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aacdec: add a decoder for AAC USAC (xHE-AAC)

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This commit adds a decoder for the frequency-domain part of USAC.

What works:
 - Mono
 - Stereo (no prediction)
 - Stereo (mid/side coding)
 - Stereo (complex prediction)

What's left:
 - SBR
 - Speech coding

Known issues:
 - Desync with certain sequences
 - Preroll crossover missing (shouldn't matter, bitrate adaptation only)
This commit is contained in:
Lynne 2024-05-16 11:36:12 +02:00
parent 23b45d7e20
commit eee5fa0808
No known key found for this signature in database
GPG Key ID: A2FEA5F03F034464
13 changed files with 2510 additions and 76 deletions
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3
libavcodec/aac/Makefile
View File

@ -2,6 +2,7 @@ clean::
$(RM) $(CLEANSUFFIXES:%=libavcodec/aac/%)
OBJS-$(CONFIG_AAC_DECODER) += aac/aacdec.o aac/aacdec_tab.o \
aac/aacdec_float.o
aac/aacdec_float.o aac/aacdec_usac.o \
aac/aacdec_ac.o aac/aacdec_lpd.o
OBJS-$(CONFIG_AAC_FIXED_DECODER) += aac/aacdec.o aac/aacdec_tab.o \
aac/aacdec_fixed.o

