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FFmpeg/libavcodec/aacdec.c
Andreas Rheinhardt fc5d22abe4 avcodec/aacdec, aactab: Move kbd tables to their only user 
The floating point kbd tables for 120 and 960 samples are only used by
the floating point decoder whereas the fixed point kbd tables for 128
and 1024 samples are only used by the fixed point AAC decoder. So move
these tables to their only users. This ensures that they are not
accidentally used somewhere else without ensuring that initializing
these tables stays thread-safe (as it is now because the only place from
where they are initialized is guarded by an AVOnce).

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2020-12-08 17:51:48 +01:00

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/*
* AAC decoder
* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
* Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
*
* AAC LATM decoder
* Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
* Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
*
* 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
* AAC decoder
* @author Oded Shimon ( ods15 ods15 dyndns org )
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
*/
#define FFT_FLOAT 1
#define FFT_FIXED_32 0
#define USE_FIXED 0
#include "libavutil/float_dsp.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "internal.h"
#include "get_bits.h"
#include "fft.h"
#include "mdct15.h"
#include "lpc.h"
#include "kbdwin.h"
#include "sinewin.h"
#include "aac.h"
#include "aactab.h"
#include "aacdectab.h"
#include "adts_header.h"
#include "cbrt_data.h"
#include "sbr.h"
#include "aacsbr.h"
#include "mpeg4audio.h"
#include "profiles.h"
#include "libavutil/intfloat.h"
#include <errno.h>
#include <math.h>
#include <stdint.h>
#include <string.h>
#if ARCH_ARM
# include "arm/aac.h"
#elif ARCH_MIPS
# include "mips/aacdec_mips.h"
#endif
DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME(sine_120))[120];
DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME(sine_960))[960];
DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME(aac_kbd_long_960))[960];
DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME(aac_kbd_short_120))[120];
static av_always_inline void reset_predict_state(PredictorState *ps)
{
ps->r0 = 0.0f;
ps->r1 = 0.0f;
ps->cor0 = 0.0f;
ps->cor1 = 0.0f;
ps->var0 = 1.0f;
ps->var1 = 1.0f;
}
#ifndef VMUL2
static inline float *VMUL2(float *dst, const float *v, unsigned idx,
const float *scale)
{
float s = *scale;
*dst++ = v[idx & 15] * s;
*dst++ = v[idx>>4 & 15] * s;
return dst;
}
#endif
#ifndef VMUL4
static inline float *VMUL4(float *dst, const float *v, unsigned idx,
const float *scale)
{
float s = *scale;
*dst++ = v[idx & 3] * s;
*dst++ = v[idx>>2 & 3] * s;
*dst++ = v[idx>>4 & 3] * s;
*dst++ = v[idx>>6 & 3] * s;
return dst;
}
#endif
#ifndef VMUL2S
static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
unsigned sign, const float *scale)
{
union av_intfloat32 s0, s1;
s0.f = s1.f = *scale;
s0.i ^= sign >> 1 << 31;
s1.i ^= sign << 31;
*dst++ = v[idx & 15] * s0.f;
*dst++ = v[idx>>4 & 15] * s1.f;
return dst;
}
#endif
#ifndef VMUL4S
static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
unsigned sign, const float *scale)
{
unsigned nz = idx >> 12;
union av_intfloat32 s = { .f = *scale };
union av_intfloat32 t;
t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx & 3] * t.f;
sign <<= nz & 1; nz >>= 1;
t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx>>2 & 3] * t.f;
sign <<= nz & 1; nz >>= 1;
t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx>>4 & 3] * t.f;
sign <<= nz & 1;
t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx>>6 & 3] * t.f;
return dst;
}
#endif
static av_always_inline float flt16_round(float pf)
{
union av_intfloat32 tmp;
tmp.f = pf;
tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
return tmp.f;
}
static av_always_inline float flt16_even(float pf)
{
union av_intfloat32 tmp;
tmp.f = pf;
tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
return tmp.f;
}
static av_always_inline float flt16_trunc(float pf)
{
union av_intfloat32 pun;
pun.f = pf;
pun.i &= 0xFFFF0000U;
return pun.f;
}
static av_always_inline void predict(PredictorState *ps, float *coef,
int output_enable)
{
const float a = 0.953125; // 61.0 / 64
const float alpha = 0.90625; // 29.0 / 32
float e0, e1;
float pv;
float k1, k2;
float r0 = ps->r0, r1 = ps->r1;
float cor0 = ps->cor0, cor1 = ps->cor1;
float var0 = ps->var0, var1 = ps->var1;
k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
pv = flt16_round(k1 * r0 + k2 * r1);
if (output_enable)
*coef += pv;
e0 = *coef;
e1 = e0 - k1 * r0;
ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
ps->r0 = flt16_trunc(a * e0);
}
/**
* Apply dependent channel coupling (applied before IMDCT).
