You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.

249 lines
12 KiB

FATE_AAC += fate-aac-al04_44
fate-aac-al04_44: CMD = pcm -i $(TARGET_SAMPLES)/aac/al04_44.mp4
fate-aac-al04_44: REF = $(SAMPLES)/aac/al04_44.s16
FATE_AAC += fate-aac-al05_44
fate-aac-al05_44: CMD = pcm -i $(TARGET_SAMPLES)/aac/al05_44.mp4
fate-aac-al05_44: REF = $(SAMPLES)/aac/al05_44.s16
FATE_AAC += fate-aac-al06_44
fate-aac-al06_44: CMD = pcm -i $(TARGET_SAMPLES)/aac/al06_44.mp4
fate-aac-al06_44: REF = $(SAMPLES)/aac/al06_44_reorder.s16
FATE_AAC += fate-aac-al07_96
fate-aac-al07_96: CMD = pcm -i $(TARGET_SAMPLES)/aac/al07_96.mp4
fate-aac-al07_96: REF = $(SAMPLES)/aac/al07_96_reorder.s16
FATE_AAC += fate-aac-al15_44
fate-aac-al15_44: CMD = pcm -i $(TARGET_SAMPLES)/aac/al15_44.mp4
fate-aac-al15_44: REF = $(SAMPLES)/aac/al15_44_reorder.s16
FATE_AAC += fate-aac-al17_44
fate-aac-al17_44: CMD = pcm -i $(TARGET_SAMPLES)/aac/al17_44.mp4
fate-aac-al17_44: REF = $(SAMPLES)/aac/al17_44.s16
FATE_AAC += fate-aac-al18_44
fate-aac-al18_44: CMD = pcm -i $(TARGET_SAMPLES)/aac/al18_44.mp4
fate-aac-al18_44: REF = $(SAMPLES)/aac/al18_44.s16
FATE_AAC += fate-aac-am00_88
fate-aac-am00_88: CMD = pcm -i $(TARGET_SAMPLES)/aac/am00_88.mp4
fate-aac-am00_88: REF = $(SAMPLES)/aac/am00_88.s16
FATE_AAC += fate-aac-am05_44
fate-aac-am05_44: CMD = pcm -i $(TARGET_SAMPLES)/aac/am05_44.mp4
fate-aac-am05_44: REF = $(SAMPLES)/aac/am05_44_reorder.s16
FATE_AAC += fate-aac-al_sbr_hq_cm_48_2
fate-aac-al_sbr_hq_cm_48_2: CMD = pcm -i $(TARGET_SAMPLES)/aac/al_sbr_cm_48_2.mp4
fate-aac-al_sbr_hq_cm_48_2: REF = $(SAMPLES)/aac/al_sbr_hq_cm_48_2.s16
FATE_AAC += fate-aac-al_sbr_hq_cm_48_5.1
fate-aac-al_sbr_hq_cm_48_5.1: CMD = pcm -i $(TARGET_SAMPLES)/aac/al_sbr_cm_48_5.1.mp4
fate-aac-al_sbr_hq_cm_48_5.1: REF = $(SAMPLES)/aac/al_sbr_hq_cm_48_5.1_reorder.s16
FATE_AAC += fate-aac-al_sbr_hq_sr_48_2_fsaac48
fate-aac-al_sbr_hq_sr_48_2_fsaac48: CMD = pcm -i $(TARGET_SAMPLES)/aac/al_sbr_sr_48_2_fsaac48.mp4
fate-aac-al_sbr_hq_sr_48_2_fsaac48: REF = $(SAMPLES)/aac/al_sbr_hq_sr_48_2_fsaac48.s16
FATE_AAC += fate-aac-al_sbr_ps_06_ur
fate-aac-al_sbr_ps_06_ur: CMD = pcm -i $(TARGET_SAMPLES)/aac/al_sbr_ps_06_new.mp4
fate-aac-al_sbr_ps_06_ur: REF = $(SAMPLES)/aac/al_sbr_ps_06_ur.s16
FATE_AAC += fate-aac-ap05_48
fate-aac-ap05_48: CMD = pcm -i $(TARGET_SAMPLES)/aac/ap05_48.mp4
fate-aac-ap05_48: REF = $(SAMPLES)/aac/ap05_48.s16
FATE_AAC += fate-aac-er_ad6000np_44_ep0
fate-aac-er_ad6000np_44_ep0: CMD = pcm -i $(TARGET_SAMPLES)/aac/er_ad6000np_44_ep0.mp4
fate-aac-er_ad6000np_44_ep0: REF = $(SAMPLES)/aac/er_ad6000np_44.s16
FATE_AAC += fate-aac-er_eld1001np_44_ep0
fate-aac-er_eld1001np_44_ep0: CMD = pcm -i $(TARGET_SAMPLES)/aac/er_eld1001np_44_ep0.mp4
fate-aac-er_eld1001np_44_ep0: REF = $(SAMPLES)/aac/er_eld1001np_44.s16
FATE_AAC += fate-aac-er_eld2000np_48_ep0
fate-aac-er_eld2000np_48_ep0: CMD = pcm -i $(TARGET_SAMPLES)/aac/er_eld2000np_48_ep0.mp4