188
libavcodec/aac/aacdec.c
View File

@ -40,6 +40,7 @@
#include "aacdec.h"
#include "aacdec_tab.h"
#include "aacdec_usac.h"
#include "libavcodec/aac.h"
#include "libavcodec/aac_defines.h"
@ -535,6 +536,8 @@ static av_cold void flush(AVCodecContext *avctx)
}
}
}
ff_aac_usac_reset_state(ac, &ac->oc[1]);
}
/**
@ -993,13 +996,14 @@ static int decode_eld_specific_config(AACDecContext *ac, AVCodecContext *avctx,
*/
static int decode_audio_specific_config_gb(AACDecContext *ac,
AVCodecContext *avctx,
MPEG4AudioConfig *m4ac,
OutputConfiguration *oc,
GetBitContext *gb,
int get_bit_alignment,
int sync_extension)
{
int i, ret;
GetBitContext gbc = *gb;
MPEG4AudioConfig *m4ac = &oc->m4ac;
MPEG4AudioConfig m4ac_bak = *m4ac;
if ((i = ff_mpeg4audio_get_config_gb(m4ac, &gbc, sync_extension, avctx)) < 0) {
@ -1033,14 +1037,22 @@ static int decode_audio_specific_config_gb(AACDecContext *ac,
case AOT_ER_AAC_LC:
case AOT_ER_AAC_LD:
if ((ret = decode_ga_specific_config(ac, avctx, gb, get_bit_alignment,
m4ac, m4ac->chan_config)) < 0)
&oc->m4ac, m4ac->chan_config)) < 0)
return ret;
break;
case AOT_ER_AAC_ELD:
if ((ret = decode_eld_specific_config(ac, avctx, gb,
m4ac, m4ac->chan_config)) < 0)
&oc->m4ac, m4ac->chan_config)) < 0)
return ret;
break;
#if CONFIG_AAC_DECODER
case AOT_USAC_NOSBR: /* fallthrough */
case AOT_USAC:
if ((ret = ff_aac_usac_config_decode(ac, avctx, gb,
oc, m4ac->chan_config)) < 0)
return ret;
break;
#endif
default:
avpriv_report_missing_feature(avctx,
"Audio object type %s%d",
@ -1060,7 +1072,7 @@ static int decode_audio_specific_config_gb(AACDecContext *ac,
static int decode_audio_specific_config(AACDecContext *ac,
AVCodecContext *avctx,
MPEG4AudioConfig *m4ac,
OutputConfiguration *oc,
const uint8_t *data, int64_t bit_size,
int sync_extension)
{
@ -1080,7 +1092,7 @@ static int decode_audio_specific_config(AACDecContext *ac,
if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
return ret;
return decode_audio_specific_config_gb(ac, avctx, m4ac, &gb, 0,
return decode_audio_specific_config_gb(ac, avctx, oc, &gb, 0,
sync_extension);
}
@ -1104,6 +1116,15 @@ static av_cold int decode_close(AVCodecContext *avctx)
{
AACDecContext *ac = avctx->priv_data;
for (int i = 0; i < 2; i++) {
OutputConfiguration *oc = &ac->oc[i];
AACUSACConfig *usac = &oc->usac;
for (int j = 0; j < usac->nb_elems; j++) {
AACUsacElemConfig *ec = &usac->elems[i];
av_freep(&ec->ext.pl_data);
}
}
for (int type = 0; type < FF_ARRAY_ELEMS(ac->che); type++) {
for (int i = 0; i < MAX_ELEM_ID; i++) {
if (ac->che[type][i]) {
@ -1181,7 +1202,7 @@ av_cold int ff_aac_decode_init(AVCodecContext *avctx)
ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
if (avctx->extradata_size > 0) {
if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1],
avctx->extradata,
avctx->extradata_size * 8LL,
1)) < 0)
@ -1549,9 +1570,16 @@ static int decode_pulses(Pulse *pulse, GetBitContext *gb,
int ff_aac_decode_tns(AACDecContext *ac, TemporalNoiseShaping *tns,
GetBitContext *gb, const IndividualChannelStream *ics)
{
int tns_max_order = INT32_MAX;
const int is_usac = ac->oc[1].m4ac.object_type == AOT_USAC ||
ac->oc[1].m4ac.object_type == AOT_USAC_NOSBR;
int w, filt, i, coef_len, coef_res, coef_compress;
const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
const int tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
/* USAC doesn't seem to have a limit */
if (!is_usac)
tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
for (w = 0; w < ics->num_windows; w++) {
if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
coef_res = get_bits1(gb);
@ -1560,7 +1588,12 @@ int ff_aac_decode_tns(AACDecContext *ac, TemporalNoiseShaping *tns,
int tmp2_idx;
tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
if (is_usac)
tns->order[w][filt] = get_bits(gb, 4 - is8);
else
tns->order[w][filt] = get_bits(gb, 5 - (2 * is8));
if (tns->order[w][filt] > tns_max_order) {
av_log(ac->avctx, AV_LOG_ERROR,
"TNS filter order %d is greater than maximum %d.\n",
tns->order[w][filt], tns_max_order);
@ -1598,6 +1631,7 @@ static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
{
int idx;
int max_idx = cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
cpe->max_sfb_ste = cpe->ch[0].ics.max_sfb;
if (ms_present == 1) {
for (idx = 0; idx < max_idx; idx++)
cpe->ms_mask[idx] = get_bits1(gb);
@ -2182,42 +2216,19 @@ static int aac_decode_er_frame(AVCodecContext *avctx, AVFrame *frame,
return 0;
}
static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame,
int *got_frame_ptr, GetBitContext *gb,
const AVPacket *avpkt)
static int decode_frame_ga(AVCodecContext *avctx, AACDecContext *ac,
GetBitContext *gb, int *got_frame_ptr)
{
AACDecContext *ac = avctx->priv_data;
ChannelElement *che = NULL, *che_prev = NULL;
int err;
int is_dmono;
int elem_id;
enum RawDataBlockType elem_type, che_prev_type = TYPE_END;
int err, elem_id;
int samples = 0, multiplier, audio_found = 0, pce_found = 0;
int is_dmono, sce_count = 0;
int payload_alignment;
uint8_t che_presence[4][MAX_ELEM_ID] = {{0}};
ChannelElement *che = NULL, *che_prev = NULL;
int samples = 0, multiplier, audio_found = 0, pce_found = 0, sce_count = 0;
