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/*
* Musepack SV8 decoder
* Copyright (c) 2007 Konstantin Shishkov
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; 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/channel_layout.h"
#include "libavutil/lfg.h"
#include "avcodec.h"
#include "bitstream.h"
#include "internal.h"
#include "mpegaudiodsp.h"
#include "mpc.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(BitstreamContext *bc, int k, int n)
{
int len = mpc8_cnk_len[k-1][n-1] - 1;
int code = len ? bitstream_read(bc, len) : 0;
if (code >= mpc8_cnk_lost[k-1][n-1])
code = ((code << 1) | bitstream_read_bit(bc)) - mpc8_cnk_lost[k - 1][n - 1];
return code;
}
static inline int mpc8_dec_enum(BitstreamContext *bc, int k, int n)
{
int bits = 0;
const uint32_t * C = mpc8_cnk[k-1];
int code = mpc8_dec_base(bc, 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(BitstreamContext *bc, int m)
{
if(mpc8_cnk_len[0][m] < 1) return 0;
return mpc8_dec_base(bc, 1, m + 1);
}
static int mpc8_get_mask(BitstreamContext *bc, int size, int t)
{
int mask = 0;
if(t && t != size)
mask = mpc8_dec_enum(bc, 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;
BitstreamContext bc;
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);
ff_mpadsp_init(&c->mpadsp);
ff_mpc_init();
bitstream_init(&bc, avctx->extradata, 16);
bitstream_skip(&bc, 3); // sample rate
c->maxbands = bitstream_read(&bc, 5) + 1;
channels = bitstream_read(&bc, 4) + 1;
if (channels > 2) {
avpriv_request_sample(avctx, "Multichannel MPC SV8");
return AVERROR_PATCHWELCOME;
}
c->MSS = bitstream_read_bit(&bc);
c->frames = 1 << (bitstream_read(&bc, 3) * 2);
avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
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;
ff_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;
ff_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 *got_frame_ptr, AVPacket *avpkt)
{
AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
MPCContext *c = avctx->priv_data;
BitstreamContext bc2, *bc = &bc2;
int i, j, k, ch, cnt, res, t;
Band *bands = c->bands;
int off;
int maxband, keyframe;
int last[2];
/* get output buffer */
frame->nb_samples = MPC_FRAME_SIZE;
if ((res = ff_get_buffer(avctx, frame, 0)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return res;
}
keyframe = c->cur_frame == 0;
if(keyframe){
memset(c->Q, 0, sizeof(c->Q));
c->last_bits_used = 0;
}
bitstream_init8(bc, buf, buf_size);
bitstream_skip(bc, c->last_bits_used & 7);
if(keyframe)
maxband = mpc8_get_mod_golomb(bc, c->maxbands + 1);
else{
maxband = c->last_max_band + bitstream_read_vlc(bc, band_vlc.table, MPC8_BANDS_BITS, 2);
if(maxband > 32) maxband -= 33;
}
if(maxband > c->maxbands + 1)
return AVERROR_INVALIDDATA;
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] = bitstream_read_vlc(bc, 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(bc, cnt);
mask = mpc8_get_mask(bc, 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 = bitstream_read_vlc(bc, 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] = bitstream_read(bc, 7) - 6;
c->oldDSCF[ch][i] = 0;
}else{
t = bitstream_read_vlc(bc, dscf_vlc[1].table, MPC8_DSCF1_BITS, 2);
if(t == 64)
t += bitstream_read(bc, 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 = bitstream_read_vlc(bc, dscf_vlc[0].table, MPC8_DSCF0_BITS, 2);
if(t == 31)
t = 64 + bitstream_read(bc, 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 = bitstream_read_vlc(bc, q1_vlc.table, MPC8_Q1_BITS, 2);
t = mpc8_get_mask(bc, 18, cnt);
for(k = 0; k < SAMPLES_PER_BAND / 2; k++, t <<= 1)
c->Q[ch][off + j + k] = (t & 0x20000) ? (bitstream_read_bit(bc) << 1) - 1 : 0;
}
break;
case 2:
cnt = 6;//2*mpc8_thres[res]
for(j = 0; j < SAMPLES_PER_BAND; j += 3){
t = bitstream_read_vlc(bc, 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 = bitstream_read_vlc(bc, 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 = bitstream_read_vlc(bc, 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] = bitstream_read_vlc(bc, q9up_vlc.table, MPC8_Q9UP_BITS, 2);
if(res != 9){
c->Q[ch][off + j] <<= res - 9;
c->Q[ch][off + j] |= bitstream_read(bc, res - 9);
}
c->Q[ch][off + j] -= (1 << (res - 2)) - 1;
}
}
}
}
ff_mpc_dequantize_and_synth(c, maxband - 1,
(int16_t **)frame->extended_data,
avctx->channels);
c->cur_frame++;
c->last_bits_used = bitstream_tell(bc);
if (bitstream_bits_left(bc) < 8) // we have only padding left
c->last_bits_used = buf_size << 3;
if(c->cur_frame >= c->frames)
c->cur_frame = 0;
*got_frame_ptr = 1;
return c->cur_frame ? c->last_bits_used >> 3 : buf_size;
}
static av_cold void mpc8_decode_flush(AVCodecContext *avctx)
{
MPCContext *c = avctx->priv_data;
c->cur_frame = 0;
}
AVCodec ff_mpc8_decoder = {
.name = "mpc8",
.long_name = NULL_IF_CONFIG_SMALL("Musepack SV8"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MUSEPACK8,
.priv_data_size = sizeof(MPCContext),
.init = mpc8_decode_init,
.decode = mpc8_decode_frame,
.flush = mpc8_decode_flush,
.capabilities = AV_CODEC_CAP_DR1,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
};