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FATE_AAC += fate-aac-al04_44
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fate-aac-al04_44: CMD = pcm -i $(TARGET_SAMPLES)/aac/al04_44.mp4
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fate-aac-al04_44: REF = $(SAMPLES)/aac/al04_44.s16
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FATE_AAC += fate-aac-al04sf_48
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fate-aac-al04sf_48: CMD = pcm -i $(TARGET_SAMPLES)/aac/al04sf_48.mp4
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fate-aac-al04sf_48: REF = $(SAMPLES)/aac/al04sf_48.s16
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FATE_AAC += fate-aac-al05_44
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fate-aac-al05_44: CMD = pcm -i $(TARGET_SAMPLES)/aac/al05_44.mp4
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fate-aac-al05_44: REF = $(SAMPLES)/aac/al05_44.s16
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FATE_AAC += fate-aac-al06_44
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fate-aac-al06_44: CMD = pcm -i $(TARGET_SAMPLES)/aac/al06_44.mp4
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fate-aac-al06_44: REF = $(SAMPLES)/aac/al06_44_reorder.s16
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FATE_AAC += fate-aac-al07_96
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fate-aac-al07_96: CMD = pcm -i $(TARGET_SAMPLES)/aac/al07_96.mp4
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fate-aac-al07_96: REF = $(SAMPLES)/aac/al07_96_reorder.s16
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FATE_AAC += fate-aac-al15_44
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fate-aac-al15_44: CMD = pcm -i $(TARGET_SAMPLES)/aac/al15_44.mp4
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fate-aac-al15_44: REF = $(SAMPLES)/aac/al15_44_reorder.s16
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FATE_AAC += fate-aac-al17_44
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fate-aac-al17_44: CMD = pcm -i $(TARGET_SAMPLES)/aac/al17_44.mp4
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fate-aac-al17_44: REF = $(SAMPLES)/aac/al17_44.s16
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FATE_AAC += fate-aac-al18_44
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fate-aac-al18_44: CMD = pcm -i $(TARGET_SAMPLES)/aac/al18_44.mp4
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fate-aac-al18_44: REF = $(SAMPLES)/aac/al18_44.s16
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FATE_AAC += fate-aac-am00_88
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fate-aac-am00_88: CMD = pcm -i $(TARGET_SAMPLES)/aac/am00_88.mp4
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fate-aac-am00_88: REF = $(SAMPLES)/aac/am00_88.s16
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FATE_AAC += fate-aac-am05_44
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fate-aac-am05_44: CMD = pcm -i $(TARGET_SAMPLES)/aac/am05_44.mp4
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fate-aac-am05_44: REF = $(SAMPLES)/aac/am05_44_reorder.s16
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FATE_AAC += fate-aac-al_sbr_hq_cm_48_2
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fate-aac-al_sbr_hq_cm_48_2: CMD = pcm -i $(TARGET_SAMPLES)/aac/al_sbr_cm_48_2.mp4
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fate-aac-al_sbr_hq_cm_48_2: REF = $(SAMPLES)/aac/al_sbr_hq_cm_48_2.s16
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FATE_AAC += fate-aac-al_sbr_hq_cm_48_5.1
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fate-aac-al_sbr_hq_cm_48_5.1: CMD = pcm -i $(TARGET_SAMPLES)/aac/al_sbr_cm_48_5.1.mp4
