ffmpeg | branch: master | Rostislav Pehlivanov <atomnu...@gmail.com> | Mon Sep 7 12:39:04 2015 +0100| [b6cc8ec7ecc3bea2a84fe65fa278e22fbb26a48c] | committer: Rostislav Pehlivanov
aaccoder: add frequency scaling and quantization correction for PNS This commit once again improves the PNS implementation by scaling the thresholds with frequency. The thresholds get looser as the frequency increases since higher frequencies are basically noise to human ears. Also, this introduces quantization error correction for PNS. Should the error be too much, no PNS will be used. The energy_ratio is used to regulate the actual encoded PNS energy: if the generated PNS energy is higher than the energy from the psy system, energy_ratio is used to correct it so that hopefully once requantized and transmitted the value in the decoder will be closer to what the encoder has. Signed-off-by: Rostislav Pehlivanov <atomnu...@gmail.com> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=b6cc8ec7ecc3bea2a84fe65fa278e22fbb26a48c --- libavcodec/aaccoder.c | 28 +++++++++++++++++----------- tests/fate/aac.mak | 2 +- 2 files changed, 18 insertions(+), 12 deletions(-) diff --git a/libavcodec/aaccoder.c b/libavcodec/aaccoder.c index 8a241d2..cf4b11b 100644 --- a/libavcodec/aaccoder.c +++ b/libavcodec/aaccoder.c @@ -876,36 +876,39 @@ static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChanne for (g = 0; g < sce->ics.num_swb; g++) { int noise_sfi, try_pns = 0; float dist1 = 0.0f, dist2 = 0.0f, noise_amp; - float energy = 0.0f, threshold = 0.0f, spread = 0.0f; + float pns_energy = 0.0f, energy_ratio, dist_thresh; + float sfb_energy = 0.0f, threshold = 0.0f, spread = 0.0f; + float freq_boost = FFMAX(0.88f*start*freq_mult/NOISE_LOW_LIMIT, 1.0f); if (start*freq_mult < NOISE_LOW_LIMIT) { start += sce->ics.swb_sizes[g]; continue; + } else { + dist_thresh = FFMIN(0.008f*(NOISE_LOW_LIMIT/start*freq_mult), 1.11f); } for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) { band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g]; - energy += band->energy; - spread += band->spread; - threshold += band->threshold; + sfb_energy += band->energy; + spread += band->spread; + threshold += band->threshold; } - sce->pns_ener[w*16+g] = energy; if (sce->zeroes[w*16+g]) { try_pns = 1; - } else if (energy < threshold) { + } else if (sfb_energy < threshold*freq_boost) { try_pns = 1; } else if (spread > spread_threshold) { try_pns = 0; - } else if (energy < threshold*thr_mult) { + } else if (sfb_energy < threshold*thr_mult*freq_boost) { try_pns = 1; } - if (!try_pns || !energy) { + if (!try_pns || !sfb_energy) { start += sce->ics.swb_sizes[g]; continue; } - noise_sfi = av_clip(roundf(log2f(energy)*2), -100, 155); /* Quantize */ - noise_amp = -ff_aac_pow2sf_tab[noise_sfi + POW_SF2_ZERO]; /* Dequantize */ + noise_sfi = av_clip(roundf(log2f(sfb_energy)*2), -100, 155); /* Quantize */ + noise_amp = -ff_aac_pow2sf_tab[noise_sfi + POW_SF2_ZERO]; /* Dequantize */ for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) { float band_energy, scale; band = &s->psy.ch[s->cur_channel+0].psy_bands[(w+w2)*16+g]; @@ -914,6 +917,7 @@ static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChanne band_energy = s->fdsp->scalarproduct_float(PNS, PNS, sce->ics.swb_sizes[g]); scale = noise_amp/sqrtf(band_energy); s->fdsp->vector_fmul_scalar(PNS, PNS, scale, sce->ics.swb_sizes[g]); + pns_energy += s->fdsp->scalarproduct_float(PNS, PNS, sce->ics.swb_sizes[g]); abs_pow34_v(NOR34, &sce->coeffs[start+(w+w2)*128], sce->ics.swb_sizes[g]); abs_pow34_v(PNS34, PNS, sce->ics.swb_sizes[g]); dist1 += quantize_band_cost(s, &sce->coeffs[start + (w+w2)*128], @@ -929,7 +933,9 @@ static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChanne NOISE_BT, lambda/band->threshold, INFINITY, NULL, 0); } - if (dist2 < dist1) { + energy_ratio = sfb_energy/pns_energy; /* Compensates for quantization error */ + sce->pns_ener[w*16+g] = energy_ratio*sfb_energy; + if (energy_ratio > 0.80f && energy_ratio < 1.20f && dist1/dist2 > dist_thresh) { sce->band_type[w*16+g] = NOISE_BT; sce->zeroes[w*16+g] = 0; if (sce->band_type[w*16+g-1] != NOISE_BT && /* Prevent holes */ diff --git a/tests/fate/aac.mak b/tests/fate/aac.mak index 910bac0..4391106 100644 --- a/tests/fate/aac.mak +++ b/tests/fate/aac.mak @@ -163,7 +163,7 @@ fate-aac-pns-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-re fate-aac-pns-encode: CMP = stddev fate-aac-pns-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav fate-aac-pns-encode: CMP_SHIFT = -4096 -fate-aac-pns-encode: CMP_TARGET = 647.11 +fate-aac-pns-encode: CMP_TARGET = 637.66 fate-aac-pns-encode: SIZE_TOLERANCE = 3560 fate-aac-pns-encode: FUZZ = 5 _______________________________________________ ffmpeg-cvslog mailing list ffmpeg-cvslog@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-cvslog
