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author | Claudio Freire <klaussfreire@gmail.com> | 2015-12-01 03:28:36 -0300 |
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committer | Claudio Freire <klaussfreire@gmail.com> | 2015-12-02 07:47:37 -0300 |
commit | ca203e9985cd2dcf42a0c0853940850d3a8edf3a (patch) | |
tree | 4dd1ad824283d75afdd1191c70be982c03c1b683 /libavcodec/aaccoder.c | |
parent | ec83efd4d3c5fe1e4bc5723d0b91abf85b722f41 (diff) | |
download | ffmpeg-ca203e9985cd2dcf42a0c0853940850d3a8edf3a.tar.gz |
AAC encoder: improve SF range utilization
This patch does 4 things, all of which interact and thus it
woudln't be possible to commit them separately without causing
either quality regressions or assertion failures.
Fate comparison targets don't all reflect improvements in
quality, yet listening tests show substantially improved quality
and stability.
1. Increase SF range utilization.
The spec requires SF delta values to be constrained within the
range -60..60. The previous code was applying that range to
the whole SF array and not only the deltas of consecutive values,
because doing so requires smarter code: zeroing or otherwise
skipping a band may invalidate lots of SF choices.
This patch implements that logic to allow the coders to utilize
the full dynamic range of scalefactors, increasing quality quite
considerably, and fixing delta-SF-related assertion failures,
since now the limitation is enforced rather than asserted.
2. PNS tweaks
The previous modification makes big improvements in twoloop's
efficiency, and every time that happens PNS logic needs to be
tweaked accordingly to avoid it from stepping all over twoloop's
decisions. This patch includes modifications of the sort.
3. Account for lowpass cutoff during PSY analysis
The closer PSY's allocation is to final allocation the better
the quality is, and given these modifications, twoloop is now
very efficient at avoiding holes. Thus, to compute accurate
thresholds, PSY needs to account for the lowpass applied
implicitly during twoloop (by zeroing high bands).
This patch makes twoloop set the cutoff in psymodel's context
the first time it runs, and makes PSY account for it during
threshold computation, making PE and threshold computations
closer to the final allocation and thus achieving better
subjective quality.
4. Tweaks to RC lambda tracking loop in relation to PNS
Without this tweak some corner cases cause quality regressions.
Basically, lambda needs to react faster to overall bitrate
efficiency changes since now PNS can be quite successful in
enforcing maximum bitrates, when PSY allocates too many bits
to the lower bands, suppressing the signals RC logic uses to
lower lambda in those cases and causing aggressive PNS.
This tweak makes PNS much less aggressive, though it can still
use some further tweaks.
Also update MIPS specializations and adjust fuzz
Also in lavc/mips/aacpsy_mips.h: remove trailing whitespace
Diffstat (limited to 'libavcodec/aaccoder.c')
-rw-r--r-- | libavcodec/aaccoder.c | 60 |
1 files changed, 41 insertions, 19 deletions
