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third_party_ffmpeg/libavcodec/mpc8.c
gg0907 65994f5530 upgrade to 4.4.1 
Signed-off-by: gg0907 <guohui_1701@163.com>
2022-05-23 11:23:35 +08:00

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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/channel_layout.h"
#include "libavutil/lfg.h"
#include "libavutil/thread.h"
#include "avcodec.h"
#include "get_bits.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 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 |= 1U << 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 av_cold void build_vlc(VLC *vlc, unsigned *buf_offset,
const uint8_t codes_counts[16],
const uint8_t **syms, int offset)
{
static VLC_TYPE vlc_buf[9296][2];
uint8_t len[MPC8_MAX_VLC_SIZE];
unsigned num = 0;
vlc->table = &vlc_buf[*buf_offset];
vlc->table_allocated = FF_ARRAY_ELEMS(vlc_buf) - *buf_offset;
for (int i = 16; i > 0; i--)
for (unsigned tmp = num + codes_counts[i - 1]; num < tmp; num++)
len[num] = i;
ff_init_vlc_from_lengths(vlc, FFMIN(len[0], 9), num, len, 1,
*syms, 1, 1, offset, INIT_VLC_STATIC_OVERLONG, NULL);
*buf_offset += vlc->table_size;
*syms += num;
}
static av_cold void mpc8_init_static(void)
{
const uint8_t *q_syms = mpc8_q_syms, *bands_syms = mpc8_bands_syms;
const uint8_t *res_syms = mpc8_res_syms, *scfi_syms = mpc8_scfi_syms;
const uint8_t *dscf_syms = mpc8_dscf_syms;
unsigned offset = 0;
build_vlc(&band_vlc, &offset, mpc8_bands_len_counts, &bands_syms, 0);
build_vlc(&q1_vlc, &offset, mpc8_q1_len_counts, &q_syms, 0);
build_vlc(&q9up_vlc, &offset, mpc8_q9up_len_counts, &q_syms, 0);
for (int i = 0; i < 2; i++){
build_vlc(&scfi_vlc[i], &offset, mpc8_scfi_len_counts[i], &scfi_syms, 0);
build_vlc(&dscf_vlc[i], &offset, mpc8_dscf_len_counts[i], &dscf_syms, 0);
build_vlc(&res_vlc[i], &offset, mpc8_res_len_counts[i], &res_syms, 0);
build_vlc(&q2_vlc[i], &offset, mpc8_q2_len_counts[i], &q_syms, 0);
build_vlc(&q3_vlc[i], &offset, mpc8_q34_len_counts[i],
&q_syms, -48 - 16 * i);
for (int j = 0; j < 4; j++)
build_vlc(&quant_vlc[j][i], &offset, mpc8_q5_8_len_counts[i][j],
&q_syms, -((8 << j) - 1));
}
ff_mpa_synth_init_fixed();
}
static av_cold int mpc8_decode_init(AVCodecContext * avctx)
{
static AVOnce init_static_once = AV_ONCE_INIT;
MPCContext *c = avctx->priv_data;
GetBitContext gb;
int channels;
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);
init_get_bits(&gb, avctx->extradata, 16);
skip_bits(&gb, 3);//sample rate
c->maxbands = get_bits(&gb, 5) + 1;
if (c->maxbands >= BANDS) {
av_log(avctx,AV_LOG_ERROR, "maxbands %d too high\n", c->maxbands);
return AVERROR_INVALIDDATA;
}
channels = get_bits(&gb, 4) + 1;
if (channels > 2) {
avpriv_request_sample(avctx, "Multichannel MPC SV8");
return AVERROR_PATCHWELCOME;
}
c->MSS = get_bits1(&gb);
c->frames = 1 << (get_bits(&gb, 3) * 2);
avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
avctx->channel_layout = (channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
avctx->channels = channels;
ff_thread_once(&init_static_once, mpc8_init_static);
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;
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;
}
if ((res = init_get_bits8(gb, buf, buf_size)) < 0)
return res;
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;
}
if (get_bits_left(gb) < 0) {
*got_frame_ptr = 0;
return buf_size;
}
if(maxband > c->maxbands + 1) {
av_log(avctx, AV_LOG_ERROR, "maxband %d too large\n",maxband);
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] = 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++)
c->Q[ch][off + j + k] = t & (1 << (SAMPLES_PER_BAND / 2 - k - 1))
? (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);
c->Q[ch][off + j + 1] = t >> 4;
c->Q[ch][off + j + 0] = sign_extend(t, 4);
}
break;
case 5:
case 6:
case 7:
case 8:
cnt = 2 * mpc8_thres[res];
for(j = 0; j < SAMPLES_PER_BAND; j++){
const VLC *vlc = &quant_vlc[res - 5][cnt > mpc8_thres[res]];
c->Q[ch][off + j] = get_vlc2(gb, vlc->table, vlc->bits, 2);
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;
}
}
}
}
frame->nb_samples = MPC_FRAME_SIZE;
if ((res = ff_get_buffer(avctx, frame, 0)) < 0)
return res;
ff_mpc_dequantize_and_synth(c, maxband - 1,
(int16_t **)frame->extended_data,
avctx->channels);
c->cur_frame++;
c->last_bits_used = get_bits_count(gb);
if(c->cur_frame >= c->frames)
c->cur_frame = 0;
if (get_bits_left(gb) < 0) {
av_log(avctx, AV_LOG_ERROR, "Overread %d\n", -get_bits_left(gb));
c->last_bits_used = buf_size << 3;
} else if (c->cur_frame == 0 && get_bits_left(gb) < 8) {// we have only padding left
c->last_bits_used = buf_size << 3;
}
*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 | AV_CODEC_CAP_CHANNEL_CONF,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
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