AVFrame *frame = ac->frame;
ac->frame = frame;
if (show_bits(gb, 12) == 0xfff) {
if ((err = parse_adts_frame_header(ac, gb)) < 0) {
av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
goto fail;
}
if (ac->oc[1].m4ac.sampling_index > 12) {
av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
err = AVERROR_INVALIDDATA;
goto fail;
}
}
if ((err = frame_configure_elements(avctx)) < 0)
goto fail;
// The AV_PROFILE_AAC_* defines are all object_type - 1
// This may lead to an undefined profile being signaled
ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
payload_alignment = get_bits_count(gb);
ac->tags_mapped = 0;
int payload_alignment = get_bits_count(gb);
// parse
while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
elem_id = get_bits(gb, 4);
@ -2225,28 +2236,23 @@ static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame,
if (avctx->debug & FF_DEBUG_STARTCODE)
av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id);
if (!avctx->ch_layout.nb_channels && elem_type != TYPE_PCE) {
err = AVERROR_INVALIDDATA;
goto fail;
}
if (!avctx->ch_layout.nb_channels && elem_type != TYPE_PCE)
return AVERROR_INVALIDDATA;
if (elem_type < TYPE_DSE) {
if (che_presence[elem_type][elem_id]) {
int error = che_presence[elem_type][elem_id] > 1;
av_log(ac->avctx, error ? AV_LOG_ERROR : AV_LOG_DEBUG, "channel element %d.%d duplicate\n",
elem_type, elem_id);
if (error) {
err = AVERROR_INVALIDDATA;
goto fail;
}
if (error)
return AVERROR_INVALIDDATA;
}
che_presence[elem_type][elem_id]++;
if (!(che=ff_aac_get_che(ac, elem_type, elem_id))) {
av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
elem_type, elem_id);
err = AVERROR_INVALIDDATA;
goto fail;
return AVERROR_INVALIDDATA;
}
samples = ac->oc[1].m4ac.frame_length_short ? 960 : 1024;
che->present = 1;
@ -2283,10 +2289,8 @@ static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame,
int tags;
int pushed = push_output_configuration(ac);
if (pce_found && !pushed) {
err = AVERROR_INVALIDDATA;
goto fail;
}
if (pce_found && !pushed)
return AVERROR_INVALIDDATA;
tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb,
payload_alignment);
@ -2312,8 +2316,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame,
elem_id += get_bits(gb, 8) - 1;
if (get_bits_left(gb) < 8 * elem_id) {
av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
err = AVERROR_INVALIDDATA;
goto fail;
return AVERROR_INVALIDDATA;
}
err = 0;
while (elem_id > 0) {
@ -2337,19 +2340,16 @@ static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame,
}
if (err)
goto fail;
return err;
if (get_bits_left(gb) < 3) {
av_log(avctx, AV_LOG_ERROR, overread_err);
err = AVERROR_INVALIDDATA;
goto fail;
return AVERROR_INVALIDDATA;
}
}
if (!avctx->ch_layout.nb_channels) {
*got_frame_ptr = 0;
if (!avctx->ch_layout.nb_channels)
return 0;
}
multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
samples <<= multiplier;
@ -2364,16 +2364,17 @@ static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame,
if (!ac->frame->data[0] && samples) {
av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
err = AVERROR_INVALIDDATA;
goto fail;
return AVERROR_INVALIDDATA;
}
if (samples) {
ac->frame->nb_samples = samples;
ac->frame->sample_rate = avctx->sample_rate;
} else
*got_frame_ptr = 1;
} else {
av_frame_unref(ac->frame);
*got_frame_ptr = !!samples;
*got_frame_ptr = 0;
}
/* for dual-mono audio (SCE + SCE) */
is_dmono = ac->dmono_mode && sce_count == 2 &&
@ -2387,6 +2388,59 @@ static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame,
}
return 0;
}
static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame,
int *got_frame_ptr, GetBitContext *gb,
const AVPacket *avpkt)
{
int err;
AACDecContext *ac = avctx->priv_data;
ac->frame = frame;
*got_frame_ptr = 0;
if (show_bits(gb, 12) == 0xfff) {
if ((err = parse_adts_frame_header(ac, gb)) < 0) {
av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
goto fail;
}
if (ac->oc[1].m4ac.sampling_index > 12) {
av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
err = AVERROR_INVALIDDATA;
goto fail;
}
}
if ((err = frame_configure_elements(avctx)) < 0)
goto fail;
// The AV_PROFILE_AAC_* defines are all object_type - 1
// This may lead to an undefined profile being signaled
ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
ac->tags_mapped = 0;
if ((ac->oc[1].m4ac.object_type == AOT_USAC) ||
(ac->oc[1].m4ac.object_type == AOT_USAC_NOSBR)) {
if (ac->is_fixed) {
avpriv_report_missing_feature(ac->avctx,
"AAC USAC fixed-point decoding");
return AVERROR_PATCHWELCOME;
}
#if CONFIG_AAC_DECODER
err = ff_aac_usac_decode_frame(avctx, ac, gb, got_frame_ptr);
if (err < 0)
goto fail;
#endif
} else {
err = decode_frame_ga(avctx, ac, gb, got_frame_ptr);
if (err < 0)
goto fail;
}
return err;
fail:
pop_output_configuration(ac);
return err;
@ -2414,7 +2468,7 @@ static int aac_decode_frame(AVCodecContext *avctx, AVFrame *frame,
if (new_extradata) {
/* discard previous configuration */
ac->oc[1].status = OC_NONE;
err = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
err = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1],
new_extradata,
new_extradata_size * 8LL, 1);
if (err < 0) {

