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ffmpeg.wasm-core
Commit 848a5815 authored by Robert Swain's avatar Robert Swain
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Synchronise AAC decoder code with that from SoC 

Originally committed as revision 14772 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent 2272e04a
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Showing with 506 additions and 459 deletions
libavcodec/aac.c
View file @ 848a5815
......@@ -91,7 +91,6 @@
#include <string.h>
#ifndef CONFIG_HARDCODED_TABLES
static float ff_aac_ivquant_tab[IVQUANT_SIZE];
static float ff_aac_pow2sf_tab[316];
#endif /* CONFIG_HARDCODED_TABLES */
......@@ -403,8 +402,6 @@ static av_cold int aac_decode_init(AVCodecContext * avccontext) {
}
#ifndef CONFIG_HARDCODED_TABLES
for (i = 1 - IVQUANT_SIZE/2; i < IVQUANT_SIZE/2; i++)
ff_aac_ivquant_tab[i + IVQUANT_SIZE/2 - 1] = cbrt(fabs(i)) * i;
for (i = 0; i < 316; i++)
ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
#endif /* CONFIG_HARDCODED_TABLES */
......@@ -468,19 +465,6 @@ static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBi
return 0;
}
/**
* inverse quantization
*
* @param a quantized value to be dequantized
* @return Returns dequantized value.
*/
static inline float ivquant(int a) {
if (a + (unsigned int)IVQUANT_SIZE/2 - 1 < (unsigned int)IVQUANT_SIZE - 1)
return ff_aac_ivquant_tab[a + IVQUANT_SIZE/2 - 1];
else
return cbrtf(fabsf(a)) * a;
}
/**
* Decode band types (section_data payload); reference: table 4.46.
*
......@@ -535,7 +519,6 @@ static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * g
int offset[3] = { global_gain, global_gain - 90, 100 };
int noise_flag = 1;
static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
ics->intensity_present = 0;
for (g = 0; g < ics->num_window_groups; g++) {
for (i = 0; i < ics->max_sfb;) {
int run_end = band_type_run_end[idx];
......@@ -543,7 +526,6 @@ static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * g
for(; i < run_end; i++, idx++)
sf[idx] = 0.;
}else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
ics->intensity_present = 1;
for(; i < run_end; i++, idx++) {
offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
if(offset[2] > 255U) {
......@@ -585,13 +567,14 @@ static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * g
/**
* Decode pulse data; reference: table 4.7.
*/
static void decode_pulses(Pulse * pulse, GetBitContext * gb) {
static void decode_pulses(Pulse * pulse, GetBitContext * gb, const uint16_t * swb_offset) {
int i;
pulse->num_pulse = get_bits(gb, 2) + 1;
pulse->start = get_bits(gb, 6);
for (i = 0; i < pulse->num_pulse; i++) {
pulse->offset[i] = get_bits(gb, 5);
pulse->amp [i] = get_bits(gb, 4);
pulse->pos[0] = get_bits(gb, 5) + swb_offset[get_bits(gb, 6)];
pulse->amp[0] = get_bits(gb, 4);
for (i = 1; i < pulse->num_pulse; i++) {
pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i-1];
pulse->amp[i] = get_bits(gb, 4);
}
}
......@@ -613,22 +596,6 @@ static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb,
}
}
/**
* Add pulses with particular amplitudes to the quantized spectral data; reference: 4.6.3.3.
*
* @param pulse pointer to pulse data struct
* @param icoef array of quantized spectral data
*/
static void add_pulses(int icoef[1024], const Pulse * pulse, const IndividualChannelStream * ics) {
int i, off = ics->swb_offset[pulse->start];
for (i = 0; i < pulse->num_pulse; i++) {
int ic;
off += pulse->offset[i];
ic = (icoef[off] - 1)>>31;
icoef[off] += (pulse->amp[i]^ic) - ic;
}
}
/**
* Decode an individual_channel_stream payload; reference: table 4.44.
*
......@@ -638,18 +605,16 @@ static void add_pulses(int icoef[1024], const Pulse * pulse, const IndividualCha
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) {
int icoeffs[1024];
Pulse pulse;
TemporalNoiseShaping * tns = &sce->tns;
IndividualChannelStream * ics = &sce->ics;
float * out = sce->coeffs;
int global_gain, pulse_present = 0;
/* These two assignments are to silence some GCC warnings about the
* variables being used uninitialised when in fact they always are.
/* This assignment is to silence a GCC warning about the variable being used
* uninitialized when in fact it always is.
*/
pulse.num_pulse = 0;
pulse.start = 0;
global_gain = get_bits(gb, 8);
......@@ -670,7 +635,7 @@ static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext
av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
return -1;
}
decode_pulses(&pulse, gb);
decode_pulses(&pulse, gb, ics->swb_offset);
}
if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
return -1;
......@@ -680,11 +645,8 @@ static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext
}
}
if (decode_spectrum(ac, icoeffs, gb, ics, sce->band_type) < 0)
if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
return -1;
if (pulse_present)
add_pulses(icoeffs, &pulse, ics);
dequant(ac, out, icoeffs, sce->sf, ics, sce->band_type);
return 0;
}
......@@ -722,8 +684,7 @@ static int decode_cpe(AACContext * ac, GetBitContext * gb, int elem_id) {
if (common_window && ms_present)
apply_mid_side_stereo(cpe);
if (cpe->ch[1].ics.intensity_present)
apply_intensity_stereo(cpe, ms_present);
apply_intensity_stereo(cpe, ms_present);
return 0;
}
......@@ -906,10 +867,10 @@ static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) {
float * in = sce->coeffs;
float * out = sce->ret;
float * saved = sce->saved;
const float * lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024;
const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128;
const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024;
const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128;
const float * lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
float * buf = ac->buf_mdct;
int i;
......@@ -1093,7 +1054,7 @@ static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data
if (!ac->is_saved) {
ac->is_saved = 1;
*data_size = 0;
return 0;
return buf_size;
}
data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t);
......
libavcodec/aac.h
View file @ 848a5815
......@@ -45,8 +45,6 @@
#define MAX_CHANNELS 64
#define MAX_ELEM_ID 16
#define IVQUANT_SIZE 1024
enum AudioObjectType {
AOT_NULL,
// Support? Name
......@@ -165,8 +163,7 @@ typedef struct {
typedef struct {
int num_pulse;
int start;
int offset[4];
int pos[4];
int amp[4];
} Pulse;
......
libavcodec/aactab.c
View file @ 848a5815
This diff is collapsed.
libavcodec/aactab.h
View file @ 848a5815
......@@ -54,10 +54,9 @@ extern const uint16_t *ff_aac_spectral_codes[11];
extern const uint8_t *ff_aac_spectral_bits [11];
extern const uint16_t ff_aac_spectral_sizes[11];
extern const int8_t *ff_aac_codebook_vectors[];
extern const float *ff_aac_codebook_vectors[];
#ifdef CONFIG_HARDCODED_TABLES
extern const float ff_aac_ivquant_tab[IVQUANT_SIZE];
extern const float ff_aac_pow2sf_tab[316];
#endif /* CONFIG_HARDCODED_TABLES */
......
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