*
* @param index index into coupling gain array
*/
static void apply_dependent_coupling(AACContext *ac,
SingleChannelElement *target,
ChannelElement *cce, int index)
{
IndividualChannelStream *ics = &cce->ch[0].ics;
const uint16_t *offsets = ics->swb_offset;
float *dest = target->coeffs;
const float *src = cce->ch[0].coeffs;
int g, i, group, k, idx = 0;
if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
av_log(ac->avctx, AV_LOG_ERROR,
"Dependent coupling is not supported together with LTP\n");
return;
}
for (g = 0; g < ics->num_window_groups; g++) {
for (i = 0; i < ics->max_sfb; i++, idx++) {
if (cce->ch[0].band_type[idx] != ZERO_BT) {
const float gain = cce->coup.gain[index][idx];
for (group = 0; group < ics->group_len[g]; group++) {
for (k = offsets[i]; k < offsets[i + 1]; k++) {
// FIXME: SIMDify
dest[group * 128 + k] += gain * src[group * 128 + k];
}
}
}
}
dest += ics->group_len[g] * 128;
src += ics->group_len[g] * 128;
}
}
/**
* Apply independent channel coupling (applied after IMDCT).
*
* @param index index into coupling gain array
*/
static void apply_independent_coupling(AACContext *ac,
SingleChannelElement *target,
ChannelElement *cce, int index)
{
const float gain = cce->coup.gain[index][0];
const float *src = cce->ch[0].ret;
float *dest = target->ret;
const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
ac->fdsp->vector_fmac_scalar(dest, src, gain, len);
}
#include "aacdec_template.c"
#define LOAS_SYNC_WORD 0x2b7 ///< 11 bits LOAS sync word
struct LATMContext {
AACContext aac_ctx; ///< containing AACContext
int initialized; ///< initialized after a valid extradata was seen
// parser data
int audio_mux_version_A; ///< LATM syntax version
int frame_length_type; ///< 0/1 variable/fixed frame length
int frame_length; ///< frame length for fixed frame length
};
static inline uint32_t latm_get_value(GetBitContext *b)
{
int length = get_bits(b, 2);
return get_bits_long(b, (length+1)*8);
}
static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
GetBitContext *gb, int asclen)
{
AACContext *ac = &latmctx->aac_ctx;
AVCodecContext *avctx = ac->avctx;
MPEG4AudioConfig m4ac = { 0 };
GetBitContext gbc;
int config_start_bit = get_bits_count(gb);
int sync_extension = 0;
int bits_consumed, esize, i;
if (asclen > 0) {
sync_extension = 1;
asclen = FFMIN(asclen, get_bits_left(gb));
init_get_bits(&gbc, gb->buffer, config_start_bit + asclen);
skip_bits_long(&gbc, config_start_bit);
} else if (asclen == 0) {
gbc = *gb;
} else {
return AVERROR_INVALIDDATA;
}
if (get_bits_left(gb) <= 0)
return AVERROR_INVALIDDATA;
bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &m4ac,
&gbc, config_start_bit,
sync_extension);
if (bits_consumed < config_start_bit)
return AVERROR_INVALIDDATA;
bits_consumed -= config_start_bit;
if (asclen == 0)
asclen = bits_consumed;
if (!latmctx->initialized ||
ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
ac->oc[1].m4ac.chan_config != m4ac.chan_config) {
if (latmctx->initialized) {
av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", m4ac.sample_rate, m4ac.chan_config);
} else {
av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n");
}
latmctx->initialized = 0;
esize = (asclen + 7) / 8;
if (avctx->extradata_size < esize) {
av_free(avctx->extradata);
avctx->extradata = av_malloc(esize + AV_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata)
return AVERROR(ENOMEM);
}
avctx->extradata_size = esize;
gbc = *gb;
for (i = 0; i < esize; i++) {
avctx->extradata[i] = get_bits(&gbc, 8);
}
memset(avctx->extradata+esize, 0, AV_INPUT_BUFFER_PADDING_SIZE);
}
skip_bits_long(gb, asclen);
return 0;
}
static int read_stream_mux_config(struct LATMContext *latmctx,
GetBitContext *gb)
{
int ret, audio_mux_version = get_bits(gb, 1);
latmctx->audio_mux_version_A = 0;
if (audio_mux_version)
latmctx->audio_mux_version_A = get_bits(gb, 1);
if (!latmctx->audio_mux_version_A) {
if (audio_mux_version)
latm_get_value(gb); // taraFullness
skip_bits(gb, 1); // allStreamSameTimeFraming
skip_bits(gb, 6); // numSubFrames
// numPrograms
if (get_bits(gb, 4)) { // numPrograms
avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple programs");
return AVERROR_PATCHWELCOME;
}
// for each program (which there is only one in DVB)
// for each layer (which there is only one in DVB)
if (get_bits(gb, 3)) { // numLayer
avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers");
return AVERROR_PATCHWELCOME;
}
// for all but first stream: use_same_config = get_bits(gb, 1);
if (!audio_mux_version) {
if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0)
return ret;
} else {
int ascLen = latm_get_value(gb);
if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0)
return ret;
}
latmctx->frame_length_type = get_bits(gb, 3);
switch (latmctx->frame_length_type) {
case 0:
skip_bits(gb, 8); // latmBufferFullness
break;
case 1:
