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-rw-r--r--libavcodec/aac.h32
-rw-r--r--libavcodec/aacdec.c156
-rw-r--r--libavcodec/aacdectab.h8
-rw-r--r--libavcodec/mpeg4audio.h2
4 files changed, 178 insertions, 20 deletions
diff --git a/libavcodec/aac.h b/libavcodec/aac.h
index cff476a6eb..a2bf70b001 100644
--- a/libavcodec/aac.h
+++ b/libavcodec/aac.h
@@ -43,6 +43,7 @@
#define MAX_ELEM_ID 16
#define TNS_MAX_ORDER 20
+#define MAX_LTP_LONG_SFB 40
enum RawDataBlockType {
TYPE_SCE,
@@ -131,6 +132,16 @@ typedef struct {
#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
*/
typedef struct {
@@ -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;
diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index bc92f5662f..ee5affebed 100644
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -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);
@@ -631,6 +633,20 @@ static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
}
/**
+ * 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.
*
* @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
@@ -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;
@@ -1684,15 +1703,119 @@ 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.
*/
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
@@ -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;
}
diff --git a/libavcodec/aacdectab.h b/libavcodec/aacdectab.h
index b4307f133c..500e8f2ad0 100644
--- a/libavcodec/aacdectab.h
+++ b/libavcodec/aacdectab.h
@@ -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
diff --git a/libavcodec/mpeg4audio.h b/libavcodec/mpeg4audio.h
index b94185079d..174624e15d 100644
--- a/libavcodec/mpeg4audio.h
+++ b/libavcodec/mpeg4audio.h
@@ -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