187
libavcodec/aac/aacdec.h
View File

@ -42,6 +42,8 @@
#include "libavcodec/avcodec.h"
#include "libavcodec/mpeg4audio.h"
#include "aacdec_ac.h"
typedef struct AACDecContext AACDecContext;
/**
@ -69,6 +71,32 @@ enum CouplingPoint {
AFTER_IMDCT = 3,
};
enum AACUsacElem {
ID_USAC_SCE = 0,
ID_USAC_CPE = 1,
ID_USAC_LFE = 2,
ID_USAC_EXT = 3,
};
enum ExtensionHeaderType {
ID_CONFIG_EXT_FILL = 0,
ID_CONFIG_EXT_LOUDNESS_INFO = 2,
ID_CONFIG_EXT_STREAM_ID = 7,
};
enum AACUsacExtension {
ID_EXT_ELE_FILL,
ID_EXT_ELE_MPEGS,
ID_EXT_ELE_SAOC,
ID_EXT_ELE_AUDIOPREROLL,
ID_EXT_ELE_UNI_DRC,
};
enum AACUSACLoudnessExt {
UNIDRCLOUDEXT_TERM = 0x0,
UNIDRCLOUDEXT_EQ = 0x1,
};
// Supposed to be equal to AAC_RENAME() in case of USE_FIXED.
#define RENAME_FIXED(name) name ## _fixed
@ -93,6 +121,40 @@ typedef struct LongTermPrediction {
int8_t used[MAX_LTP_LONG_SFB];
} LongTermPrediction;
/* Per channel core mode */
typedef struct AACUsacElemData {
uint8_t core_mode;
uint8_t scale_factor_grouping;
/* Timewarping ratio */
#define NUM_TW_NODES 16
uint8_t tw_ratio[NUM_TW_NODES];
struct {
uint8_t acelp_core_mode : 3;
uint8_t lpd_mode : 5;
uint8_t bpf_control_info : 1;
uint8_t core_mode_last : 1;
uint8_t fac_data_present : 1;
int last_lpd_mode;
} ldp;
struct {
unsigned int seed;
uint8_t level : 3;
uint8_t offset : 5;
} noise;
struct {
uint8_t gain;
uint32_t kv[8 /* (1024 / 16) / 8 */][8];
} fac;
AACArithState ac;
} AACUsacElemData;
/**
* Individual Channel Stream
*/
@ -145,11 +207,13 @@ typedef struct ChannelCoupling {
*/
typedef struct SingleChannelElement {
IndividualChannelStream ics;
AACUsacElemData ue; ///< USAC element data
TemporalNoiseShaping tns;
enum BandType band_type[128]; ///< band types
int sfo[128]; ///< scalefactor offsets
INTFLOAT_UNION(sf, [128]); ///< scalefactors (8 windows * 16 sfb max)
INTFLOAT_ALIGNED_UNION(32, coeffs, 1024); ///< coefficients for IMDCT, maybe processed
INTFLOAT_ALIGNED_UNION(32, prev_coeffs, 1024); ///< unscaled previous contents of coeffs[] for USAC
INTFLOAT_ALIGNED_UNION(32, saved, 1536); ///< overlap
INTFLOAT_ALIGNED_UNION(32, ret_buf, 2048); ///< PCM output buffer
INTFLOAT_ALIGNED_UNION(16, ltp_state, 3072); ///< time signal for LTP
@ -163,25 +227,148 @@ typedef struct SingleChannelElement {
};
} SingleChannelElement;
typedef struct AACUsacStereo {
uint8_t common_window;
uint8_t common_tw;
uint8_t ms_mask_mode;
uint8_t config_idx;
/* Complex prediction */
uint8_t use_prev_frame;
uint8_t pred_dir;
uint8_t complex_coef;
uint8_t pred_used[128];
INTFLOAT_ALIGNED_UNION(32, alpha_q_re, 1024);
INTFLOAT_ALIGNED_UNION(32, alpha_q_im, 1024);
INTFLOAT_ALIGNED_UNION(32, prev_alpha_q_re, 1024);
INTFLOAT_ALIGNED_UNION(32, prev_alpha_q_im, 1024);
INTFLOAT_ALIGNED_UNION(32, dmix_re, 1024);
INTFLOAT_ALIGNED_UNION(32, prev_dmix_re, 1024); /* Recalculated on every frame */
INTFLOAT_ALIGNED_UNION(32, dmix_im, 1024); /* Final prediction data */
} AACUsacStereo;
/**
* channel element - generic struct for SCE/CPE/CCE/LFE
*/
typedef struct ChannelElement {
int present;
// CPE specific
uint8_t max_sfb_ste; ///< (USAC) Maximum of both max_sfb values
uint8_t ms_mask[128]; ///< Set if mid/side stereo is used for each scalefactor window band
// shared
SingleChannelElement ch[2];
// CCE specific
ChannelCoupling coup;
// USAC stereo coupling data
AACUsacStereo us;
} ChannelElement;
typedef struct AACUSACLoudnessInfo {
uint8_t drc_set_id : 6;
uint8_t downmix_id : 7;
struct {
uint16_t lvl : 12;
uint8_t present : 1;
} sample_peak;
struct {
uint16_t lvl : 12;
uint8_t measurement : 4;
uint8_t reliability : 2;
uint8_t present : 1;
} true_peak;
uint8_t nb_measurements : 4;
struct {
uint8_t method_def : 4;
uint8_t method_val;
uint8_t measurement : 4;
uint8_t reliability : 2;
} measurements[16];
} AACUSACLoudnessInfo;
typedef struct AACUsacElemConfig {
enum AACUsacElem type;
uint8_t tw_mdct : 1;
uint8_t noise_fill : 1;
uint8_t stereo_config_index;
struct {
int ratio;
uint8_t harmonic_sbr : 1; /* harmonicSBR */
uint8_t bs_intertes : 1; /* bs_interTes */
uint8_t bs_pvc : 1; /* bs_pvc */
struct {
uint8_t start_freq; /* dflt_start_freq */
uint8_t stop_freq; /* dflt_stop_freq */
uint8_t freq_scale; /* dflt_freq_scale */
uint8_t alter_scale : 1; /* dflt_alter_scale */
uint8_t noise_scale; /* dflt_noise_scale */
uint8_t limiter_bands; /* dflt_limiter_bands */
uint8_t limiter_gains; /* dflt_limiter_gains */
uint8_t interpol_freq : 1; /* dflt_interpol_freq */
uint8_t smoothing_mode : 1; /* dflt_smoothing_mode */
} dflt;
} sbr;
struct {
uint8_t freq_res; /* bsFreqRes */
uint8_t fixed_gain; /* bsFixedGainDMX */
uint8_t temp_shape_config; /* bsTempShapeConfig */
uint8_t decorr_config; /* bsDecorrConfig */
uint8_t high_rate_mode : 1; /* bsHighRateMode */
uint8_t phase_coding : 1; /* bsPhaseCoding */
uint8_t otts_bands_phase; /* bsOttBandsPhase */
uint8_t residual_coding; /* bsResidualCoding */
uint8_t residual_bands; /* bsResidualBands */
uint8_t pseudo_lr : 1; /* bsPseudoLr */
uint8_t env_quant_mode : 1; /* bsEnvQuantMode */
} mps;
struct {
enum AACUsacExtension type;
uint8_t payload_frag;
uint32_t default_len;
uint32_t pl_data_offset;
uint8_t *pl_data;
} ext;
} AACUsacElemConfig;
typedef struct AACUSACConfig {
uint8_t core_sbr_frame_len_idx; /* coreSbrFrameLengthIndex */
uint8_t rate_idx;
uint16_t core_frame_len;
uint16_t stream_identifier;
AACUsacElemConfig elems[64];
int nb_elems;
struct {
uint8_t nb_album;
AACUSACLoudnessInfo album_info[64];
uint8_t nb_info;
AACUSACLoudnessInfo info[64];
} loudness;
} AACUSACConfig;
typedef struct OutputConfiguration {
MPEG4AudioConfig m4ac;
uint8_t layout_map[MAX_ELEM_ID*4][3];
int layout_map_tags;
AVChannelLayout ch_layout;
enum OCStatus status;
AACUSACConfig usac;
} OutputConfiguration;
/**