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fate-aac-al_sbr_hq_cm_48_5.1: REF = $(SAMPLES)/aac/al_sbr_hq_cm_48_5.1_reorder.s16
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FATE_AAC += fate-aac-al_sbr_hq_sr_48_2_fsaac48
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fate-aac-al_sbr_hq_sr_48_2_fsaac48: CMD = pcm -i $(TARGET_SAMPLES)/aac/al_sbr_sr_48_2_fsaac48.mp4
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fate-aac-al_sbr_hq_sr_48_2_fsaac48: REF = $(SAMPLES)/aac/al_sbr_hq_sr_48_2_fsaac48.s16
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FATE_AAC += fate-aac-al_sbr_ps_04_ur
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fate-aac-al_sbr_ps_04_ur: CMD = pcm -i $(TARGET_SAMPLES)/aac/al_sbr_ps_04_new.mp4
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fate-aac-al_sbr_ps_04_ur: REF = $(SAMPLES)/aac/al_sbr_ps_04_ur.s16
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FATE_AAC += fate-aac-al_sbr_ps_06_ur
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fate-aac-al_sbr_ps_06_ur: CMD = pcm -i $(TARGET_SAMPLES)/aac/al_sbr_ps_06_new.mp4
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fate-aac-al_sbr_ps_06_ur: REF = $(SAMPLES)/aac/al_sbr_ps_06_ur.s16
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FATE_AAC += fate-aac-ap05_48
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fate-aac-ap05_48: CMD = pcm -i $(TARGET_SAMPLES)/aac/ap05_48.mp4
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fate-aac-ap05_48: REF = $(SAMPLES)/aac/ap05_48.s16
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FATE_AAC += fate-aac-er_ad6000np_44_ep0
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fate-aac-er_ad6000np_44_ep0: CMD = pcm -i $(TARGET_SAMPLES)/aac/er_ad6000np_44_ep0.mp4
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fate-aac-er_ad6000np_44_ep0: REF = $(SAMPLES)/aac/er_ad6000np_44.s16
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FATE_AAC += fate-aac-er_eld1001np_44_ep0
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fate-aac-er_eld1001np_44_ep0: CMD = pcm -i $(TARGET_SAMPLES)/aac/er_eld1001np_44_ep0.mp4
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fate-aac-er_eld1001np_44_ep0: REF = $(SAMPLES)/aac/er_eld1001np_44.s16
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FATE_AAC += fate-aac-er_eld2000np_48_ep0
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fate-aac-er_eld2000np_48_ep0: CMD = pcm -i $(TARGET_SAMPLES)/aac/er_eld2000np_48_ep0.mp4
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fate-aac-er_eld2000np_48_ep0: REF = $(SAMPLES)/aac/er_eld2000np_48_ep0.s16
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FATE_AAC += fate-aac-er_eld2100np_48_ep0
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fate-aac-er_eld2100np_48_ep0: CMD = pcm -i $(TARGET_SAMPLES)/aac/er_eld2100np_48_ep0.mp4
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fate-aac-er_eld2100np_48_ep0: REF = $(SAMPLES)/aac/er_eld2100np_48.s16
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FATE_AAC_FIXED += fate-aac-fixed-al04_44
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fate-aac-fixed-al04_44: CMD = pcm -c aac_fixed -i $(TARGET_SAMPLES)/aac/al04_44.mp4
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fate-aac-fixed-al04_44: REF = $(SAMPLES)/aac/al04_44.s16
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FATE_AAC_FIXED += fate-aac-fixed-al05_44
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fate-aac-fixed-al05_44: CMD = pcm -c aac_fixed -i $(TARGET_SAMPLES)/aac/al05_44.mp4
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fate-aac-fixed-al05_44: REF = $(SAMPLES)/aac/al05_44.s16
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FATE_AAC_FIXED += fate-aac-fixed-al06_44
