].table_allocated = vlc_offsets[7+i] - vlc_offsets[6+i];
init_vlc(&quant_vlc[1][i], MPC8_Q6_BITS, MPC8_Q6_SIZE,
&mpc8_q6_bits[i], 1, 1,
&mpc8_q6_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
quant_vlc[2][i].table = &codes_table[vlc_offsets[8+i]];
quant_vlc[2][i].table_allocated = vlc_offsets[9+i] - vlc_offsets[8+i];
init_vlc(&quant_vlc[2][i], MPC8_Q7_BITS, MPC8_Q7_SIZE,
&mpc8_q7_bits[i], 1, 1,
&mpc8_q7_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
quant_vlc[3][i].table = &codes_table[vlc_offsets[10+i]];
quant_vlc[3][i].table_allocated = vlc_offsets[11+i] - vlc_offsets[10+i];
init_vlc(&quant_vlc[3][i], MPC8_Q8_BITS, MPC8_Q8_SIZE,
&mpc8_q8_bits[i], 1, 1,
&mpc8_q8_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
}
vlc_initialized = 1;
return 0;
}
static int mpc8_decode_frame(AVCodecContext * avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
MPCContext *c = avctx->priv_data;
GetBitContext gb2, *gb = &gb2;
int i, j, k, ch, cnt, res, t;
Band *bands = c->bands;
int off;
int maxband, keyframe;
int last[2];
keyframe = c->cur_frame == 0;
if(keyframe){
memset(c->Q, 0, sizeof(c->Q));
c->last_bits_used = 0;
}
init_get_bits(gb, buf, buf_size * 8);
skip_bits(gb, c->last_bits_used & 7);
if(keyframe)
maxband = mpc8_get_mod_golomb(gb, c->maxbands + 1);
else{
maxband = c->last_max_band + get_vlc2(gb, band_vlc.table, MPC8_BANDS_BITS, 2);
if(maxband > 32) maxband -= 33;
}
c->last_max_band = maxband;
/* read subband indexes */
if(maxband){
last[0] = last[1] = 0;
for(i = maxband - 1; i >= 0; i--){
for(ch = 0; ch < 2; ch++){
last[ch] = get_vlc2(gb, res_vlc[last[ch] > 2].table, MPC8_RES_BITS, 2) + last[ch];
if(last[ch] > 15) last[ch] -= 17;
bands[i].res[ch] = last[ch];
}
}
if(c->MSS){
int mask;
cnt = 0;
for(i = 0; i < maxband; i++)
if(bands[i].res[0] || bands[i].res[1])
cnt++;
t = mpc8_get_mod_golomb(gb, cnt);
mask = mpc8_get_mask(gb, cnt, t);
for(i = maxband - 1; i >= 0; i--)
if(bands[i].res[0] || bands[i].res[1]){
bands[i].msf = mask & 1;
mask >>= 1;
}
}
}
for(i = maxband; i < c->maxbands; i++)
bands[i].res[0] = bands[i].res[1] = 0;
if(keyframe){
for(i = 0; i < 32; i++)
c->oldDSCF[0][i] = c->oldDSCF[1][i] = 1;
}
for(i = 0; i < maxband; i++){
if(bands[i].res[0] || bands[i].res[1]){
cnt = !!bands[i].res[0] + !!bands[i].res[1] - 1;
if(cnt >= 0){
t = get_vlc2(gb, scfi_vlc[cnt].table, scfi_vlc[cnt].bits, 1);
if(bands[i].res[0]) bands[i].scfi[0] = t >> (2 * cnt);
if(bands[i].res[1]) bands[i].scfi[1] = t & 3;
}
}
}
for(i = 0; i < maxband; i++){
for(ch = 0; ch < 2; ch++){
if(!bands[i].res[ch]) continue;
if(c->oldDSCF[ch][i]){
bands[i].scf_idx[ch][0] = get_bits(gb, 7) - 6;
c->oldDSCF[ch][i] = 0;
}else{
