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FFmpeg/libavcodec/mpc7.c
Andreas Rheinhardt 16bb8247b4 avcodec/mpegaudiodsp: Make initializing synth windows thread-safe 
These arrays are used by the Musepack decoders, the MPEG audio decoders
as well as qdm2 and up until now, these arrays might be initialized more
than once, leading to potential data races as well as unnecessary
initializations. Therefore this commit ensures that each array will only
be initialized once.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2020-11-24 11:35:03 +01:00

327 lines
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C
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/*
* 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/channel_layout.h"
#include "libavutil/internal.h"
#include "libavutil/lfg.h"
#include "avcodec.h"
#include "get_bits.h"
#include "internal.h"
#include "mpegaudiodsp.h"
#include "mpc.h"
#include "mpc7data.h"
static VLC scfi_vlc, dscf_vlc, hdr_vlc, quant_vlc[MPC7_QUANT_VLC_TABLES][2];
static const uint16_t quant_sizes[MPC7_QUANT_VLC_TABLES*2] =
{
512, 512, 512, 516, 512, 512, 512, 512, 512, 512, 512, 528, 512, 548
};
static av_cold int mpc7_decode_init(AVCodecContext * avctx)
{
int i, j;
MPCContext *c = avctx->priv_data;
GetBitContext gb;
LOCAL_ALIGNED_16(uint8_t, buf, [16]);
static int vlc_initialized = 0;
static VLC_TYPE quant_tables[7224][2];
VLC_TYPE (*quant_table)[2] = quant_tables;
const uint16_t *raw_quant_table = mpc7_quant_vlcs;
/* Musepack SV7 is always stereo */
if (avctx->channels != 2) {
avpriv_request_sample(avctx, "%d channels", avctx->channels);
return AVERROR_PATCHWELCOME;
}
if(avctx->extradata_size < 16){
av_log(avctx, AV_LOG_ERROR, "Too small extradata size (%i)!\n", avctx->extradata_size);
return AVERROR_INVALIDDATA;
}
memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
av_lfg_init(&c->rnd, 0xDEADBEEF);
ff_bswapdsp_init(&c->bdsp);
ff_mpadsp_init(&c->mpadsp);
c->bdsp.bswap_buf((uint32_t *) buf, (const uint32_t *) avctx->extradata, 4);
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 AVERROR_INVALIDDATA;
}
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_S16P;
avctx->channel_layout = AV_CH_LAYOUT_STEREO;
if(vlc_initialized) return 0;
av_log(avctx, AV_LOG_DEBUG, "Initing VLC\n");
INIT_VLC_STATIC(&scfi_vlc, MPC7_SCFI_BITS, MPC7_SCFI_SIZE,
&mpc7_scfi[1], 2, 1,
&mpc7_scfi[0], 2, 1, 1 << MPC7_SCFI_BITS);
INIT_VLC_STATIC(&dscf_vlc, MPC7_DSCF_BITS, MPC7_DSCF_SIZE,
&mpc7_dscf[1], 2, 1,
&mpc7_dscf[0], 2, 1, 1 << MPC7_DSCF_BITS);
INIT_VLC_STATIC(&hdr_vlc, MPC7_HDR_BITS, MPC7_HDR_SIZE,
&mpc7_hdr[1], 2, 1,
&mpc7_hdr[0], 2, 1, 1 << MPC7_HDR_BITS);
for(i = 0; i < MPC7_QUANT_VLC_TABLES; i++){
for(j = 0; j < 2; j++){
quant_vlc[i][j].table = quant_table;
quant_vlc[i][j].table_allocated = quant_sizes[i * 2 + j];
quant_table += quant_sizes[i * 2 + j];
init_vlc(&quant_vlc[i][j], 9, mpc7_quant_vlc_sizes[i],
raw_quant_table + 1, 4, 2,
raw_quant_table, 4, 2, INIT_VLC_USE_NEW_STATIC);
raw_quant_table += 2 * mpc7_quant_vlc_sizes[i];
}
}
vlc_initialized = 1;
ff_mpa_synth_init_fixed();
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 *got_frame_ptr, AVPacket *avpkt)
{
AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size;
MPCContext *c = avctx->priv_data;
GetBitContext gb;
int i, ch;
int mb = -1;
Band *bands = c->bands;
int off, ret, last_frame, skip;
int bits_used, bits_avail;
memset(bands, 0, sizeof(*bands) * (c->maxbands + 1));
buf_size = avpkt->size & ~3;
if (buf_size <= 0) {
av_log(avctx, AV_LOG_ERROR, "packet size is too small (%i bytes)\n",
avpkt->size);
return AVERROR_INVALIDDATA;
}
if (buf_size != avpkt->size) {
av_log(avctx, AV_LOG_WARNING, "packet size is not a multiple of 4. "
"extra bytes at the end will be skipped.\n");
}
skip = buf[0];
last_frame = buf[1];
buf += 4;
buf_size -= 4;
/* get output buffer */
frame->nb_samples = MPC_FRAME_SIZE;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
av_fast_padded_malloc(&c->bits, &c->buf_size, buf_size);
if (!c->bits)
return AVERROR(ENOMEM);
c->bdsp.bswap_buf((uint32_t *) c->bits, (const uint32_t *) buf,
buf_size >> 2);
if ((ret = init_get_bits8(&gb, c->bits, buf_size)) < 0)
return ret;
skip_bits_long(&gb, skip);
/* 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[ch] < -1 || bands[i].res[ch] > 17) {
av_log(avctx, AV_LOG_ERROR, "subband index invalid\n");
return AVERROR_INVALIDDATA;
}
}
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, (int16_t **)frame->extended_data, 2);
if(last_frame)
frame->nb_samples = c->lastframelen;
bits_used = get_bits_count(&gb);
bits_avail = buf_size * 8;
if (!last_frame && ((bits_avail < bits_used) || (bits_used + 32 <= bits_avail))) {
av_log(avctx, AV_LOG_ERROR, "Error decoding frame: used %i of %i bits\n", bits_used, bits_avail);
return AVERROR_INVALIDDATA;
}
if(c->frames_to_skip){
c->frames_to_skip--;
*got_frame_ptr = 0;
return avpkt->size;
}
*got_frame_ptr = 1;
return avpkt->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;
}
static av_cold int mpc7_decode_close(AVCodecContext *avctx)
{
MPCContext *c = avctx->priv_data;
av_freep(&c->bits);
c->buf_size = 0;
return 0;
}
AVCodec ff_mpc7_decoder = {
.name = "mpc7",
.long_name = NULL_IF_CONFIG_SMALL("Musepack SV7"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MUSEPACK7,
.priv_data_size = sizeof(MPCContext),
.init = mpc7_decode_init,
.close = mpc7_decode_close,
.decode = mpc7_decode_frame,
.flush = mpc7_decode_flush,
.capabilities = AV_CODEC_CAP_DR1,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
};
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