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aacdec: Implement LTP support.

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Ported from gsoc svn.
This commit is contained in:
Young Han Lee 2011-02-14 18:09:43 +09:00 committed by Alex Converse
parent 77c330a046
commit ead15f1dc1
4 changed files with 178 additions and 20 deletions
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32
libavcodec/aac.h
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@ -43,6 +43,7 @@
#define MAX_ELEM_ID 16
#define TNS_MAX_ORDER 20
#define MAX_LTP_LONG_SFB 40
enum RawDataBlockType {
TYPE_SCE,
@ -130,6 +131,16 @@ typedef struct {
#define SCALE_MAX_DIFF 60 ///< maximum scalefactor difference allowed by standard
#define SCALE_DIFF_ZERO 60 ///< codebook index corresponding to zero scalefactor indices difference
/**
* Long Term Prediction
*/
typedef struct {
int8_t present;
int16_t lag;
float coef;
int8_t used[MAX_LTP_LONG_SFB];
} LongTermPrediction;
/**
* Individual Channel Stream
*/
@ -139,6 +150,7 @@ typedef struct {
uint8_t use_kb_window[2]; ///< If set, use Kaiser-Bessel window, otherwise use a sinus window.
int num_window_groups;
uint8_t group_len[8];
LongTermPrediction ltp;
const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window
const uint8_t *swb_sizes; ///< table of scalefactor band sizes for a particular window
int num_swb; ///< number of scalefactor window bands
@ -206,14 +218,15 @@ typedef struct {
IndividualChannelStream ics;
TemporalNoiseShaping tns;
Pulse pulse;
enum BandType band_type[128]; ///< band types
int band_type_run_end[120]; ///< band type run end points
float sf[120]; ///< scalefactors
int sf_idx[128]; ///< scalefactor indices (used by encoder)
uint8_t zeroes[128]; ///< band is not coded (used by encoder)
DECLARE_ALIGNED(16, float, coeffs)[1024]; ///< coefficients for IMDCT
DECLARE_ALIGNED(16, float, saved)[1024]; ///< overlap
DECLARE_ALIGNED(16, float, ret)[2048]; ///< PCM output
enum BandType band_type[128]; ///< band types
int band_type_run_end[120]; ///< band type run end points
float sf[120]; ///< scalefactors
int sf_idx[128]; ///< scalefactor indices (used by encoder)
uint8_t zeroes[128]; ///< band is not coded (used by encoder)
DECLARE_ALIGNED(16, float, coeffs)[1024]; ///< coefficients for IMDCT
DECLARE_ALIGNED(16, float, saved)[1024]; ///< overlap
DECLARE_ALIGNED(16, float, ret)[2048]; ///< PCM output
DECLARE_ALIGNED(16, int16_t, ltp_state)[3072]; ///< time signal for LTP
PredictorState predictor_state[MAX_PREDICTORS];
} SingleChannelElement;
@ -259,7 +272,7 @@ typedef struct {
* @defgroup temporary aligned temporary buffers (We do not want to have these on the stack.)
* @{
*/
DECLARE_ALIGNED(16, float, buf_mdct)[1024];
DECLARE_ALIGNED(16, float, buf_mdct)[2048];
/** @} */
/**
@ -268,6 +281,7 @@ typedef struct {
*/
FFTContext mdct;
FFTContext mdct_small;
FFTContext mdct_ltp;
DSPContext dsp;
FmtConvertContext fmt_conv;
int random_state;