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fate-aac-fixed-al06_44: CMD = pcm -c aac_fixed -i $(TARGET_SAMPLES)/aac/al06_44.mp4
|
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fate-aac-fixed-al06_44: REF = $(SAMPLES)/aac/al06_44_reorder.s16
|
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FATE_AAC_FIXED += fate-aac-fixed-al15_44
|
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fate-aac-fixed-al15_44: CMD = pcm -c aac_fixed -i $(TARGET_SAMPLES)/aac/al15_44.mp4
|
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fate-aac-fixed-al15_44: REF = $(SAMPLES)/aac/al15_44_reorder.s16
|
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|
FATE_AAC_FIXED += fate-aac-fixed-al17_44
|
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fate-aac-fixed-al17_44: CMD = pcm -c aac_fixed -i $(TARGET_SAMPLES)/aac/al17_44.mp4
|
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|
fate-aac-fixed-al17_44: REF = $(SAMPLES)/aac/al17_44.s16
|
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|
FATE_AAC_FIXED += fate-aac-fixed-al18_44
|
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|
fate-aac-fixed-al18_44: CMD = pcm -c aac_fixed -i $(TARGET_SAMPLES)/aac/al18_44.mp4
|
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|
fate-aac-fixed-al18_44: REF = $(SAMPLES)/aac/al18_44.s16
|
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|
FATE_AAC_FIXED += fate-aac-fixed-al_sbr_hq_cm_48_2
|
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|
fate-aac-fixed-al_sbr_hq_cm_48_2: CMD = pcm -c aac_fixed -i $(TARGET_SAMPLES)/aac/al_sbr_cm_48_2.mp4
|
|
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|
fate-aac-fixed-al_sbr_hq_cm_48_2: REF = $(SAMPLES)/aac/al_sbr_hq_cm_48_2.s16
|
|
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|
|
FATE_AAC_FIXED += fate-aac-fixed-al_sbr_hq_cm_48_5.1
|
|
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|
|
fate-aac-fixed-al_sbr_hq_cm_48_5.1: CMD = pcm -c aac_fixed -i $(TARGET_SAMPLES)/aac/al_sbr_cm_48_5.1.mp4
|
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|
|
fate-aac-fixed-al_sbr_hq_cm_48_5.1: REF = $(SAMPLES)/aac/al_sbr_hq_cm_48_5.1_reorder.s16
|
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|
|
FATE_AAC_FIXED += fate-aac-fixed-al_sbr_hq_sr_48_2_fsaac48
|
|
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|
|
fate-aac-fixed-al_sbr_hq_sr_48_2_fsaac48: CMD = pcm -c aac_fixed -i $(TARGET_SAMPLES)/aac/al_sbr_sr_48_2_fsaac48.mp4
|
|
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|
|
fate-aac-fixed-al_sbr_hq_sr_48_2_fsaac48: REF = $(SAMPLES)/aac/al_sbr_hq_sr_48_2_fsaac48.s16
|
|
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|
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|
|
#FATE_AAC_FIXED += fate-aac-fixed-al_sbr_ps_06_ur
|
|
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|
#fate-aac-fixed-al_sbr_ps_06_ur: CMD = pcm -c aac_fixed-i $(TARGET_SAMPLES)/aac/al_sbr_ps_06_new.mp4
|
|
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|
|
#fate-aac-fixed-al_sbr_ps_06_ur: REF = $(SAMPLES)/aac/al_sbr_ps_06_ur.s16
|
|
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|
|
FATE_AAC_FIXED += fate-aac-fixed-ap05_48
|
|
|
|
|
fate-aac-fixed-ap05_48: CMD = pcm -c aac_fixed -i $(TARGET_SAMPLES)/aac/ap05_48.mp4
|
|
|
|
|
fate-aac-fixed-ap05_48: REF = $(SAMPLES)/aac/ap05_48.s16
|
|
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|
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|
|
FATE_AAC_FIXED += fate-aac-fixed-er_ad6000np_44_ep0
|
|
|
|
|
fate-aac-fixed-er_ad6000np_44_ep0: CMD = pcm -c aac_fixed -i $(TARGET_SAMPLES)/aac/er_ad6000np_44_ep0.mp4
|
|
|
|
|
fate-aac-fixed-er_ad6000np_44_ep0: REF = $(SAMPLES)/aac/er_ad6000np_44.s16
