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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-02 03:06:28 +02:00
FFmpeg/libavcodec/mpc7.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

305 lines
10 KiB
C

/*
* Musepack SV7 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
* 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 "mpc7data.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 const uint16_t quant_offsets[MPC7_QUANT_VLC_TABLES*2 + 1] =
{
0, 512, 1024, 1536, 2052, 2564, 3076, 3588, 4100, 4612, 5124,
5636, 6164, 6676, 7224
};
static av_cold int mpc7_decode_init(AVCodecContext * avctx)
{
int i, j;
MPCContext *c = avctx->priv_data;
GetBitContext gb;
uint8_t buf[16];
static int vlc_initialized = 0;
static VLC_TYPE scfi_table[1 << MPC7_SCFI_BITS][2];
static VLC_TYPE dscf_table[1 << MPC7_DSCF_BITS][2];
static VLC_TYPE hdr_table[1 << MPC7_HDR_BITS][2];
static VLC_TYPE quant_tables[7224][2];
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));
av_lfg_init(&c->rnd, 0xDEADBEEF);
dsputil_init(&c->dsp, avctx);
ff_mpadsp_init(&c->mpadsp);
c->dsp.bswap_buf((uint32_t*)buf, (const uint32_t*)avctx->extradata, 4);
ff_mpc_init();
init_get_bits(&gb, buf, 128);
c->IS = get_bits1(&gb);
c->MSS = get_bits1(&gb);
c->maxbands = get_bits(&gb, 6);
if(c->maxbands >= BANDS){
av_log(avctx, AV_LOG_ERROR, "Too many bands: %i\n", c->maxbands);
return -1;
}
skip_bits_long(&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->maxbands);
c->frames_to_skip = 0;
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");
scfi_vlc.table = scfi_table;
scfi_vlc.table_allocated = 1 << MPC7_SCFI_BITS;
if(init_vlc(&scfi_vlc, MPC7_SCFI_BITS, MPC7_SCFI_SIZE,
&mpc7_scfi[1], 2, 1,
&mpc7_scfi[0], 2, 1, INIT_VLC_USE_NEW_STATIC)){
av_log(avctx, AV_LOG_ERROR, "Cannot init SCFI VLC\n");
return -1;
}
dscf_vlc.table = dscf_table;
dscf_vlc.table_allocated = 1 << MPC7_DSCF_BITS;
if(init_vlc(&dscf_vlc, MPC7_DSCF_BITS, MPC7_DSCF_SIZE,
&mpc7_dscf[1], 2, 1,
&mpc7_dscf[0], 2, 1, INIT_VLC_USE_NEW_STATIC)){
av_log(avctx, AV_LOG_ERROR, "Cannot init DSCF VLC\n");
return -1;
}
hdr_vlc.table = hdr_table;
hdr_vlc.table_allocated = 1 << MPC7_HDR_BITS;
if(init_vlc(&hdr_vlc, MPC7_HDR_BITS, MPC7_HDR_SIZE,
&mpc7_hdr[1], 2, 1,
&mpc7_hdr[0], 2, 1, INIT_VLC_USE_NEW_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++){
quant_vlc[i][j].table = &quant_tables[quant_offsets[i*2 + j]];
quant_vlc[i][j].table_allocated = quant_offsets[i*2 + j + 1] - quant_offsets[i*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_NEW_STATIC)){
av_log(avctx, AV_LOG_ERROR, "Cannot init QUANT VLC %i,%i\n",i,j);
return -1;
}
}
}
vlc_initialized = 1;
return 0;
}
/**
* Fill samples for given subband
*/
static inline void 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++){
*dst++ = (av_lfg_get(&c->rnd) & 0x3FC) - 510;
}
break;
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++ = mpc7_idx30[t];
*dst++ = mpc7_idx31[t];
*dst++ = mpc7_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++ = mpc7_idx50[t];
*dst++ = mpc7_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 get_scale_idx(GetBitContext *gb, int ref)
{
int t = get_vlc2(gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7;
if (t == 8)
return get_bits(gb, 6);
return ref + t;
}
static int mpc7_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 gb;
uint8_t *bits;
int i, ch;
int mb = -1;
Band *bands = c->bands;
int off;
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((uint32_t*)bits, (const uint32_t*)(buf + 4), (buf_size - 4) >> 2);
init_get_bits(&gb, bits, (buf_size - 4)* 8);
skip_bits_long(&gb, buf[0]);
/* read subband indexes */
for(i = 0; i <= c->maxbands; i++){
for(ch = 0; ch < 2; ch++){
int t = 4;
if(i) t = get_vlc2(&gb, hdr_vlc.table, MPC7_HDR_BITS, 1) - 5;
if(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];
bands[i].scf_idx[ch][0] = get_scale_idx(&gb, bands[i].scf_idx[ch][2]);
switch(bands[i].scfi[ch]){
case 0:
bands[i].scf_idx[ch][1] = get_scale_idx(&gb, bands[i].scf_idx[ch][0]);
bands[i].scf_idx[ch][2] = get_scale_idx(&gb, bands[i].scf_idx[ch][1]);
break;
case 1:
bands[i].scf_idx[ch][1] = get_scale_idx(&gb, bands[i].scf_idx[ch][0]);
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];
bands[i].scf_idx[ch][2] = get_scale_idx(&gb, bands[i].scf_idx[ch][1]);
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(c->Q, 0, sizeof(c->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], c->Q[ch] + off);
ff_mpc_dequantize_and_synth(c, mb, data, 2);
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 ff_mpc7_decoder = {
"mpc7",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_MUSEPACK7,
sizeof(MPCContext),
mpc7_decode_init,
NULL,
NULL,
mpc7_decode_frame,
.flush = mpc7_decode_flush,
.long_name = NULL_IF_CONFIG_SMALL("Musepack SV7"),
};