156
libavcodec/aacdec.c
View File

@ -42,7 +42,7 @@
* Y filterbank - standard
* N (code in SoC repo) filterbank - Scalable Sample Rate
* Y Temporal Noise Shaping
* N (code in SoC repo) Long Term Prediction
* Y Long Term Prediction
* Y intensity stereo
* Y channel coupling
* Y frequency domain prediction
@ -478,6 +478,7 @@ static int decode_audio_specific_config(AACContext *ac,
switch (m4ac->object_type) {
case AOT_AAC_MAIN:
case AOT_AAC_LC:
case AOT_AAC_LTP:
if (decode_ga_specific_config(ac, avctx, &gb, m4ac, m4ac->chan_config))
return -1;
break;
@ -580,8 +581,9 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
352);
ff_mdct_init(&ac->mdct, 11, 1, 1.0);
ff_mdct_init(&ac->mdct_small, 8, 1, 1.0);
ff_mdct_init(&ac->mdct, 11, 1, 1.0);
ff_mdct_init(&ac->mdct_small, 8, 1, 1.0);
ff_mdct_init(&ac->mdct_ltp, 11, 0, 1.0);
// window initialization
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
@ -630,6 +632,20 @@ static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
return 0;
}
/**
* Decode Long Term Prediction data; reference: table 4.xx.
*/
static void decode_ltp(AACContext *ac, LongTermPrediction *ltp,
GetBitContext *gb, uint8_t max_sfb)
{
int sfb;
ltp->lag = get_bits(gb, 11);
ltp->coef = ltp_coef[get_bits(gb, 3)] * ac->sf_scale;
for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
ltp->used[sfb] = get_bits1(gb);
}
/**
* Decode Individual Channel Stream info; reference: table 4.6.
*
@ -684,9 +700,8 @@ static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
memset(ics, 0, sizeof(IndividualChannelStream));
return -1;
} else {
av_log_missing_feature(ac->avctx, "Predictor bit set but LTP is", 1);
memset(ics, 0, sizeof(IndividualChannelStream));
return -1;
if ((ics->ltp.present = get_bits(gb, 1)))
decode_ltp(ac, &ics->ltp, gb, ics->max_sfb);
}
}
}
@ -1420,6 +1435,9 @@ static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
i = cpe->ch[1].ics.use_kb_window[0];
cpe->ch[1].ics = cpe->ch[0].ics;
cpe->ch[1].ics.use_kb_window[1] = i;
if (cpe->ch[1].ics.predictor_present && (ac->m4ac.object_type != AOT_AAC_MAIN))
if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
decode_ltp(ac, &cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
ms_present = get_bits(gb, 2);
if (ms_present == 3) {
av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
@ -1659,6 +1677,7 @@ static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
int w, filt, m, i;
int bottom, top, order, start, end, size, inc;
float lpc[TNS_MAX_ORDER];
float tmp[TNS_MAX_ORDER];
for (w = 0; w < ics->num_windows; w++) {
bottom = ics->num_swb;
@ -1684,14 +1703,118 @@ static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
}
start += w * 128;
// ar filter
for (m = 0; m < size; m++, start += inc)
for (i = 1; i <= FFMIN(m, order); i++)
coef[start] -= coef[start - i * inc] * lpc[i - 1];
if (decode) {
// ar filter
for (m = 0; m < size; m++, start += inc)
for (i = 1; i <= FFMIN(m, order); i++)
coef[start] -= coef[start - i * inc] * lpc[i - 1];
} else {
// ma filter
for (m = 0; m < size; m++, start += inc) {
tmp[0] = coef[start];
for (i = 1; i <= FFMIN(m, order); i++)
coef[start] += tmp[i] * lpc[i - 1];
for (i = order; i > 0; i--)
tmp[i] = tmp[i - 1];
}
}
}
}
}
/**
* Apply windowing and MDCT to obtain the spectral
* coefficient from the predicted sample by LTP.
*/
static void windowing_and_mdct_ltp(AACContext *ac, float *out,
float *in, IndividualChannelStream *ics)
{
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;
if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
ac->dsp.vector_fmul(in, in, lwindow_prev, 1024);
} else {
memset(in, 0, 448 * sizeof(float));
ac->dsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
memcpy(in + 576, in + 576, 448 * sizeof(float));
}
if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
ac->dsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
} else {
memcpy(in + 1024, in + 1024, 448 * sizeof(float));
ac->dsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
memset(in + 1024 + 576, 0, 448 * sizeof(float));
}
ff_mdct_calc(&ac->mdct_ltp, out, in);