t = get_vlc2(gb, dscf_vlc[1].table, MPC8_DSCF1_BITS, 2);
if(t == 64)
t += get_bits(gb, 6);
bands[i].scf_idx[ch][0] = ((bands[i].scf_idx[ch][2] + t - 25) & 0x7F) - 6;
}
for(j = 0; j < 2; j++){
if((bands[i].scfi[ch] << j) & 2)
bands[i].scf_idx[ch][j + 1] = bands[i].scf_idx[ch][j];
else{
t = get_vlc2(gb, dscf_vlc[0].table, MPC8_DSCF0_BITS, 2);
if(t == 31)
t = 64 + get_bits(gb, 6);
bands[i].scf_idx[ch][j + 1] = ((bands[i].scf_idx[ch][j] + t - 25) & 0x7F) - 6;
}
}
}
}
for(i = 0, off = 0; i < maxband; i++, off += SAMPLES_PER_BAND){
for(ch = 0; ch < 2; ch++){
res = bands[i].res[ch];
switch(res){
case -1:
for(j = 0; j < SAMPLES_PER_BAND; j++)
c->Q[ch][off + j] = (av_lfg_get(&c->rnd) & 0x3FC) - 510;
break;
case 0:
break;
case 1:
for(j = 0; j < SAMPLES_PER_BAND; j += SAMPLES_PER_BAND / 2){
cnt = get_vlc2(gb, q1_vlc.table, MPC8_Q1_BITS, 2);
t = mpc8_get_mask(gb, 18, cnt);
for(k = 0; k < SAMPLES_PER_BAND / 2; k++, t <<= 1)
c->Q[ch][off + j + k] = (t & 0x20000) ? (get_bits1(gb) << 1) - 1 : 0;
}
break;
case 2:
cnt = 6;//2*mpc8_thres[res]
for(j = 0; j < SAMPLES_PER_BAND; j += 3){
t = get_vlc2(gb, q2_vlc[cnt > 3].table, MPC8_Q2_BITS, 2);
c->Q[ch][off + j + 0] = mpc8_idx50[t];
c->Q[ch][off + j + 1] = mpc8_idx51[t];
c->Q[ch][off + j + 2] = mpc8_idx52[t];
cnt = (cnt >> 1) + mpc8_huffq2[t];
}
break;
case 3:
case 4:
for(j = 0; j < SAMPLES_PER_BAND; j += 2){
t = get_vlc2(gb, q3_vlc[res - 3].table, MPC8_Q3_BITS, 2) + q3_offsets[res - 3];
c->Q[ch][off + j + 1] = t >> 4;
c->Q[ch][off + j + 0] = (t & 8) ? (t & 0xF) - 16 : (t & 0xF);
}
break;
case 5:
case 6:
case 7:
case 8:
cnt = 2 * mpc8_thres[res];
for(j = 0; j < SAMPLES_PER_BAND; j++){
t = get_vlc2(gb, quant_vlc[res - 5][cnt > mpc8_thres[res]].table, quant_vlc[res - 5][cnt > mpc8_thres[res]].bits, 2) + quant_offsets[res - 5];
c->Q[ch][off + j] = t;
cnt = (cnt >> 1) + FFABS(c->Q[ch][off + j]);
}
break;
default:
for(j = 0; j < SAMPLES_PER_BAND; j++){
c->Q[ch][off + j] = get_vlc2(gb, q9up_vlc.table, MPC8_Q9UP_BITS, 2);
if(res != 9){
c->Q[ch][off + j] <<= res - 9;
c->Q[ch][off + j] |= get_bits(gb, res - 9);
}
c->Q[ch][off + j] -= (1 << (res - 2)) - 1;
}
}
}
}
ff_mpc_dequantize_and_synth(c, maxband, data, avctx->channels);
c->cur_frame++;
c->last_bits_used = get_bits_count(gb);
if(c->cur_frame >= c->frames)
c->cur_frame = 0;
*data_size = MPC_FRAME_SIZE * 2 * avctx->channels;
return c->cur_frame ? c->last_bits_used >> 3 : buf_size;
}
AVCodec ff_mpc8_decoder = {
"mpc8",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_MUSEPACK8,
sizeof(MPCContext),
mpc8_decode_init,
NULL,
NULL,
mpc8_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Musepack SV8"),
};
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