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FFmpeg/libavcodec/mpc7.c
Andreas Rheinhardt 20f9727018 avcodec/codec_internal: Add FFCodec, hide internal part of AVCodec 
Up until now, codec.h contains both public and private parts
of AVCodec. This exposes the internals of AVCodec to users
and leads them into the temptation of actually using them
and forces us to forward-declare structures and types that
users can't use at all.

This commit changes this by adding a new structure FFCodec to
codec_internal.h that extends AVCodec, i.e. contains the public
AVCodec as first member; the private fields of AVCodec are moved
to this structure, leaving codec.h clean.

Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-03-21 01:33:09 +01:00

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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 "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 av_cold void mpc7_init_static(void)
{
static VLC_TYPE quant_tables[7224][2];
const uint8_t *raw_quant_table = mpc7_quant_vlcs;
INIT_VLC_STATIC_FROM_LENGTHS(&scfi_vlc, MPC7_SCFI_BITS, MPC7_SCFI_SIZE,
&mpc7_scfi[1], 2,
&mpc7_scfi[0], 2, 1, 0, 0, 1 << MPC7_SCFI_BITS);
INIT_VLC_STATIC_FROM_LENGTHS(&dscf_vlc, MPC7_DSCF_BITS, MPC7_DSCF_SIZE,
&mpc7_dscf[1], 2,
&mpc7_dscf[0], 2, 1, -7, 0, 1 << MPC7_DSCF_BITS);
INIT_VLC_STATIC_FROM_LENGTHS(&hdr_vlc, MPC7_HDR_BITS, MPC7_HDR_SIZE,
&mpc7_hdr[1], 2,
&mpc7_hdr[0], 2, 1, -5, 0, 1 << MPC7_HDR_BITS);
for (unsigned i = 0, offset = 0; i < MPC7_QUANT_VLC_TABLES; i++){
for (int j = 0; j < 2; j++) {
quant_vlc[i][j].table = &quant_tables[offset];
quant_vlc[i][j].table_allocated = FF_ARRAY_ELEMS(quant_tables) - offset;
ff_init_vlc_from_lengths(&quant_vlc[i][j], 9, mpc7_quant_vlc_sizes[i],
&raw_quant_table[1], 2,
&raw_quant_table[0], 2, 1,
mpc7_quant_vlc_off[i],
INIT_VLC_STATIC_OVERLONG, NULL);
raw_quant_table += 2 * mpc7_quant_vlc_sizes[i];
offset += quant_vlc[i][j].table_size;
}
}
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].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);
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);
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 = i ? get_vlc2(&gb, hdr_vlc.table, 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.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;
}
const FFCodec ff_mpc7_decoder = {
.p.name = "mpc7",
.p.long_name = NULL_IF_CONFIG_SMALL("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,
.decode = 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 },
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
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