208
libavcodec/aac/aacdec_ac.c Normal file
View File

@ -0,0 +1,208 @@
/*
* AAC definitions and structures
* Copyright (c) 2024 Lynne
*
* 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
*/
#include "libavcodec/aactab.h"
#include "aacdec_ac.h"
uint32_t ff_aac_ac_map_process(AACArithState *state, int reset, int N)
{
float ratio;
if (reset) {
memset(state->last, 0, sizeof(state->last));
state->last_len = N;
} else if (state->last_len != N) {
int i;
uint8_t last[512 /* 2048 / 4 */];
memcpy(last, state->last, sizeof(last));
ratio = state->last_len / (float)N;
for (i = 0; i < N/2; i++) {
int k = (int)(i * ratio);
state->last[i] = last[k];
}
for (; i < FF_ARRAY_ELEMS(state->last); i++)
state->last[i] = 0;
state->last_len = N;
}
state->cur[3] = 0;
state->cur[2] = 0;
state->cur[1] = 0;
state->cur[0] = 1;
state->state_pre = state->last[0] << 12;
return state->last[0] << 12;
}
uint32_t ff_aac_ac_get_context(AACArithState *state, uint32_t c, int i, int N)
{
c = state->state_pre >> 8;
c = c + (state->last[i + 1] << 8);
c = (c << 4);
c += state->cur[1];
state->state_pre = c;
if (i > 3 &&
((state->cur[3] + state->cur[2] + state->cur[1]) < 5))
return c + 0x10000;
return c;
}
uint32_t ff_aac_ac_get_pk(uint32_t c)
{
int i_min = -1;
int i, j;
int i_max = FF_ARRAY_ELEMS(ff_aac_ac_lookup_m) - 1;
while ((i_max - i_min) > 1) {
i = i_min + ((i_max - i_min) / 2);
j = ff_aac_ac_hash_m[i];
if (c < (j >> 8))
i_max = i;
else if (c > (j >> 8))
i_min = i;
else
return (j & 0xFF);
}
return ff_aac_ac_lookup_m[i_max];
}
void ff_aac_ac_update_context(AACArithState *state, int idx,
uint16_t a, uint16_t b)
{
state->cur[0] = a + b + 1;
if (state->cur[0] > 0xF)
state->cur[0] = 0xF;
state->cur[3] = state->cur[2];
state->cur[2] = state->cur[1];
state->cur[1] = state->cur[0];
state->last[idx] = state->cur[0];
}
/* Initialize AC */
void ff_aac_ac_init(AACArith *ac, GetBitContext *gb)
{
ac->low = 0;
ac->high = UINT16_MAX;
ac->val = get_bits(gb, 16);
}
uint16_t ff_aac_ac_decode(AACArith *ac, GetBitContext *gb,
const uint16_t *cdf, uint16_t cdf_len)
{
int val = ac->val;
int low = ac->low;
int high = ac->high;
int sym;
int rng = high - low + 1;
int c = ((((int)(val - low + 1)) << 14) - ((int)1));
const uint16_t *p = cdf - 1;
/* One for each possible CDF length in the spec */
switch (cdf_len) {
case 2:
if ((p[1] * rng) > c)
p += 1;
break;
case 4:
if ((p[2] * rng) > c)
p += 2;
if ((p[1] * rng) > c)
p += 1;
break;
case 17:
/* First check if the current probability is even met at all */
if ((p[1] * rng) <= c)
break;
p += 1;
for (int i = 8; i >= 1; i >>= 1)
if ((p[i] * rng) > c)
p += i;
break;
case 27:
if ((p[16] * rng) > c)
p += 16;
if ((p[8] * rng) > c)
p += 8;
if (p != (cdf - 1 + 24))
if ((p[4] * rng) > c)
p += 4;
if ((p[2] * rng) > c)
p += 2;
if (p != (cdf - 1 + 24 + 2))
if ((p[1] * rng) > c)
p += 1;
break;
default:
/* This should never happen */
av_assert2(0);
}
sym = (int)((ptrdiff_t)(p - cdf)) + 1;
if (sym)
high = low + ((rng * cdf[sym - 1]) >> 14) - 1;
low += (rng * cdf[sym]) >> 14;
/* This loop could be done faster */
while (1) {
if (high < 32768) {
;
} else if (low >= 32768) {
val -= 32768;
low -= 32768;
high -= 32768;
} else if (low >= 16384 && high < 49152) {
val -= 16384;
low -= 16384;
high -= 16384;
} else {
break;
}
low += low;
high += high + 1;
val = (val << 1) | get_bits1(gb);
};
ac->low = low;
ac->high = high;
ac->val = val;
return sym;
}
void ff_aac_ac_finish(AACArithState *state, int offset, int N)
{
int i;
for (i = offset; i < N/2; i++)
state->last[i] = 1;
for (; i < FF_ARRAY_ELEMS(state->last); i++)
state->last[i] = 0;
}