|
|
|
|
|
|
|
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|
|
FATE_AAC_FIXED += fate-aac-fixed-er_eld1001np_44_ep0
|
|
|
|
|
fate-aac-fixed-er_eld1001np_44_ep0: CMD = pcm -c aac_fixed -i $(TARGET_SAMPLES)/aac/er_eld1001np_44_ep0.mp4
|
|
|
|
|
fate-aac-fixed-er_eld1001np_44_ep0: REF = $(SAMPLES)/aac/er_eld1001np_44.s16
|
|
|
|
|
|
|
|
|
|
FATE_AAC_FIXED += fate-aac-fixed-er_eld2000np_48_ep0
|
|
|
|
|
fate-aac-fixed-er_eld2000np_48_ep0: CMD = pcm -c aac_fixed -i $(TARGET_SAMPLES)/aac/er_eld2000np_48_ep0.mp4
|
|
|
|
|
fate-aac-fixed-er_eld2000np_48_ep0: REF = $(SAMPLES)/aac/er_eld2000np_48_ep0.s16
|
|
|
|
|
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|
|
fate-aac-ct%: CMD = pcm -i $(TARGET_SAMPLES)/aac/CT_DecoderCheck/$(@:fate-aac-ct-%=%)
|
|
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|
|
fate-aac-ct%: REF = $(SAMPLES)/aac/CT_DecoderCheck/aacPlusv2.wav
|
|
|
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|
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|
|
FATE_AAC_CT_RAW = fate-aac-ct-sbr_i-ps_i.aac
|
|
|
|
|
|
|
|
|
|
FATE_AAC_CT = sbr_bc-ps_i.3gp \
|
|
|
|
|
sbr_bic-ps_i.3gp \
|
|
|
|
|
sbr_bc-ps_bc.mp4 \
|
|
|
|
|
sbr_bc-ps_i.mp4 \
|
|
|
|
|
sbr_i-ps_bic.mp4 \
|
|
|
|
|
sbr_i-ps_i.mp4
|
|
|
|
|
|
|
|
|
|
FATE_AAC += $(FATE_AAC_CT:%=fate-aac-ct-%)
|
|
|
|
|
|
|
|
|
|
FATE_AAC_ENCODE += fate-aac-aref-encode
|
|
|
|
|
fate-aac-aref-encode: ./tests/data/asynth-44100-2.wav
|
|
|
|
|
fate-aac-aref-encode: CMD = enc_dec_pcm adts wav s16le $(REF) -c:a aac -aac_is 0 -aac_pns 0 -aac_ms 0 -aac_tns 0 -b:a 512k
|
|
|
|
|
fate-aac-aref-encode: CMP = stddev
|
|
|
|
|
fate-aac-aref-encode: REF = ./tests/data/asynth-44100-2.wav
|
|
|
|
|
fate-aac-aref-encode: CMP_SHIFT = -4096
|
|
|
|
|
fate-aac-aref-encode: CMP_TARGET = 596
|
|
|
|
|
fate-aac-aref-encode: SIZE_TOLERANCE = 2464
|
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
9 years ago
|
|
|
fate-aac-aref-encode: FUZZ = 89
|
|
|
|
|
|
|
|
|
|
FATE_AAC_ENCODE += fate-aac-ln-encode
|
|
|
|
|
fate-aac-ln-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -c:a aac -aac_is 0 -aac_pns 0 -aac_ms 0 -aac_tns 0 -b:a 512k
|
|
|
|
|
fate-aac-ln-encode: CMP = stddev
|
|
|
|
|
fate-aac-ln-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
|
|
|
|
|
fate-aac-ln-encode: CMP_SHIFT = -4096
|
|
|
|
|
fate-aac-ln-encode: CMP_TARGET = 72
|
|
|
|
|
fate-aac-ln-encode: SIZE_TOLERANCE = 3560
|
AAC encoder: Extensive improvements
This finalizes merging of the work in the patches in ticket #2686.
Improvements to twoloop and RC logic are extensive.
The non-exhaustive list of twoloop improvments includes:
- Tweaks to distortion limits on the RD optimization phase of twoloop
- Deeper search in twoloop
- PNS information marking to let twoloop decide when to use it
(turned out having the decision made separately wasn't working)
- Tonal band detection and priorization
- Better band energy conservation rules
- Strict hole avoidance
For rate control:
- Use psymodel's bit allocation to allow proper use of the bit
reservoir. Don't work against the bit reservoir by moving lambda
in the opposite direction when psymodel decides to allocate more/less
bits to a frame.
- Retry the encode if the effective rate lies outside a reasonable
margin of psymodel's allocation or the selected ABR.
- Log average lambda at the end. Useful info for everyone, but especially
for tuning of the various encoder constants that relate to lambda
feedback.
Psy:
- Do not apply lowpass with a FIR filter, instead just let the coder
zero bands above the cutoff. The FIR filter induces group delay,
and while zeroing bands causes ripple, it's lost in the quantization
noise.