fate-aac-er_eld2000np_48_ep0: REF = $(SAMPLES)/aac/er_eld2000np_48_ep0.s16
FATE_AAC += fate-aac-er_eld2100np_48_ep0
fate-aac-er_eld2100np_48_ep0: CMD = pcm -i $(TARGET_SAMPLES)/aac/er_eld2100np_48_ep0.mp4
fate-aac-er_eld2100np_48_ep0: REF = $(SAMPLES)/aac/er_eld2100np_48.s16
FATE_AAC_FIXED += fate-aac-fixed-al04_44
fate-aac-fixed-al04_44: CMD = pcm -c aac_fixed -i $(TARGET_SAMPLES)/aac/al04_44.mp4
fate-aac-fixed-al04_44: REF = $(SAMPLES)/aac/al04_44.s16
FATE_AAC_FIXED += fate-aac-fixed-al05_44
fate-aac-fixed-al05_44: CMD = pcm -c aac_fixed -i $(TARGET_SAMPLES)/aac/al05_44.mp4
fate-aac-fixed-al05_44: REF = $(SAMPLES)/aac/al05_44.s16
FATE_AAC_FIXED += fate-aac-fixed-al06_44
fate-aac-fixed-al06_44: CMD = pcm -c aac_fixed -i $(TARGET_SAMPLES)/aac/al06_44.mp4
fate-aac-fixed-al06_44: REF = $(SAMPLES)/aac/al06_44_reorder.s16
FATE_AAC_FIXED += fate-aac-fixed-al15_44
fate-aac-fixed-al15_44: CMD = pcm -c aac_fixed -i $(TARGET_SAMPLES)/aac/al15_44.mp4
fate-aac-fixed-al15_44: REF = $(SAMPLES)/aac/al15_44_reorder.s16
FATE_AAC_FIXED += fate-aac-fixed-al17_44
fate-aac-fixed-al17_44: CMD = pcm -c aac_fixed -i $(TARGET_SAMPLES)/aac/al17_44.mp4
fate-aac-fixed-al17_44: REF = $(SAMPLES)/aac/al17_44.s16
FATE_AAC_FIXED += fate-aac-fixed-al18_44
fate-aac-fixed-al18_44: CMD = pcm -c aac_fixed -i $(TARGET_SAMPLES)/aac/al18_44.mp4
fate-aac-fixed-al18_44: REF = $(SAMPLES)/aac/al18_44.s16
FATE_AAC_FIXED += fate-aac-fixed-al_sbr_hq_cm_48_2
fate-aac-fixed-al_sbr_hq_cm_48_2: CMD = pcm -c aac_fixed -i $(TARGET_SAMPLES)/aac/al_sbr_cm_48_2.mp4
fate-aac-fixed-al_sbr_hq_cm_48_2: REF = $(SAMPLES)/aac/al_sbr_hq_cm_48_2.s16
FATE_AAC_FIXED += fate-aac-fixed-al_sbr_hq_cm_48_5.1
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
fate-aac-fixed-al_sbr_hq_cm_48_5.1: REF = $(SAMPLES)/aac/al_sbr_hq_cm_48_5.1_reorder.s16
FATE_AAC_FIXED += fate-aac-fixed-al_sbr_hq_sr_48_2_fsaac48
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
fate-aac-fixed-al_sbr_hq_sr_48_2_fsaac48: REF = $(SAMPLES)/aac/al_sbr_hq_sr_48_2_fsaac48.s16
#FATE_AAC_FIXED += fate-aac-fixed-al_sbr_ps_06_ur
#fate-aac-fixed-al_sbr_ps_06_ur: CMD = pcm -c aac_fixed-i $(TARGET_SAMPLES)/aac/al_sbr_ps_06_new.mp4
#fate-aac-fixed-al_sbr_ps_06_ur: REF = $(SAMPLES)/aac/al_sbr_ps_06_ur.s16
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
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
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
fate-aac-ct%: CMD = pcm -i $(TARGET_SAMPLES)/aac/CT_DecoderCheck/$(@:fate-aac-ct-%=%)
fate-aac-ct%: REF = $(SAMPLES)/aac/CT_DecoderCheck/aacPlusv2.wav
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) -strict -2 -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 = 669
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 -strict -2 -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 = 61
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
fate-aac-ln-encode: FUZZ = 30
FATE_AAC_ENCODE += fate-aac-ln-encode-128k
