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aacenc_ltp: adjust and speed up autocorrelation calculations 

There were some errors in the calculation as well as an entire
unnecessary loop to find the gain coefficient. Merge the
two loops.
Thanks to @ubitux for the suggestions and testing.
This commit is contained in:
Rostislav Pehlivanov 2015-10-17 22:50:55 +01:00
parent 4ffdba2418
commit 7303962f14
2 changed files with 31 additions and 33 deletions
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60
libavcodec/aacenc_ltp.c
View File

@ -66,6 +66,7 @@ void ff_aac_ltp_insert_new_frame(AACEncContext *s)
memcpy(&sce->ltp_state[0], &sce->ltp_state[1024], 1024*sizeof(sce->ltp_state[0]));
memcpy(&sce->ltp_state[1024], &s->planar_samples[cur_channel][2048], 1024*sizeof(sce->ltp_state[0]));
memcpy(&sce->ltp_state[2048], &sce->ret_buf[0], 1024*sizeof(sce->ltp_state[0]));
sce->ics.ltp.lag = 0;
}
start_ch += chans;
}
@ -77,54 +78,44 @@ void ff_aac_ltp_insert_new_frame(AACEncContext *s)
*/
void ff_aac_update_ltp(AACEncContext *s, SingleChannelElement *sce)
{
int i, j, lag;
float corr, s0, s1, max_corr = 0.0f;
float *samples = &s->planar_samples[s->cur_channel][1024];
int i, j, lag, samples_num;
float corr, max_ratio, max_corr;
float *pred_signal = &sce->ltp_state[0];
int samples_num = 2048;
const float *samples = &s->planar_samples[s->cur_channel][1024];
if (s->profile != FF_PROFILE_AAC_LTP)
return;
/* Calculate lag */
for (i = 0; i < samples_num; i++) {
s0 = s1 = 0.0f;
for (j = 0; j < samples_num; j++) {
if (j + 1024 < i)
continue;
s0 += samples[j]*pred_signal[j-i+1024];
s1 += pred_signal[j-i+1024]*pred_signal[j-i+1024];
max_corr = 0.0f;
for (i = 0; i < 2048; i++) {
float 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];
}
corr = s1 > 0.0f ? s0/sqrt(s1) : 0.0f;
if (corr > max_corr) {
max_corr = corr;
lag = i;
max_ratio = corr/(2048-start);
}
}
lag = av_clip_uintp2(lag, 11); /* 11 bits => 2^11 = 0->2047 */
if (!lag) {
sce->ics.ltp.lag = lag;
if (lag < 1)
return;
}
s0 = s1 = 0.0f;
for (i = 0; i < lag; i++) {
s0 += samples[i];
s1 += pred_signal[i-lag+1024];
}
sce->ics.ltp.coef_idx = quant_array_idx(s0/s1, ltp_coef, 8);
sce->ics.ltp.coef = ltp_coef[sce->ics.ltp.coef_idx];
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 */
if (lag < 1024)
samples_num = lag + 1024;
for (i = 0; i < samples_num; i++)
pred_signal[i+1024] = sce->ics.ltp.coef*pred_signal[i-lag+1024];
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));
sce->ics.ltp.lag = lag;
}
void ff_aac_adjust_common_ltp(AACEncContext *s, ChannelElement *cpe)
@ -163,8 +154,15 @@ void ff_aac_search_for_ltp(AACEncContext *s, SingleChannelElement *sce,
float *PCD34 = &s->scoefs[128*2];
const int max_ltp = FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB);
if (sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE ||
!sce->ics.ltp.lag)
if (sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
if (sce->ics.ltp.lag) {
memset(&sce->lcoeffs[0], 0.0f, 3072*sizeof(sce->lcoeffs[0]));
memset(&sce->ics.ltp, 0, sizeof(LongTermPrediction));
}
return;
}
if (!sce->ics.ltp.lag)
return;
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {

4
tests/fate/aac.mak
View File

@ -205,11 +205,11 @@ fate-aac-ms-encode: SIZE_TOLERANCE = 3560
fate-aac-ms-encode: FUZZ = 10
FATE_AAC_ENCODE += fate-aac-ltp-encode
fate-aac-ltp-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -strict -2 -c:a aac -profile:a aac_ltp -aac_pns 0 -aac_is 0 -aac_ms 0 -aac_tns 0 -b:a 82k -cutoff 22050
fate-aac-ltp-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -strict -2 -c:a aac -profile:a aac_ltp -aac_pns 0 -aac_is 0 -aac_ms 0 -aac_tns 0 -b:a 36k
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 = 2370
fate-aac-ltp-encode: CMP_TARGET = 1535
fate-aac-ltp-encode: SIZE_TOLERANCE = 3560
fate-aac-ltp-encode: FUZZ = 10