- Experimental VBR bit allocation code
- Tweak automatic lowpass filter threshold to maximize audio bandwidth
at all bitrates while still providing acceptable, stable quality.
I/S:
- Phase decision fixes. Unrelated to #2686, but the bugs only surfaced
when the merge was finalized. Measure I/S band energy accounting for
phase, and prevent I/S and M/S from being applied both.
PNS:
- Avoid marking short bands with PNS when they're part of a window
group in which there's a large variation of energy from one window
to the next. PNS can't preserve those and the effect is extremely
noticeable.
M/S:
- Implement BMLD protection similar to the specified in
ISO-IEC/13818:7-2003, Appendix C Section 6.1. Since M/S decision
doesn't conform to section 6.1, a different method had to be
implemented, but should provide equivalent protection.
- Move the decision logic closer to the method specified in
ISO-IEC/13818:7-2003, Appendix C Section 6.1. Specifically,
make sure M/S needs less bits than dual stereo.
- Don't apply M/S in bands that are using I/S
Now, this of course needed adjustments in the compare targets and
fuzz factors of the AAC encoder's fate tests, but if wondering why
the targets go up (more distortion), consider the previous coder
was using too many bits on LF content (far more than required by
psy), and thus those signals will now be more distorted, not less.
The extra distortion isn't audible though, I carried extensive
ABX testing to make sure.
A very similar patch was also extensively tested by Kamendo2 in
the context of #2686.
9 years ago
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fate-aac-ln-encode: FUZZ = 30
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FATE_AAC_ENCODE += fate-aac-ln-encode-128k
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fate-aac-ln-encode-128k: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -c:a aac -aac_is 0 -aac_pns 0 -aac_ms 0 -aac_tns 0 -b:a 128k -cutoff 22050
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fate-aac-ln-encode-128k: CMP = stddev
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fate-aac-ln-encode-128k: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
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fate-aac-ln-encode-128k: CMP_SHIFT = -4096
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fate-aac-ln-encode-128k: CMP_TARGET = 622
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fate-aac-ln-encode-128k: SIZE_TOLERANCE = 3560
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fate-aac-ln-encode-128k: FUZZ = 5
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FATE_AAC_ENCODE += fate-aac-pns-encode
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fate-aac-pns-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -c:a aac -aac_pns 1 -aac_is 0 -aac_ms 0 -aac_tns 0 -b:a 128k -cutoff 22050 -fflags +bitexact -flags +bitexact
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fate-aac-pns-encode: CMP = stddev
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fate-aac-pns-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
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fate-aac-pns-encode: CMP_SHIFT = -4096
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fate-aac-pns-encode: CMP_TARGET = 655
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fate-aac-pns-encode: SIZE_TOLERANCE = 3560
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fate-aac-pns-encode: FUZZ = 74
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FATE_AAC_ENCODE += fate-aac-tns-encode
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fate-aac-tns-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -c:a aac -aac_tns 1 -aac_is 0 -aac_pns 0 -aac_ms 0 -b:a 128k -cutoff 22050 -fflags +bitexact -flags +bitexact
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fate-aac-tns-encode: CMP = stddev
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fate-aac-tns-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
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fate-aac-tns-encode: CMP_SHIFT = -4096
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fate-aac-tns-encode: CMP_TARGET = 637
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fate-aac-tns-encode: FUZZ = 7
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fate-aac-tns-encode: SIZE_TOLERANCE = 3560
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FATE_AAC_ENCODE += fate-aac-is-encode
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fate-aac-is-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -c:a aac -aac_pns 0 -aac_is 1 -aac_ms 0 -b:a 128k -aac_tns 0 -cutoff 22050
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fate-aac-is-encode: CMP = stddev
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fate-aac-is-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
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fate-aac-is-encode: CMP_SHIFT = -4096
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fate-aac-is-encode: CMP_TARGET = 514
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fate-aac-is-encode: SIZE_TOLERANCE = 3560
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fate-aac-is-encode: FUZZ = 10
