ffmpeg | branch: master | Rostislav Pehlivanov <atomnu...@gmail.com> | Sat Oct 17 22:50:55 2015 +0100| [7303962f1467e302906561be53ca4d51abbe5522] | committer: Rostislav Pehlivanov
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. > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=7303962f1467e302906561be53ca4d51abbe5522 --- libavcodec/aacenc_ltp.c | 60 +++++++++++++++++++++++------------------------ tests/fate/aac.mak | 4 ++-- 2 files changed, 31 insertions(+), 33 deletions(-) diff --git a/libavcodec/aacenc_ltp.c b/libavcodec/aacenc_ltp.c index 4a5a8ce..d240460 100644 --- a/libavcodec/aacenc_ltp.c +++ b/libavcodec/aacenc_ltp.c @@ -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]) { diff --git a/tests/fate/aac.mak b/tests/fate/aac.mak index 331e66f..d578b4b 100644 --- a/tests/fate/aac.mak +++ b/tests/fate/aac.mak @@ -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 _______________________________________________ ffmpeg-cvslog mailing list ffmpeg-cvslog@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-cvslog