diff --git a/libavcodec/aaccoder.c b/libavcodec/aaccoder.c index 2a66045c8e..2a0cb1f6b5 100644 --- a/libavcodec/aaccoder.c +++ b/libavcodec/aaccoder.c @@ -54,7 +54,7 @@ /* Parameter of f(x) = a*(lambda/100), defines the maximum fourier spread * beyond which no PNS is used (since the SFBs contain tone rather than noise) */ -#define NOISE_SPREAD_THRESHOLD 0.5073f +#define NOISE_SPREAD_THRESHOLD 0.9f /* Parameter of f(x) = a*(100/lambda), defines how much PNS is allowed to * replace low energy non zero bands */ @@ -591,6 +591,7 @@ static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChanne int bandwidth, cutoff; float *PNS = &s->scoefs[0*128], *PNS34 = &s->scoefs[1*128]; float *NOR34 = &s->scoefs[3*128]; + uint8_t nextband[128]; const float lambda = s->lambda; const float freq_mult = avctx->sample_rate*0.5f/wlen; const float thr_mult = NOISE_LAMBDA_REPLACE*(100.0f/lambda); @@ -604,6 +605,7 @@ static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChanne /** Keep this in sync with twoloop's cutoff selection */ float rate_bandwidth_multiplier = 1.5f; + int prev = -1000, prev_sf = -1; int frame_bit_rate = (avctx->flags & CODEC_FLAG_QSCALE) ? (refbits * rate_bandwidth_multiplier * avctx->sample_rate / 1024) : (avctx->bit_rate / avctx->channels); @@ -619,6 +621,7 @@ static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChanne cutoff = bandwidth * 2 * wlen / avctx->sample_rate; memcpy(sce->band_alt, sce->band_type, sizeof(sce->band_type)); + ff_init_nextband_map(sce, nextband); for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { int wstart = w*128; for (g = 0; g < sce->ics.num_swb; g++) { @@ -655,16 +658,27 @@ static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChanne * * At this stage, point 2 is relaxed for zeroed bands near the noise threshold (hole avoidance is more important) */ - if (((sce->zeroes[w*16+g] || !sce->band_alt[w*16+g]) && sfb_energy < threshold*sqrtf(1.5f/freq_boost)) || spread < spread_threshold || + if ((!sce->zeroes[w*16+g] && !ff_sfdelta_can_remove_band(sce, nextband, prev_sf, w*16+g)) || + ((sce->zeroes[w*16+g] || !sce->band_alt[w*16+g]) && sfb_energy < threshold*sqrtf(1.0f/freq_boost)) || spread < spread_threshold || (!sce->zeroes[w*16+g] && sce->band_alt[w*16+g] && sfb_energy > threshold*thr_mult*freq_boost) || min_energy < pns_transient_energy_r * max_energy ) { sce->pns_ener[w*16+g] = sfb_energy; + if (!sce->zeroes[w*16+g]) + prev_sf = sce->sf_idx[w*16+g]; continue; } pns_tgt_energy = sfb_energy*FFMIN(1.0f, spread*spread); noise_sfi = av_clip(roundf(log2f(pns_tgt_energy)*2), -100, 155); /* Quantize */ noise_amp = -ff_aac_pow2sf_tab[noise_sfi + POW_SF2_ZERO]; /* Dequantize */ + if (prev != -1000) { + int noise_sfdiff = noise_sfi - prev + SCALE_DIFF_ZERO; + if (noise_sfdiff < 0 || noise_sfdiff > 2*SCALE_MAX_DIFF) { + if (!sce->zeroes[w*16+g]) + prev_sf = sce->sf_idx[w*16+g]; + continue; + } + } for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) { float band_energy, scale, pns_senergy; const int start_c = (w+w2)*128+sce->ics.swb_offset[g]; @@ -697,7 +711,10 @@ static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChanne if (sce->zeroes[w*16+g] || !sce->band_alt[w*16+g] || (energy_ratio > 0.85f && energy_ratio < 1.25f && dist2 < dist1)) { sce->band_type[w*16+g] = NOISE_BT; sce->zeroes[w*16+g] = 0; + prev = noise_sfi; } + if (!sce->zeroes[w*16+g]) + prev_sf = sce->sf_idx[w*16+g]; } } } @@ -775,7 +792,8 @@ static void mark_pns(AACEncContext *s, AVCodecContext *avctx, SingleChannelEleme static void search_for_ms(AACEncContext *s, ChannelElement *cpe) { - int start = 0, i, w, w2, g, sid_sf_boost; + int start = 0, i, w, w2, g, sid_sf_boost, prev_mid, prev_side; + uint8_t nextband0[128], nextband1[128]; float M[128], S[128]; float *L34 = s->scoefs, *R34 = s->scoefs + 128, *M34 = s->scoefs + 128*2, *S34 = s->scoefs + 128*3; const float lambda = s->lambda; @@ -784,21 +802,19 @@ static void search_for_ms(AACEncContext *s, ChannelElement *cpe) SingleChannelElement *sce1 = &cpe->ch[1]; if (!cpe->common_window) return; - for (w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]) { - int min_sf_idx_mid = SCALE_MAX_POS; - int min_sf_idx_side = SCALE_MAX_POS; - for (g = 0; g < sce0->ics.num_swb; g++) { - if (!sce0->zeroes[w*16+g] && sce0->band_type[w*16+g] < RESERVED_BT) - min_sf_idx_mid = FFMIN(min_sf_idx_mid, sce0->sf_idx[w*16+g]); - if (!sce1->zeroes[w*16+g] && sce1->band_type[w*16+g] < RESERVED_BT) - min_sf_idx_side = FFMIN(min_sf_idx_side, sce1->sf_idx[w*16+g]); - } + /** Scout out next nonzero bands */ + ff_init_nextband_map(sce0, nextband0); + ff_init_nextband_map(sce1, nextband1); + + prev_mid = sce0->sf_idx[0]; + prev_side = sce1->sf_idx[0]; + for (w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]) { start = 0; for (g = 0; g < sce0->ics.num_swb; g++) { float bmax = bval2bmax(g * 17.0f / sce0->ics.num_swb) / 0.0045f; cpe->ms_mask[w*16+g] = 0; - if (!cpe->ch[0].zeroes[w*16+g] && !cpe->ch[1].zeroes[w*16+g]) { + if (!sce0->zeroes[w*16+g] && !sce1->zeroes[w*16+g]) { float Mmax = 0.0f, Smax = 0.0f; /* Must compute mid/side SF and book for the whole window group */ @@ -825,16 +841,18 @@ static void search_for_ms(AACEncContext *s, ChannelElement *cpe) int midcb, sidcb; minidx = FFMIN(sce0->sf_idx[w*16+g], sce1->sf_idx[w*16+g]); - mididx = av_clip(minidx, min_sf_idx_mid, min_sf_idx_mid + SCALE_MAX_DIFF); - sididx = av_clip(minidx - sid_sf_boost * 3, min_sf_idx_side, min_sf_idx_side + SCALE_MAX_DIFF); - midcb = find_min_book(Mmax, mididx); - sidcb = find_min_book(Smax, sididx); - - if ((mididx > minidx) || (sididx > minidx)) { + mididx = av_clip(minidx, 0, SCALE_MAX_POS - SCALE_DIV_512); + sididx = av_clip(minidx - sid_sf_boost * 3, 0, SCALE_MAX_POS - SCALE_DIV_512); + if (!cpe->is_mask[w*16+g] && sce0->band_type[w*16+g] != NOISE_BT && sce1->band_type[w*16+g] != NOISE_BT + && ( !ff_sfdelta_can_replace(sce0, nextband0, prev_mid, mididx, w*16+g) + || !ff_sfdelta_can_replace(sce1, nextband1, prev_side, sididx, w*16+g))) { /* scalefactor range violation, bad stuff, will decrease quality unacceptably */ continue; } + midcb = find_min_book(Mmax, mididx); + sidcb = find_min_book(Smax, sididx); + /* No CB can be zero */ midcb = FFMAX(1,midcb); sidcb = FFMAX(1,sidcb); @@ -900,6 +918,10 @@ static void search_for_ms(AACEncContext *s, ChannelElement *cpe) } } } + if (!sce0->zeroes[w*16+g] && sce0->band_type[w*16+g] < RESERVED_BT) + prev_mid = sce0->sf_idx[w*16+g]; + if (!sce1->zeroes[w*16+g] && !cpe->is_mask[w*16+g] && sce1->band_type[w*16+g] < RESERVED_BT) + prev_side = sce1->sf_idx[w*16+g]; start += sce0->ics.swb_sizes[g]; } } |