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FATE_AAC_ENCODE += fate-aac-ms-encode
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fate-aac-ms-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -c:a aac -aac_pns 0 -aac_is 0 -aac_ms 1 -aac_tns 0 -b:a 128k -cutoff 22050
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fate-aac-ms-encode: CMP = stddev
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fate-aac-ms-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
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fate-aac-ms-encode: CMP_SHIFT = -4096
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fate-aac-ms-encode: CMP_TARGET = 558
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fate-aac-ms-encode: SIZE_TOLERANCE = 3560
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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
9 years ago
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fate-aac-ms-encode: FUZZ = 15
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#Ticket1784
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FATE_AAC_ENCODE += fate-aac-yoraw-encode
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fate-aac-yoraw-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/yo.raw-short.wav -c:a aac -fflags +bitexact -flags +bitexact
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fate-aac-yoraw-encode: CMP = stddev
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fate-aac-yoraw-encode: REF = $(SAMPLES)/audio-reference/yo.raw-short.wav
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fate-aac-yoraw-encode: CMP_SHIFT = -12288
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fate-aac-yoraw-encode: CMP_TARGET = 226
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fate-aac-yoraw-encode: SIZE_TOLERANCE = 3560
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fate-aac-yoraw-encode: FUZZ = 17
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FATE_AAC_ENCODE += fate-aac-pred-encode
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fate-aac-pred-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -profile:a aac_main -c:a aac -aac_is 0 -aac_pns 0 -aac_ms 0 -aac_tns 0 -b:a 128k -cutoff 22050
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fate-aac-pred-encode: CMP = stddev
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fate-aac-pred-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
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fate-aac-pred-encode: CMP_SHIFT = -4096
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fate-aac-pred-encode: CMP_TARGET = 662
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fate-aac-pred-encode: FUZZ = 12
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fate-aac-pred-encode: SIZE_TOLERANCE = 3560
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FATE_AAC_LATM += fate-aac-latm_000000001180bc60
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fate-aac-latm_000000001180bc60: CMD = pcm -i $(TARGET_SAMPLES)/aac/latm_000000001180bc60.mpg
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fate-aac-latm_000000001180bc60: REF = $(SAMPLES)/aac/latm_000000001180bc60.s16
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FATE_AAC_LATM += fate-aac-latm_stereo_to_51
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fate-aac-latm_stereo_to_51: CMD = pcm -i $(TARGET_SAMPLES)/aac/latm_stereo_to_51.ts -channel_layout 5.1
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fate-aac-latm_stereo_to_51: REF = $(SAMPLES)/aac/latm_stereo_to_51_ref.s16
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fate-aac-autobsf-adtstoasc: CMD = transcode "aac" $(TARGET_SAMPLES)/audiomatch/tones_afconvert_16000_mono_aac_lc.adts \
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matroska "-c:a copy" "-c:a copy"
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FATE_AAC-$(call DEMDEC, AAC, AAC) += $(FATE_AAC_CT_RAW)
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FATE_AAC-$(call DEMDEC, MOV, AAC) += $(FATE_AAC)
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FATE_AAC_LATM-$(call DEMDEC, MPEGTS, AAC_LATM) += $(FATE_AAC_LATM)
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FATE_AAC-$(call DEMDEC, AAC, AAC_FIXED)+= $(FATE_AAC_FIXED)
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FATE_AAC_ALL = $(FATE_AAC-yes) $(FATE_AAC_LATM-yes) $(FATE_AAC_FIXED-yes)
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$(FATE_AAC_ALL): CMP = oneoff
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$(FATE_AAC_ALL): FUZZ = 2
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FATE_AAC_ENCODE-$(call ENCMUX, AAC, ADTS) += $(FATE_AAC_ENCODE)
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FATE_AAC_BSF-$(call ALLYES, AAC_DEMUXER AAC_ADTSTOASC_BSF MATROSKA_MUXER) += fate-aac-autobsf-adtstoasc
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FATE_SAMPLES_FFMPEG += $(FATE_AAC_ALL) $(FATE_AAC_ENCODE-yes) $(FATE_AAC_BSF-yes)
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fate-aac: $(FATE_AAC_ALL) $(FATE_AAC_ENCODE) $(FATE_AAC_BSF-yes)
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fate-aac-latm: $(FATE_AAC_LATM-yes)
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