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aac_ltp: split, reorder and improve prediction algorithm 

This commit attempts to mirror what the decoder does more closely
in addition to fixing some shortcomings.
This commit is contained in:
Rostislav Pehlivanov 2015-11-26 17:40:04 +00:00
parent a239ce7074
commit 1e5dbb3409
2 changed files with 38 additions and 29 deletions
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65
libavcodec/aacenc_ltp.c
View File

@ -72,29 +72,17 @@ void ff_aac_ltp_insert_new_frame(AACEncContext *s)
}
}
/**
* Process LTP parameters
* @see Patent WO2006070265A1
*/
void ff_aac_update_ltp(AACEncContext *s, SingleChannelElement *sce)
static void get_lag(float *buf, const float *new, LongTermPrediction *ltp)
{
int i, j, lag, samples_num;
float corr, max_ratio, max_corr;
float *pred_signal = &sce->ltp_state[0];
const float *samples = &s->planar_samples[s->cur_channel][1024];
if (s->profile != FF_PROFILE_AAC_LTP)
return;
/* Calculate lag */
max_corr = 0.0f;
int i, j, lag, max_corr = 0;
float max_ratio;
for (i = 0; i < 2048; i++) {
float s0 = 0.0f, s1 = 0.0f;
float corr, s0 = 0.0f, s1 = 0.0f;
const int start = FFMAX(0, i - 1024);
for (j = start; j < 2048; j++) {
const int idx = j - i + 1024;
s0 += samples[j]*pred_signal[idx];
s1 += pred_signal[idx]*pred_signal[idx];
s0 += new[j]*buf[idx];
s1 += buf[idx]*buf[idx];
}
corr = s1 > 0.0f ? s0/sqrt(s1) : 0.0f;
if (corr > max_corr) {
@ -103,19 +91,40 @@ void ff_aac_update_ltp(AACEncContext *s, SingleChannelElement *sce)
max_ratio = corr/(2048-start);
}
}
ltp->lag = FFMAX(av_clip_uintp2(lag, 11), 0);
ltp->coef_idx = quant_array_idx(max_ratio, ltp_coef, 8);
ltp->coef = ltp_coef[ltp->coef_idx];
}
if (lag < 1)
static void generate_samples(float *buf, LongTermPrediction *ltp)
{
int i, samples_num = 2048;
if (!ltp->lag) {
ltp->present = 0;
return;
} else if (ltp->lag < 1024) {
samples_num = ltp->lag + 1024;
}
for (i = 0; i < samples_num; i++)
buf[i] = ltp->coef*buf[i + 2048 - ltp->lag];
memset(&buf[i], 0, (2048 - i)*sizeof(float));
}
/**
* Process LTP parameters
* @see Patent WO2006070265A1
*/
void ff_aac_update_ltp(AACEncContext *s, SingleChannelElement *sce)
{
float *pred_signal = &sce->ltp_state[0];
const float *samples = &s->planar_samples[s->cur_channel][1024];
if (s->profile != FF_PROFILE_AAC_LTP)
return;
sce->ics.ltp.lag = lag = av_clip_uintp2(lag, 11);
sce->ics.ltp.coef_idx = quant_array_idx(max_ratio, ltp_coef, 8);
sce->ics.ltp.coef = ltp_coef[sce->ics.ltp.coef_idx];
/* Predict the new samples */
samples_num = 1024 + (lag < 1024 ? lag : 1024);
for (i = 1024; i < samples_num + 1024; i++)
pred_signal[i] = sce->ics.ltp.coef*pred_signal[i-lag];
memset(&pred_signal[samples_num], 0, (2048 - samples_num)*sizeof(float));
/* Calculate lag */
get_lag(pred_signal, samples, &sce->ics.ltp);
generate_samples(pred_signal, &sce->ics.ltp);
}
void ff_aac_adjust_common_ltp(AACEncContext *s, ChannelElement *cpe)

2
tests/fate/aac.mak
View File

@ -209,7 +209,7 @@ fate-aac-ltp-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-re
fate-aac-ltp-encode: CMP = stddev
fate-aac-ltp-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
fate-aac-ltp-encode: CMP_SHIFT = -4096
fate-aac-ltp-encode: CMP_TARGET = 1535
fate-aac-ltp-encode: CMP_TARGET = 1120
fate-aac-ltp-encode: SIZE_TOLERANCE = 3560
fate-aac-ltp-encode: FUZZ = 17