fate-aac-ln-encode-128k: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -strict -2 -c:a aac -aac_is 0 -aac_pns 0 -aac_ms 0 -aac_tns 0 -b:a 128k -cutoff 22050
fate-aac-ln-encode-128k: CMP = stddev
fate-aac-ln-encode-128k: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
fate-aac-ln-encode-128k: CMP_SHIFT = -4096
fate-aac-ln-encode-128k: CMP_TARGET = 800
fate-aac-ln-encode-128k: SIZE_TOLERANCE = 3560
fate-aac-ln-encode-128k: FUZZ = 5
FATE_AAC_ENCODE += fate-aac-pns-encode
fate-aac-pns-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -strict -2 -c:a aac -aac_pns 1 -aac_is 0 -aac_ms 0 -aac_tns 0 -b:a 128k -cutoff 22050 -fflags +bitexact -flags +bitexact
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 = 662
fate-aac-pns-encode: SIZE_TOLERANCE = 3560
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-pns-encode: FUZZ = 72
FATE_AAC_ENCODE += fate-aac-tns-encode
fate-aac-tns-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -strict -2 -c:a aac -aac_tns 1 -aac_is 0 -aac_pns 0 -aac_ms 0 -b:a 128k -cutoff 22050
fate-aac-tns-encode: CMP = stddev
fate-aac-tns-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
fate-aac-tns-encode: CMP_SHIFT = -4096
fate-aac-tns-encode: CMP_TARGET = 817
fate-aac-tns-encode: FUZZ = 7
fate-aac-tns-encode: SIZE_TOLERANCE = 3560
FATE_AAC_ENCODE += fate-aac-is-encode
fate-aac-is-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -strict -2 -c:a aac -aac_pns 0 -aac_is 1 -aac_ms 0 -b:a 128k -aac_tns 0 -cutoff 22050
fate-aac-is-encode: CMP = stddev
fate-aac-is-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
fate-aac-is-encode: CMP_SHIFT = -4096
fate-aac-is-encode: CMP_TARGET = 615
fate-aac-is-encode: SIZE_TOLERANCE = 3560
fate-aac-is-encode: FUZZ = 10
FATE_AAC_ENCODE += fate-aac-ms-encode
fate-aac-ms-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -strict -2 -c:a aac -aac_pns 0 -aac_is 0 -aac_ms 1 -aac_tns 0 -b:a 128k -cutoff 22050
fate-aac-ms-encode: CMP = stddev
fate-aac-ms-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
fate-aac-ms-encode: CMP_SHIFT = -4096
fate-aac-ms-encode: CMP_TARGET = 675
fate-aac-ms-encode: SIZE_TOLERANCE = 3560
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-ms-encode: FUZZ = 15
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 36k -fflags +bitexact -flags +bitexact
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 = 1270
fate-aac-ltp-encode: SIZE_TOLERANCE = 3560
fate-aac-ltp-encode: FUZZ = 17
FATE_AAC_ENCODE += fate-aac-pred-encode
fate-aac-pred-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -strict -2 -profile:a aac_main -c:a aac -aac_is 0 -aac_pns 0 -aac_ms 0 -aac_tns 0 -b:a 128k -cutoff 22050
fate-aac-pred-encode: CMP = stddev
fate-aac-pred-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
fate-aac-pred-encode: CMP_SHIFT = -4096
fate-aac-pred-encode: CMP_TARGET = 841
fate-aac-pred-encode: FUZZ = 12
fate-aac-pred-encode: SIZE_TOLERANCE = 3560
FATE_AAC_LATM += fate-aac-latm_000000001180bc60