}
/**
* Apply the long term prediction
*/
static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
{
const LongTermPrediction *ltp = &sce->ics.ltp;
const uint16_t *offsets = sce->ics.swb_offset;
int i, sfb;
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
float *predTime = ac->buf_mdct;
float *predFreq = sce->ret;
int16_t num_samples = 2048;
if (ltp->lag < 1024)
num_samples = ltp->lag + 1024;
for (i = 0; i < num_samples; i++)
predTime[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
memset(&predTime[i], 0, (2048 - i) * sizeof(float));
windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
if (sce->tns.present)
apply_tns(predFreq, &sce->tns, &sce->ics, 0);
for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
if (ltp->used[sfb])
for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
sce->coeffs[i] += predFreq[i];
}
}
/**
* Update the LTP buffer for next frame
*/
static void update_ltp(AACContext *ac, SingleChannelElement *sce)
{
IndividualChannelStream *ics = &sce->ics;
float *saved = sce->saved;
float *saved_ltp = sce->coeffs;
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;
int i;
for (i = 0; i < 512; i++)
ac->buf_mdct[1535 - i] = ac->buf_mdct[512 + i];
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
memcpy(saved_ltp, saved, 512 * sizeof(float));
memset(saved_ltp + 576, 0, 448 * sizeof(float));
ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, swindow, 128);
} else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(float));
memset(saved_ltp + 576, 0, 448 * sizeof(float));
ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, swindow, 128);
} else { // LONG_STOP or ONLY_LONG
ac->dsp.vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, lwindow, 1024);
}
memcpy(sce->ltp_state, &sce->ltp_state[1024], 1024 * sizeof(int16_t));
ac->fmt_conv.float_to_int16(&(sce->ltp_state[1024]), sce->ret, 1024);
ac->fmt_conv.float_to_int16(&(sce->ltp_state[2048]), saved_ltp, 1024);
}
/**
* Conduct IMDCT and windowing.
*/
@ -1857,6 +1980,14 @@ static void spectral_to_sample(AACContext *ac)
if (che) {
if (type <= TYPE_CPE)
apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
if (ac->m4ac.object_type == AOT_AAC_LTP) {
if (che->ch[0].ics.predictor_present) {
if (che->ch[0].ics.ltp.present)
apply_ltp(ac, &che->ch[0]);
if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
apply_ltp(ac, &che->ch[1]);
}
}
if (che->ch[0].tns.present)
apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
if (che->ch[1].tns.present)
@ -1865,8 +1996,12 @@ static void spectral_to_sample(AACContext *ac)
apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
imdct_and_windowing(ac, &che->ch[0]);
if (ac->m4ac.object_type == AOT_AAC_LTP)
update_ltp(ac, &che->ch[0]);
if (type == TYPE_CPE) {
imdct_and_windowing(ac, &che->ch[1]);
if (ac->m4ac.object_type == AOT_AAC_LTP)
update_ltp(ac, &che->ch[1]);
}
if (ac->m4ac.sbr > 0) {
ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
@ -2080,6 +2215,7 @@ static av_cold int aac_decode_close(AVCodecContext *avctx)
ff_mdct_end(&ac->mdct);
ff_mdct_end(&ac->mdct_small);
ff_mdct_end(&ac->mdct_ltp);
return 0;
}

8
libavcodec/aacdectab.h
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@ -35,6 +35,14 @@
#include <stdint.h>
/* @name ltp_coef
* Table of the LTP coefficient (multiplied by 2)
*/
static const float ltp_coef[8] = {
1.141658, 1.393232, 1.626008, 1.822608,
1.969800, 2.135788, 2.2389202, 2.739066,
};
/* @name tns_tmp2_map
* Tables of the tmp2[] arrays of LPC coefficients used for TNS.
* The suffix _M_N[] indicate the values of coef_compress and coef_res

2
libavcodec/mpeg4audio.h
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@ -57,7 +57,7 @@ enum AudioObjectType {
AOT_AAC_MAIN, ///< Y Main
AOT_AAC_LC, ///< Y Low Complexity
AOT_AAC_SSR, ///< N (code in SoC repo) Scalable Sample Rate
AOT_AAC_LTP, ///< N (code in SoC repo) Long Term Prediction
AOT_AAC_LTP, ///< Y Long Term Prediction
AOT_SBR, ///< Y Spectral Band Replication
AOT_AAC_SCALABLE, ///< N Scalable
AOT_TWINVQ, ///< N Twin Vector Quantizer