latmctx->frame_length = get_bits(gb, 9);
break;
case 3:
case 4:
case 5:
skip_bits(gb, 6); // CELP frame length table index
break;
case 6:
case 7:
skip_bits(gb, 1); // HVXC frame length table index
break;
}
if (get_bits(gb, 1)) { // other data
if (audio_mux_version) {
latm_get_value(gb); // other_data_bits
} else {
int esc;
do {
if (get_bits_left(gb) < 9)
return AVERROR_INVALIDDATA;
esc = get_bits(gb, 1);
skip_bits(gb, 8);
} while (esc);
}
}
if (get_bits(gb, 1)) // crc present
skip_bits(gb, 8); // config_crc
}
return 0;
}
static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
{
uint8_t tmp;
if (ctx->frame_length_type == 0) {
int mux_slot_length = 0;
do {
if (get_bits_left(gb) < 8)
return AVERROR_INVALIDDATA;
tmp = get_bits(gb, 8);
mux_slot_length += tmp;
} while (tmp == 255);
return mux_slot_length;
} else if (ctx->frame_length_type == 1) {
return ctx->frame_length;
} else if (ctx->frame_length_type == 3 ||
ctx->frame_length_type == 5 ||
ctx->frame_length_type == 7) {
skip_bits(gb, 2); // mux_slot_length_coded
}
return 0;
}
static int read_audio_mux_element(struct LATMContext *latmctx,
GetBitContext *gb)
{
int err;
uint8_t use_same_mux = get_bits(gb, 1);
if (!use_same_mux) {
if ((err = read_stream_mux_config(latmctx, gb)) < 0)
return err;
} else if (!latmctx->aac_ctx.avctx->extradata) {
av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
"no decoder config found\n");
return 1;
}
if (latmctx->audio_mux_version_A == 0) {
int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
if (mux_slot_length_bytes < 0 || mux_slot_length_bytes * 8LL > get_bits_left(gb)) {
av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
return AVERROR_INVALIDDATA;
} else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
"frame length mismatch %d << %d\n",
mux_slot_length_bytes * 8, get_bits_left(gb));
return AVERROR_INVALIDDATA;
}
}
return 0;
}
static int latm_decode_frame(AVCodecContext *avctx, void *out,
int *got_frame_ptr, AVPacket *avpkt)
{
struct LATMContext *latmctx = avctx->priv_data;
int muxlength, err;
GetBitContext gb;
if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0)
return err;
// check for LOAS sync word
if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
return AVERROR_INVALIDDATA;
muxlength = get_bits(&gb, 13) + 3;
// not enough data, the parser should have sorted this out
if (muxlength > avpkt->size)
return AVERROR_INVALIDDATA;
if ((err = read_audio_mux_element(latmctx, &gb)))
return (err < 0) ? err : avpkt->size;
if (!latmctx->initialized) {
if (!avctx->extradata) {
*got_frame_ptr = 0;
return avpkt->size;
} else {
push_output_configuration(&latmctx->aac_ctx);
if ((err = decode_audio_specific_config(
&latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac,
avctx->extradata, avctx->extradata_size*8LL, 1)) < 0) {
pop_output_configuration(&latmctx->aac_ctx);
return err;
}
latmctx->initialized = 1;
}
}
if (show_bits(&gb, 12) == 0xfff) {
av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
"ADTS header detected, probably as result of configuration "
"misparsing\n");
return AVERROR_INVALIDDATA;
}
switch (latmctx->aac_ctx.oc[1].m4ac.object_type) {
case AOT_ER_AAC_LC:
case AOT_ER_AAC_LTP:
case AOT_ER_AAC_LD:
case AOT_ER_AAC_ELD:
err = aac_decode_er_frame(avctx, out, got_frame_ptr, &gb);
break;
default:
err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt);
}
if (err < 0)
return err;
return muxlength;
}
static av_cold int latm_decode_init(AVCodecContext *avctx)
{
struct LATMContext *latmctx = avctx->priv_data;
int ret = aac_decode_init(avctx);
if (avctx->extradata_size > 0)
latmctx->initialized = !ret;
return ret;
}
AVCodec ff_aac_decoder = {
.name = "aac",
.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_AAC,
.priv_data_size = sizeof(AACContext),
.init = aac_decode_init,
.close = aac_decode_close,
.decode = aac_decode_frame,
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
},
.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
.channel_layouts = aac_channel_layout,
.flush = flush,
.priv_class = &aac_decoder_class,
.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
};
/*
Note: This decoder filter is intended to decode LATM streams transferred
in MPEG transport streams which only contain one program.
To do a more complex LATM demuxing a separate LATM demuxer should be used.
*/
AVCodec ff_aac_latm_decoder = {
.name = "aac_latm",
.long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_AAC_LATM,
.priv_data_size = sizeof(struct LATMContext),
.init = latm_decode_init,
.close = aac_decode_close,
.decode = latm_decode_frame,
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
},
.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
.channel_layouts = aac_channel_layout,
.flush = flush,
.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
};
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