54
libavcodec/aac/aacdec_ac.h Normal file
View File

@ -0,0 +1,54 @@
/*
* AAC definitions and structures
* Copyright (c) 2024 Lynne
*
* 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
*/
#ifndef AVCODEC_AAC_AACDEC_AC_H
#define AVCODEC_AAC_AACDEC_AC_H
#include "libavcodec/get_bits.h"
typedef struct AACArithState {
uint8_t last[512 /* 2048 / 4 */];
int last_len;
uint8_t cur[4];
uint16_t state_pre;
} AACArithState;
typedef struct AACArith {
uint16_t low;
uint16_t high;
uint16_t val;
} AACArith;
#define FF_AAC_AC_ESCAPE 16
uint32_t ff_aac_ac_map_process(AACArithState *state, int reset, int len);
uint32_t ff_aac_ac_get_context(AACArithState *state, uint32_t old_c, int idx, int len);
uint32_t ff_aac_ac_get_pk(uint32_t c);
void ff_aac_ac_update_context(AACArithState *state, int idx, uint16_t a, uint16_t b);
void ff_aac_ac_init(AACArith *ac, GetBitContext *gb);
uint16_t ff_aac_ac_decode(AACArith *ac, GetBitContext *gb,
const uint16_t *cdf, uint16_t cdf_len);
void ff_aac_ac_finish(AACArithState *state, int offset, int nb);
#endif /* AVCODEC_AACDEC_AC_H */

4
libavcodec/aac/aacdec_dsp_template.c
View File

@ -88,8 +88,8 @@ static void AAC_RENAME(apply_mid_side_stereo)(AACDecContext *ac, ChannelElement
INTFLOAT *ch1 = cpe->ch[1].AAC_RENAME(coeffs);
const uint16_t *offsets = ics->swb_offset;
for (int g = 0; g < ics->num_window_groups; g++) {
for (int sfb = 0; sfb < ics->max_sfb; sfb++) {
const int idx = g*ics->max_sfb + sfb;
for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb++) {
const int idx = g*cpe->max_sfb_ste + sfb;
if (cpe->ms_mask[idx] &&
cpe->ch[0].band_type[idx] < NOISE_BT &&
cpe->ch[1].band_type[idx] < NOISE_BT) {

14
libavcodec/aac/aacdec_latm.h
View File

@ -56,7 +56,8 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
{
AACDecContext *ac = &latmctx->aac_ctx;
AVCodecContext *avctx = ac->avctx;
MPEG4AudioConfig m4ac = { 0 };
OutputConfiguration oc = { 0 };
MPEG4AudioConfig *m4ac = &oc.m4ac;
GetBitContext gbc;
int config_start_bit = get_bits_count(gb);
int sync_extension = 0;
@ -76,7 +77,7 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
if (get_bits_left(gb) <= 0)
return AVERROR_INVALIDDATA;
bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &m4ac,
bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &oc,
&gbc, config_start_bit,
sync_extension);
@ -88,11 +89,12 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
asclen = bits_consumed;
if (!latmctx->initialized ||
ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
ac->oc[1].m4ac.chan_config != m4ac.chan_config) {
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);
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");
}
@ -280,7 +282,7 @@ static int latm_decode_frame(AVCodecContext *avctx, AVFrame *out,
} else {
push_output_configuration(&latmctx->aac_ctx);
if ((err = decode_audio_specific_config(
&latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac,
&latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1],
avctx->extradata, avctx->extradata_size*8LL, 1)) < 0) {
pop_output_configuration(&latmctx->aac_ctx);
return err;

