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aacenc_ltp: adjust and speed up autocorrelation calculations

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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