| Message ID | 20160806235130.13284-1-atomnuker@gmail.com |
|---|---|
| State | Accepted |
| Commit | c92cc2d559dee8723e07754282906f2855f0c61f |
| Headers | show |
On Sun, Aug 07, 2016 at 12:53:47AM +0100, Rostislav Pehlivanov wrote: > On 7 August 2016 at 00:51, Rostislav Pehlivanov <atomnuker@gmail.com> wrote: > > > Does nothing fancy but still sounds very decent at 128kbps. > > Still room to improve by bringing in the low pass and PNS management > > from the main big twoloop which should improve its quality but not > > sacrifice that much speed. > > > > Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com> > > --- > > libavcodec/aaccoder.c | 154 ++++++++++++++++++++++++++++++ > > +++++++++++++------- > > 1 file changed, 134 insertions(+), 20 deletions(-) [...] > > > Forgot to mention speed is around 6-7 times faster than the regular twoloop > codec, which is around 3-4 times faster than libfaac. very niiice! Acked-by: Michael thanks alot! [...]
On Sun, Aug 07, 2016 at 00:51:29 +0100, Rostislav Pehlivanov wrote: > + //XXX: some heuristic to determine initial quantizers will reduce search time Is the "XXX" trying to tell someone something? > + for (g = 0; g < sce->ics.num_swb; g++) { I detect a spurious space in there. ;-) (This happens five times throughout your patch, BTW.) > [AAC_CODER_FAST] = { > search_for_quantizers_fast, > - encode_window_bands_info, > + codebook_trellis_rate, > quantize_and_encode_band, > ff_aac_encode_tns_info, > ff_aac_encode_ltp_info, Without understanding the code: Is this change related or even desired? Moritz
On 7 August 2016 at 14:38, Moritz Barsnick <barsnick@gmx.net> wrote: > On Sun, Aug 07, 2016 at 00:51:29 +0100, Rostislav Pehlivanov wrote: > > + //XXX: some heuristic to determine initial quantizers will reduce > search time > > Is the "XXX" trying to tell someone something? > > > + for (g = 0; g < sce->ics.num_swb; g++) { > > I detect a spurious space in there. ;-) (This happens five times > throughout your patch, BTW.) > > The patch is a relatively untouched version of the old twoloop coder, so anything that was there was left there. I'll remove the spaces when pushing. > > [AAC_CODER_FAST] = { > > search_for_quantizers_fast, > > - encode_window_bands_info, > > + codebook_trellis_rate, > > quantize_and_encode_band, > > ff_aac_encode_tns_info, > > ff_aac_encode_ltp_info, > > Without understanding the code: Is this change related or even desired? > > Related, desired and needed. Twoloop only supports codebook_trellis_rate. encode_window_bands_info() is very limited in terms of features, quality and an absolute pain to maintain, the second ANMR gets ported to support codebook_trellis_rate it will be removed. Also I just noticed the caching system for the RDO function is unneeded here since it gets called once per window+group. I'll remove that too when pusing.
On 7 August 2016 at 00:51, Rostislav Pehlivanov <atomnuker@gmail.com> wrote: > Does nothing fancy but still sounds very decent at 128kbps. > Still room to improve by bringing in the low pass and PNS management > from the main big twoloop which should improve its quality but not > sacrifice that much speed. > > Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com> > --- > libavcodec/aaccoder.c | 154 ++++++++++++++++++++++++++++++ > +++++++++++++------- > 1 file changed, 134 insertions(+), 20 deletions(-) > > diff --git a/libavcodec/aaccoder.c b/libavcodec/aaccoder.c > index bca1f59..edf29f4 100644 > --- a/libavcodec/aaccoder.c > +++ b/libavcodec/aaccoder.c > @@ -396,34 +396,148 @@ static void search_for_quantizers_fast(AVCodecContext > *avctx, AACEncContext *s, > SingleChannelElement *sce, > const float lambda) > { > - int i, w, w2, g; > - int minq = 255; > - > - memset(sce->sf_idx, 0, sizeof(sce->sf_idx)); > + int start = 0, i, w, w2, g; > + int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / > avctx->channels * (lambda / 120.f); > + float dists[128] = { 0 }, uplims[128] = { 0 }; > + float maxvals[128]; > + int fflag, minscaler; > + int its = 0; > + int allz = 0; > + float minthr = INFINITY; > + > + // for values above this the decoder might end up in an endless loop > + // due to always having more bits than what can be encoded. > + destbits = FFMIN(destbits, 5800); > + //XXX: some heuristic to determine initial quantizers will reduce > search time > + //determine zero bands and upper limits > for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { > - for (g = 0; g < sce->ics.num_swb; g++) { > + start = 0; > + for (g = 0; g < sce->ics.num_swb; g++) { > + int nz = 0; > + float uplim = 0.0f, energy = 0.0f; > for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) { > FFPsyBand *band = &s->psy.ch[s->cur_channel]. > psy_bands[(w+w2)*16+g]; > - if (band->energy <= band->threshold) { > - sce->sf_idx[(w+w2)*16+g] = 218; > + uplim += band->threshold; > + energy += band->energy; > + if (band->energy <= band->threshold || band->threshold == > 0.0f) { > sce->zeroes[(w+w2)*16+g] = 1; > - } else { > - sce->sf_idx[(w+w2)*16+g] = av_clip(SCALE_ONE_POS - > SCALE_DIV_512 + log2f(band->threshold), 80, 218); > - sce->zeroes[(w+w2)*16+g] = 0; > + continue; > } > - minq = FFMIN(minq, sce->sf_idx[(w+w2)*16+g]); > + nz = 1; > } > + uplims[w*16+g] = uplim *512; > + sce->band_type[w*16+g] = 0; > + sce->zeroes[w*16+g] = !nz; > + if (nz) > + minthr = FFMIN(minthr, uplim); > + allz |= nz; > + start += sce->ics.swb_sizes[g]; > } > } > - for (i = 0; i < 128; i++) { > - sce->sf_idx[i] = 140; > - //av_clip(sce->sf_idx[i], minq, minq + SCALE_MAX_DIFF - 1); > + for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { > + for (g = 0; g < sce->ics.num_swb; g++) { > + if (sce->zeroes[w*16+g]) { > + sce->sf_idx[w*16+g] = SCALE_ONE_POS; > + continue; > + } > + sce->sf_idx[w*16+g] = SCALE_ONE_POS + > FFMIN(log2f(uplims[w*16+g]/minthr)*4,59); > + } > } > - //set the same quantizers inside window groups > - for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) > - for (g = 0; g < sce->ics.num_swb; g++) > - for (w2 = 1; w2 < sce->ics.group_len[w]; w2++) > - sce->sf_idx[(w+w2)*16+g] = sce->sf_idx[w*16+g]; > + > + if (!allz) > + return; > + abs_pow34_v(s->scoefs, sce->coeffs, 1024); > + ff_quantize_band_cost_cache_init(s); > + > + for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { > + start = w*128; > + for (g = 0; g < sce->ics.num_swb; g++) { > + const float *scaled = s->scoefs + start; > + maxvals[w*16+g] = find_max_val(sce->ics.group_len[w], > sce->ics.swb_sizes[g], scaled); > + start += sce->ics.swb_sizes[g]; > + } > + } > + > + //perform two-loop search > + //outer loop - improve quality > + do { > + int tbits, qstep; > + minscaler = sce->sf_idx[0]; > + //inner loop - quantize spectrum to fit into given number of bits > + qstep = its ? 1 : 32; > + do { > + int prev = -1; > + tbits = 0; > + for (w = 0; w < sce->ics.num_windows; w += > sce->ics.group_len[w]) { > + start = w*128; > + for (g = 0; g < sce->ics.num_swb; g++) { > + const float *coefs = sce->coeffs + start; > + const float *scaled = s->scoefs + start; > + int bits = 0; > + int cb; > + float dist = 0.0f; > + > + if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= > 218) { > + start += sce->ics.swb_sizes[g]; > + continue; > + } > + minscaler = FFMIN(minscaler, sce->sf_idx[w*16+g]); > + cb = find_min_book(maxvals[w*16+g], > sce->sf_idx[w*16+g]); > + for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) { > + int b; > + dist += quantize_band_cost_cached(s, w + w2, g, > + coefs + w2*128, > + scaled + w2*128, > + > sce->ics.swb_sizes[g], > + > sce->sf_idx[w*16+g], > + cb, 1.0f, > INFINITY, > + &b, NULL, 0); > + bits += b; > + } > + dists[w*16+g] = dist - bits; > + if (prev != -1) { > + bits += ff_aac_scalefactor_bits[sce->sf_idx[w*16+g] > - prev + SCALE_DIFF_ZERO]; > + } > + tbits += bits; > + start += sce->ics.swb_sizes[g]; > + prev = sce->sf_idx[w*16+g]; > + } > + } > + if (tbits > destbits) { > + for (i = 0; i < 128; i++) > + if (sce->sf_idx[i] < 218 - qstep) > + sce->sf_idx[i] += qstep; > + } else { > + for (i = 0; i < 128; i++) > + if (sce->sf_idx[i] > 60 - qstep) > + sce->sf_idx[i] -= qstep; > + } > + qstep >>= 1; > + if (!qstep && tbits > destbits*1.02 && sce->sf_idx[0] < 217) > + qstep = 1; > + } while (qstep); > + > + fflag = 0; > + minscaler = av_clip(minscaler, 60, 255 - SCALE_MAX_DIFF); > + > + for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) > { > + for (g = 0; g < sce->ics.num_swb; g++) { > + int prevsc = sce->sf_idx[w*16+g]; > + if (dists[w*16+g] > uplims[w*16+g] && sce->sf_idx[w*16+g] > > 60) { > + if (find_min_book(maxvals[w*16+g], > sce->sf_idx[w*16+g]-1)) > + sce->sf_idx[w*16+g]--; > + else //Try to make sure there is some energy in every > band > + sce->sf_idx[w*16+g]-=2; > + } > + sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], > minscaler, minscaler + SCALE_MAX_DIFF); > + sce->sf_idx[w*16+g] = FFMIN(sce->sf_idx[w*16+g], 219); > + if (sce->sf_idx[w*16+g] != prevsc) > + fflag = 1; > + sce->band_type[w*16+g] = find_min_book(maxvals[w*16+g], > sce->sf_idx[w*16+g]); > + } > + } > + its++; > + } while (fflag && its < 10); > } > > static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, > SingleChannelElement *sce) > @@ -828,7 +942,7 @@ AACCoefficientsEncoder ff_aac_coders[AAC_CODER_NB] = { > }, > [AAC_CODER_FAST] = { > search_for_quantizers_fast, > - encode_window_bands_info, > + codebook_trellis_rate, > quantize_and_encode_band, > ff_aac_encode_tns_info, > ff_aac_encode_ltp_info, > -- > 2.8.1.369.geae769a > > Pushed, thanks for the reviews
diff --git a/libavcodec/aaccoder.c b/libavcodec/aaccoder.c index bca1f59..edf29f4 100644 --- a/libavcodec/aaccoder.c +++ b/libavcodec/aaccoder.c @@ -396,34 +396,148 @@ static void search_for_quantizers_fast(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce, const float lambda) { - int i, w, w2, g; - int minq = 255; - - memset(sce->sf_idx, 0, sizeof(sce->sf_idx)); + int start = 0, i, w, w2, g; + int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->channels * (lambda / 120.f); + float dists[128] = { 0 }, uplims[128] = { 0 }; + float maxvals[128]; + int fflag, minscaler; + int its = 0; + int allz = 0; + float minthr = INFINITY; + + // for values above this the decoder might end up in an endless loop + // due to always having more bits than what can be encoded. + destbits = FFMIN(destbits, 5800); + //XXX: some heuristic to determine initial quantizers will reduce search time + //determine zero bands and upper limits for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { - for (g = 0; g < sce->ics.num_swb; g++) { + start = 0; + for (g = 0; g < sce->ics.num_swb; g++) { + int nz = 0; + float uplim = 0.0f, energy = 0.0f; for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) { FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g]; - if (band->energy <= band->threshold) { - sce->sf_idx[(w+w2)*16+g] = 218; + uplim += band->threshold; + energy += band->energy; + if (band->energy <= band->threshold || band->threshold == 0.0f) { sce->zeroes[(w+w2)*16+g] = 1; - } else { - sce->sf_idx[(w+w2)*16+g] = av_clip(SCALE_ONE_POS - SCALE_DIV_512 + log2f(band->threshold), 80, 218); - sce->zeroes[(w+w2)*16+g] = 0; + continue; } - minq = FFMIN(minq, sce->sf_idx[(w+w2)*16+g]); + nz = 1; } + uplims[w*16+g] = uplim *512; + sce->band_type[w*16+g] = 0; + sce->zeroes[w*16+g] = !nz; + if (nz) + minthr = FFMIN(minthr, uplim); + allz |= nz; + start += sce->ics.swb_sizes[g]; } } - for (i = 0; i < 128; i++) { - sce->sf_idx[i] = 140; - //av_clip(sce->sf_idx[i], minq, minq + SCALE_MAX_DIFF - 1); + for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { + for (g = 0; g < sce->ics.num_swb; g++) { + if (sce->zeroes[w*16+g]) { + sce->sf_idx[w*16+g] = SCALE_ONE_POS; + continue; + } + sce->sf_idx[w*16+g] = SCALE_ONE_POS + FFMIN(log2f(uplims[w*16+g]/minthr)*4,59); + } } - //set the same quantizers inside window groups - for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) - for (g = 0; g < sce->ics.num_swb; g++) - for (w2 = 1; w2 < sce->ics.group_len[w]; w2++) - sce->sf_idx[(w+w2)*16+g] = sce->sf_idx[w*16+g]; + + if (!allz) + return; + abs_pow34_v(s->scoefs, sce->coeffs, 1024); + ff_quantize_band_cost_cache_init(s); + + for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { + start = w*128; + for (g = 0; g < sce->ics.num_swb; g++) { + const float *scaled = s->scoefs + start; + maxvals[w*16+g] = find_max_val(sce->ics.group_len[w], sce->ics.swb_sizes[g], scaled); + start += sce->ics.swb_sizes[g]; + } + } + + //perform two-loop search + //outer loop - improve quality + do { + int tbits, qstep; + minscaler = sce->sf_idx[0]; + //inner loop - quantize spectrum to fit into given number of bits + qstep = its ? 1 : 32; + do { + int prev = -1; + tbits = 0; + for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { + start = w*128; + for (g = 0; g < sce->ics.num_swb; g++) { + const float *coefs = sce->coeffs + start; + const float *scaled = s->scoefs + start; + int bits = 0; + int cb; + float dist = 0.0f; + + if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218) { + start += sce->ics.swb_sizes[g]; + continue; + } + minscaler = FFMIN(minscaler, sce->sf_idx[w*16+g]); + cb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]); + for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) { + int b; + dist += quantize_band_cost_cached(s, w + w2, g, + coefs + w2*128, + scaled + w2*128, + sce->ics.swb_sizes[g], + sce->sf_idx[w*16+g], + cb, 1.0f, INFINITY, + &b, NULL, 0); + bits += b; + } + dists[w*16+g] = dist - bits; + if (prev != -1) { + bits += ff_aac_scalefactor_bits[sce->sf_idx[w*16+g] - prev + SCALE_DIFF_ZERO]; + } + tbits += bits; + start += sce->ics.swb_sizes[g]; + prev = sce->sf_idx[w*16+g]; + } + } + if (tbits > destbits) { + for (i = 0; i < 128; i++) + if (sce->sf_idx[i] < 218 - qstep) + sce->sf_idx[i] += qstep; + } else { + for (i = 0; i < 128; i++) + if (sce->sf_idx[i] > 60 - qstep) + sce->sf_idx[i] -= qstep; + } + qstep >>= 1; + if (!qstep && tbits > destbits*1.02 && sce->sf_idx[0] < 217) + qstep = 1; + } while (qstep); + + fflag = 0; + minscaler = av_clip(minscaler, 60, 255 - SCALE_MAX_DIFF); + + for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { + for (g = 0; g < sce->ics.num_swb; g++) { + int prevsc = sce->sf_idx[w*16+g]; + if (dists[w*16+g] > uplims[w*16+g] && sce->sf_idx[w*16+g] > 60) { + if (find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]-1)) + sce->sf_idx[w*16+g]--; + else //Try to make sure there is some energy in every band + sce->sf_idx[w*16+g]-=2; + } + sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler, minscaler + SCALE_MAX_DIFF); + sce->sf_idx[w*16+g] = FFMIN(sce->sf_idx[w*16+g], 219); + if (sce->sf_idx[w*16+g] != prevsc) + fflag = 1; + sce->band_type[w*16+g] = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]); + } + } + its++; + } while (fflag && its < 10); } static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce) @@ -828,7 +942,7 @@ AACCoefficientsEncoder ff_aac_coders[AAC_CODER_NB] = { }, [AAC_CODER_FAST] = { search_for_quantizers_fast, - encode_window_bands_info, + codebook_trellis_rate, quantize_and_encode_band, ff_aac_encode_tns_info, ff_aac_encode_ltp_info,
Does nothing fancy but still sounds very decent at 128kbps. Still room to improve by bringing in the low pass and PNS management from the main big twoloop which should improve its quality but not sacrifice that much speed. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com> --- libavcodec/aaccoder.c | 154 +++++++++++++++++++++++++++++++++++++++++++------- 1 file changed, 134 insertions(+), 20 deletions(-)