198
libavcodec/aac/aacdec_lpd.c Normal file
View File

@ -0,0 +1,198 @@
/*
* Copyright (c) 2024 Lynne <dev@lynne.ee>
*
* 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
*/
#include "aacdec_lpd.h"
#include "aacdec_usac.h"
#include "libavcodec/unary.h"
const uint8_t ff_aac_lpd_mode_tab[32][4] = {
{ 0, 0, 0, 0 },
{ 1, 0, 0, 0 },
{ 0, 1, 0, 0 },
{ 1, 1, 0, 0 },
{ 0, 0, 1, 0 },
{ 1, 0, 1, 0 },
{ 0, 1, 1, 0 },
{ 1, 1, 1, 0 },
{ 0, 0, 0, 1 },
{ 1, 0, 0, 1 },
{ 0, 1, 0, 1 },
{ 1, 1, 0, 1 },
{ 0, 0, 1, 1 },
{ 1, 0, 1, 1 },
{ 0, 1, 1, 1 },
{ 1, 1, 1, 1 },
{ 2, 2, 0, 0 },
{ 2, 2, 1, 0 },
{ 2, 2, 0, 1 },
{ 2, 2, 1, 1 },
{ 0, 0, 2, 2 },
{ 1, 0, 2, 2 },
{ 0, 1, 2, 2 },
{ 1, 1, 2, 2 },
{ 2, 2, 2, 2 },
{ 3, 3, 3, 3 },
/* Larger values are reserved, but permit them for resilience */
{ 0, 0, 0, 0 },
{ 0, 0, 0, 0 },
{ 0, 0, 0, 0 },
{ 0, 0, 0, 0 },
{ 0, 0, 0, 0 },
{ 0, 0, 0, 0 },
};
static void parse_qn(GetBitContext *gb, int *qn, int nk_mode, int no_qn)
{
if (nk_mode == 1) {
for (int k = 0; k < no_qn; k++) {
qn[k] = get_unary(gb, 0, INT32_MAX); // TODO: find proper ranges
if (qn[k])
qn[k]++;
}
return;
}
for (int k = 0; k < no_qn; k++)
qn[k] = get_bits(gb, 2) + 2;
if (nk_mode == 2) {
for (int k = 0; k < no_qn; k++) {
if (qn[k] > 4) {
qn[k] = get_unary(gb, 0, INT32_MAX);;
if (qn[k])
qn[k] += 4;
}
}
return;
}
for (int k = 0; k < no_qn; k++) {
if (qn[k] > 4) {
int qn_ext = get_unary(gb, 0, INT32_MAX);;
switch (qn_ext) {
case 0: qn[k] = 5; break;
case 1: qn[k] = 6; break;
case 2: qn[k] = 0; break;
default: qn[k] = qn_ext + 4; break;
}
}
}
}
static int parse_codebook_idx(GetBitContext *gb, uint32_t *kv,
int nk_mode, int no_qn)
{
int idx, n, nk;
int qn[2];
parse_qn(gb, qn, nk_mode, no_qn);
for (int k = 0; k < no_qn; k++) {
if (qn[k] > 4) {
nk = (qn[k] - 3) / 2;
n = qn[k] - nk*2;
} else {
nk = 0;
n = qn[k];
}
}
idx = get_bits(gb, 4*n);
if (nk > 0)
for (int i = 0; i < 8; i++)
kv[i] = get_bits(gb, nk);
return 0;
}
int ff_aac_parse_fac_data(AACUsacElemData *ce, GetBitContext *gb,
int use_gain, int len)
{
int ret;
if (use_gain)
ce->fac.gain = get_bits(gb, 7);
for (int i = 0; i < len/8; i++) {
ret = parse_codebook_idx(gb, ce->fac.kv[i], 1, 1);
if (ret < 0)
return ret;
}
return 0;
}
int ff_aac_ldp_parse_channel_stream(AACDecContext *ac, AACUSACConfig *usac,
AACUsacElemData *ce, GetBitContext *gb)
{
int k;
const uint8_t *mod;
int first_ldp_flag;
int first_tcx_flag;
ce->ldp.acelp_core_mode = get_bits(gb, 3);
ce->ldp.lpd_mode = get_bits(gb, 5);
ce->ldp.bpf_control_info = get_bits1(gb);
ce->ldp.core_mode_last = get_bits1(gb);
ce->ldp.fac_data_present = get_bits1(gb);
mod = ff_aac_lpd_mode_tab[ce->ldp.lpd_mode];
first_ldp_flag = !ce->ldp.core_mode_last;
first_tcx_flag = 1;
if (first_ldp_flag)
ce->ldp.last_lpd_mode = -1; /* last_ldp_mode is a **STATEFUL** value */
k = 0;
while (k < 0) {
if (!k) {
if (ce->ldp.core_mode_last && ce->ldp.fac_data_present)
ff_aac_parse_fac_data(ce, gb, 0, usac->core_frame_len/8);
} else {
if (!ce->ldp.last_lpd_mode && mod[k] > 0 ||
ce->ldp.last_lpd_mode && !mod[k])
ff_aac_parse_fac_data(ce, gb, 0, usac->core_frame_len/8);
}
if (!mod[k]) {
// parse_acelp_coding();
ce->ldp.last_lpd_mode = 0;
k++;
} else {
// parse_tcx_coding();
ce->ldp.last_lpd_mode = mod[k];
k += (1 << (mod[k] - 1));
first_tcx_flag = 0;
}
}
// parse_lpc_data(first_lpd_flag);
if (!ce->ldp.core_mode_last && ce->ldp.fac_data_present) {
uint16_t len_8 = usac->core_frame_len / 8;
uint16_t len_16 = usac->core_frame_len / 16;
uint16_t fac_len = get_bits1(gb) /* short_fac_flag */ ? len_8 : len_16;
int ret = ff_aac_parse_fac_data(ce, gb, 1, fac_len);
if (ret < 0)
return ret;
}
return 0;
}

33
libavcodec/aac/aacdec_lpd.h Normal file
View File

@ -0,0 +1,33 @@
/*
* Copyright (c) 2024 Lynne <dev@lynne.ee>
*
* 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
*/
#ifndef AVCODEC_AAC_AACDEC_LPD_H
#define AVCODEC_AAC_AACDEC_LPD_H
#include "aacdec.h"
#include "libavcodec/get_bits.h"
int ff_aac_parse_fac_data(AACUsacElemData *ce, GetBitContext *gb,
int use_gain, int len);
int ff_aac_ldp_parse_channel_stream(AACDecContext *ac, AACUSACConfig *usac,
AACUsacElemData *ce, GetBitContext *gb);
#endif /* AVCODEC_AAC_AACDEC_LPD_H */

1608
libavcodec/aac/aacdec_usac.c Normal file
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File diff suppressed because it is too large Load Diff