fate-aac-latm_000000001180bc60: CMD = pcm -i $(TARGET_SAMPLES)/aac/latm_000000001180bc60.mpg
fate-aac-latm_000000001180bc60: REF = $(SAMPLES)/aac/latm_000000001180bc60.s16
FATE_AAC_LATM += fate-aac-latm_stereo_to_51
fate-aac-latm_stereo_to_51: CMD = pcm -i $(TARGET_SAMPLES)/aac/latm_stereo_to_51.ts -channel_layout 5.1
fate-aac-latm_stereo_to_51: REF = $(SAMPLES)/aac/latm_stereo_to_51_ref.s16
FATE_AAC-$(call DEMDEC, AAC, AAC) += $(FATE_AAC_CT_RAW)
FATE_AAC-$(call DEMDEC, MOV, AAC) += $(FATE_AAC)
FATE_AAC_LATM-$(call DEMDEC, MPEGTS, AAC_LATM) += $(FATE_AAC_LATM)
FATE_AAC-$(call DEMDEC, AAC, AAC_FIXED)+= $(FATE_AAC_FIXED)
FATE_AAC_ALL = $(FATE_AAC-yes) $(FATE_AAC_LATM-yes) $(FATE_AAC_FIXED-yes)
Merge remote-tracking branch 'qatar/master' * qatar/master: (27 commits) libxvid: Give more suitable names to libxvid-related files. libxvid: Separate libxvid encoder from libxvid rate control code. jpeglsdec: Remove write-only variable in ff_jpegls_decode_lse(). fate: cosmetics: lowercase some comments fate: Give more consistent names to some RealVideo/RealAudio tests. lavfi: add avfilter_get_audio_buffer_ref_from_arrays(). lavfi: add extended_data to AVFilterBuffer. lavc: check that extended_data is properly set in avcodec_encode_audio2(). lavc: pad last audio frame with silence when needed. samplefmt: add a function for filling a buffer with silence. samplefmt: add a function for copying audio samples. lavr: do not try to copy to uninitialized output audio data. lavr: make avresample_read() with NULL output discard samples. fate: split idroq audio and video into separate tests fate: improve dependencies fate: add convenient shorthands for ea-vp6, libavcodec, libavutil tests fate: split some combined tests into separate audio and video tests fate: fix dependencies for probe tests mips: intreadwrite: fix inline asm for gcc 4.8 mips: intreadwrite: remove unnecessary inline asm ... Conflicts: cmdutils.h configure doc/APIchanges doc/filters.texi ffmpeg.c ffplay.c libavcodec/internal.h libavcodec/jpeglsdec.c libavcodec/libschroedingerdec.c libavcodec/libxvid.c libavcodec/libxvid_rc.c libavcodec/utils.c libavcodec/version.h libavfilter/avfilter.c libavfilter/avfilter.h libavfilter/buffersink.h tests/Makefile tests/fate/aac.mak tests/fate/audio.mak tests/fate/demux.mak tests/fate/ea.mak tests/fate/image.mak tests/fate/libavutil.mak tests/fate/lossless-audio.mak tests/fate/lossless-video.mak tests/fate/microsoft.mak tests/fate/qt.mak tests/fate/real.mak tests/fate/screen.mak tests/fate/video.mak tests/fate/voice.mak tests/fate/vqf.mak tests/ref/fate/ea-mad tests/ref/fate/ea-tqi Merged-by: Michael Niedermayer <michaelni@gmx.at>
13 years ago
$(FATE_AAC_ALL): CMP = oneoff
$(FATE_AAC_ALL): FUZZ = 2
FATE_AAC_ENCODE-$(call ENCMUX, AAC, ADTS) += $(FATE_AAC_ENCODE)
FATE_SAMPLES_FFMPEG += $(FATE_AAC_ALL) $(FATE_AAC_ENCODE-yes)
fate-aac: $(FATE_AAC_ALL) $(FATE_AAC_ENCODE)
fate-aac-latm: $(FATE_AAC_LATM-yes)