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FFmpeg/libavcodec/mpc7.c
Andreas Rheinhardt 886fbec82f avcodec/mpc7: Avoid superfluous VLC structures 
Of all these VLCs here, only VLC.table was really used
after init, so use the ff_vlc_init_tables API
to get rid of them.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2023-10-31 20:47:00 +01:00

328 lines
11 KiB
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 "libavutil/mem_internal.h"
#include "libavutil/thread.h"
#include "avcodec.h"
#include "codec_internal.h"
#include "decode.h"
#include "get_bits.h"
#include "mpegaudiodsp.h"
#include "mpc.h"
#include "mpc7data.h"
static VLCElem scfi_vlc[1 << MPC7_SCFI_BITS];
static VLCElem dscf_vlc[1 << MPC7_DSCF_BITS];
static VLCElem hdr_vlc [1 << MPC7_HDR_BITS];
static const VLCElem *quant_vlc[MPC7_QUANT_VLC_TABLES][2];
static av_cold void mpc7_init_static(void)
{
static VLCElem quant_tables[7224];
VLCInitState state = VLC_INIT_STATE(quant_tables);
const uint8_t *raw_quant_table = mpc7_quant_vlcs;
VLC_INIT_STATIC_TABLE_FROM_LENGTHS(scfi_vlc, MPC7_SCFI_BITS, MPC7_SCFI_SIZE,
&mpc7_scfi[1], 2,
&mpc7_scfi[0], 2, 1, 0, 0);
VLC_INIT_STATIC_TABLE_FROM_LENGTHS(dscf_vlc, MPC7_DSCF_BITS, MPC7_DSCF_SIZE,
&mpc7_dscf[1], 2,
&mpc7_dscf[0], 2, 1, -7, 0);
VLC_INIT_STATIC_TABLE_FROM_LENGTHS(hdr_vlc, MPC7_HDR_BITS, MPC7_HDR_SIZE,
&mpc7_hdr[1], 2,
&mpc7_hdr[0], 2, 1, -5, 0);
for (int i = 0; i < MPC7_QUANT_VLC_TABLES; i++) {
for (int j = 0; j < 2; j++) {
quant_vlc[i][j] =
ff_vlc_init_tables_from_lengths(&state, 9, mpc7_quant_vlc_sizes[i],
&raw_quant_table[1], 2,
&raw_quant_table[0], 2, 1,
mpc7_quant_vlc_off[i], 0);
raw_quant_table += 2 * mpc7_quant_vlc_sizes[i];
}
}
ff_mpa_synth_init_fixed();
}
static av_cold int mpc7_decode_init(AVCodecContext * avctx)
{
static AVOnce init_static_once = AV_ONCE_INIT;
MPCContext *c = avctx->priv_data;
GetBitContext gb;
LOCAL_ALIGNED_16(uint8_t, buf, [16]);
/* Musepack SV7 is always stereo */
if (avctx->ch_layout.nb_channels != 2) {
avpriv_request_sample(avctx, "%d channels", avctx->ch_layout.nb_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;
av_channel_layout_uninit(&avctx->ch_layout);
avctx->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO;
ff_thread_once(&init_static_once, mpc7_init_static);
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], 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], 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], 9, 2);
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, MPC7_DSCF_BITS, 1);
if (t == 8)
return get_bits(gb, 6);
return ref + t;
}
static int mpc7_decode_frame(AVCodecContext *avctx, AVFrame *frame,
int *got_frame_ptr, AVPacket *avpkt)
{
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 = i ? get_vlc2(&gb, hdr_vlc, MPC7_HDR_BITS, 1) : 4;
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, 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;
}
const FFCodec ff_mpc7_decoder = {
.p.name = "mpc7",
CODEC_LONG_NAME("Musepack SV7"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_MUSEPACK7,
.priv_data_size = sizeof(MPCContext),
.init = mpc7_decode_init,
.close = mpc7_decode_close,
FF_CODEC_DECODE_CB(mpc7_decode_frame),
.flush = mpc7_decode_flush,
.p.capabilities = AV_CODEC_CAP_DR1,
.p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
};
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