37
libavcodec/aac/aacdec_usac.h Normal file
View File

@ -0,0 +1,37 @@
/*
* Copyright (c) 2024 Lynne <dev@lynne.ee>
*
* 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
*/
#ifndef AVCODEC_AAC_AACDEC_USAC_H
#define AVCODEC_AAC_AACDEC_USAC_H
#include "aacdec.h"
#include "libavcodec/get_bits.h"
int ff_aac_usac_config_decode(AACDecContext *ac, AVCodecContext *avctx,
GetBitContext *gb, OutputConfiguration *oc,
int channel_config);
int ff_aac_usac_reset_state(AACDecContext *ac, OutputConfiguration *oc);
int ff_aac_usac_decode_frame(AVCodecContext *avctx, AACDecContext *ac,
GetBitContext *gb, int *got_frame_ptr);
#endif /* AVCODEC_AAC_AACDEC_USAC_H */

42
libavcodec/aactab.c
View File

@ -1998,6 +1998,11 @@ const uint8_t ff_tns_max_bands_128[] = {
};
// @}
const uint8_t ff_usac_noise_fill_start_offset[2][2] = {
{ 160, 20 },
{ 120, 15 },
};
const DECLARE_ALIGNED(32, float, ff_aac_eld_window_512)[1920] = {
0.00338834, 0.00567745, 0.00847677, 0.01172641,
0.01532555, 0.01917664, 0.02318809, 0.02729259,
@ -3895,3 +3900,40 @@ DECLARE_ALIGNED(16, const float, ff_aac_deemph_weights)[16] = {
0,
USAC_EMPH_COEFF,
};
const int ff_aac_usac_samplerate[32] = {
96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
16000, 12000, 11025, 8000, 7350, -1, -1, 57600,
51200, 40000, 38400, 34150, 28800, 25600, 20000, 19200,
17075, 14400, 12800, 9600, -1, -1, -1, -1,
};
/* Window type (only long+eight, start/stop/stopstart), sine+sine, kbd+kbd, sine+kbd, kbd+sine */
const float ff_aac_usac_mdst_filt_cur[4 /* Window */][4 /* Shape */][7] =
{
{ { 0.000000, 0.000000, 0.500000, 0.000000, -0.500000, 0.000000, 0.000000 },
{ 0.091497, 0.000000, 0.581427, 0.000000, -0.581427, 0.000000, -0.091497 },
{ 0.045748, 0.057238, 0.540714, 0.000000, -0.540714, -0.057238, -0.045748 },
{ 0.045748, -0.057238, 0.540714, 0.000000, -0.540714, 0.057238, -0.045748 } },
{ { 0.102658, 0.103791, 0.567149, 0.000000, -0.567149, -0.103791, -0.102658 },
{ 0.150512, 0.047969, 0.608574, 0.000000, -0.608574, -0.047969, -0.150512 },
{ 0.104763, 0.105207, 0.567861, 0.000000, -0.567861, -0.105207, -0.104763 },
{ 0.148406, 0.046553, 0.607863, 0.000000, -0.607863, -0.046553, -0.148406 } },
{ { 0.102658, -0.103791, 0.567149, 0.000000, -0.567149, 0.103791, -0.102658 },
{ 0.150512, -0.047969, 0.608574, 0.000000, -0.608574, 0.047969, -0.150512 },
{ 0.148406, -0.046553, 0.607863, 0.000000, -0.607863, 0.046553, -0.148406 },
{ 0.104763, -0.105207, 0.567861, 0.000000, -0.567861, 0.105207, -0.104763 } },
{ { 0.205316, 0.000000, 0.634298, 0.000000, -0.634298, 0.000000, -0.205316 },
{ 0.209526, 0.000000, 0.635722, 0.000000, -0.635722, 0.000000, -0.209526 },
{ 0.207421, 0.001416, 0.635010, 0.000000, -0.635010, -0.001416, -0.207421 },
{ 0.207421, -0.001416, 0.635010, 0.000000, -0.635010, 0.001416, -0.207421 } }
};
/* Window type (everything/longstop+stopstart), sine or kbd */
const float ff_aac_usac_mdst_filt_prev[2 /* Window */][2 /* sine/kbd */][7] =
{
{ { 0.000000, 0.106103, 0.250000, 0.318310, 0.250000, 0.106103, 0.000000 },
{ 0.059509, 0.123714, 0.186579, 0.213077, 0.186579, 0.123714, 0.059509 } },
{ { 0.038498, 0.039212, 0.039645, 0.039790, 0.039645, 0.039212, 0.038498 },
{ 0.026142, 0.026413, 0.026577, 0.026631, 0.026577, 0.026413, 0.026142 } }
};

10
libavcodec/aactab.h
View File

@ -115,4 +115,14 @@ extern const uint8_t ff_tns_max_bands_512 [13];
extern const uint8_t ff_tns_max_bands_480 [13];
extern const uint8_t ff_tns_max_bands_128 [13];
/* [x][y], x == 1 -> frame len is 768 frames, y == 1 -> is eight_short */
extern const uint8_t ff_usac_noise_fill_start_offset[2][2];
extern const int ff_aac_usac_samplerate[32];
/* Window type (only long+eight, start/stop/stopstart), sine+sine, kbd+kbd, sine+kbd, kbd+sine */
extern const float ff_aac_usac_mdst_filt_cur[4 /* Window */][4 /* Shape */][7];
/* Window type (everything/longstop+stopstart), sine or kbd */
extern const float ff_aac_usac_mdst_filt_prev[2 /* Window */][2 /* sine/kbd */][7];
#endif /* AVCODEC_AACTAB_H */