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FFmpeg/tests/fate/filter-audio.mak

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fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, ADELAY ARESAMPLE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-adelay
fate: add adelay test Signed-off-by: Paul B Mahol <onemda@gmail.com>
2013-09-19 20:32:09 +03:00
fate-filter-adelay: tests/data/asynth-44100-2.wav
fate-filter-adelay: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate: disable automatic conversions on many tests. Explicitly insert the scale or aresample filter where it would have been inserted by the negotiation. Re-enable conversions if it cannot be done easily. If a conversion is needed in a test, we want to know about it. If the negotiation changes and makes new conversion necessary, we want to know about it even more.
2020-08-14 18:58:27 +02:00
fate-filter-adelay: CMD = framecrc -i $(SRC) -af aresample,adelay=42,aresample
fate: add adelay test Signed-off-by: Paul B Mahol <onemda@gmail.com>
2013-09-19 20:32:09 +03:00
fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, AECHO ARESAMPLE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-aecho
fate: add aecho test Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-05-22 09:45:49 +02:00
fate-filter-aecho: tests/data/asynth-44100-2.wav
fate-filter-aecho: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate: disable automatic conversions on many tests. Explicitly insert the scale or aresample filter where it would have been inserted by the negotiation. Re-enable conversions if it cannot be done easily. If a conversion is needed in a test, we want to know about it. If the negotiation changes and makes new conversion necessary, we want to know about it even more.
2020-08-14 18:58:27 +02:00
fate-filter-aecho: CMD = framecrc -i $(SRC) -af aresample,aecho=0.5:0.5:32:0.5,aresample
fate: add aecho test Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-05-22 09:45:49 +02:00
fate: add aemphasis test Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-05-28 17:20:25 +02:00
FATE_FILTER_AEMPHASIS += fate-filter-aemphasis-50fm
fate-filter-aemphasis-50fm: tests/data/asynth-44100-2.wav
fate-filter-aemphasis-50fm: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate: disable automatic conversions on many tests. Explicitly insert the scale or aresample filter where it would have been inserted by the negotiation. Re-enable conversions if it cannot be done easily. If a conversion is needed in a test, we want to know about it. If the negotiation changes and makes new conversion necessary, we want to know about it even more.
2020-08-14 18:58:27 +02:00
fate-filter-aemphasis-50fm: CMD = framecrc -i $(SRC) -af aresample,aemphasis=1:5:reproduction:50fm,aresample
fate: add aemphasis test Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-05-28 17:20:25 +02:00
FATE_FILTER_AEMPHASIS += fate-filter-aemphasis-75kf
fate-filter-aemphasis-75kf: tests/data/asynth-44100-2.wav
fate-filter-aemphasis-75kf: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate: disable automatic conversions on many tests. Explicitly insert the scale or aresample filter where it would have been inserted by the negotiation. Re-enable conversions if it cannot be done easily. If a conversion is needed in a test, we want to know about it. If the negotiation changes and makes new conversion necessary, we want to know about it even more.
2020-08-14 18:58:27 +02:00
fate-filter-aemphasis-75kf: CMD = framecrc -i $(SRC) -af aresample,aemphasis=2:8:reproduction:75kf,aresample
fate: add aemphasis test Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-05-28 17:20:25 +02:00
fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, AEMPHASIS ARESAMPLE, WAV, PCM_S16LE, PCM_S16LE, WAV) += $(FATE_FILTER_AEMPHASIS)
fate: add aemphasis test Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-05-28 17:20:25 +02:00
fate: add afade test Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-11 13:30:09 +02:00
FATE_FILTER_AFADE += fate-filter-afade-qsin
fate-filter-afade-qsin: tests/data/asynth-44100-2.wav
fate-filter-afade-qsin: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-afade-qsin: CMD = framecrc -i $(SRC) -af afade=t=in:ss=0:d=2:curve=qsin
FATE_FILTER_AFADE += fate-filter-afade-iqsin
fate-filter-afade-iqsin: tests/data/asynth-44100-2.wav
fate-filter-afade-iqsin: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-afade-iqsin: CMD = framecrc -i $(SRC) -af afade=t=in:ss=0:d=2:curve=iqsin
FATE_FILTER_AFADE += fate-filter-afade-esin
fate-filter-afade-esin: tests/data/asynth-44100-2.wav
fate-filter-afade-esin: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-afade-esin: CMD = framecrc -i $(SRC) -af afade=t=in:ss=0:d=2:curve=esin
FATE_FILTER_AFADE += fate-filter-afade-hsin
fate-filter-afade-hsin: tests/data/asynth-44100-2.wav
fate-filter-afade-hsin: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-afade-hsin: CMD = framecrc -i $(SRC) -af afade=t=in:ss=0:d=2:curve=hsin
FATE_FILTER_AFADE += fate-filter-afade-exp
fate-filter-afade-exp: tests/data/asynth-44100-2.wav
fate-filter-afade-exp: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-afade-exp: CMD = framecrc -i $(SRC) -af afade=t=in:ss=0:d=2:curve=exp
FATE_FILTER_AFADE += fate-filter-afade-log
fate-filter-afade-log: tests/data/asynth-44100-2.wav
fate-filter-afade-log: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-afade-log: CMD = framecrc -i $(SRC) -af afade=t=in:ss=1:d=2.5:curve=log
FATE_AFILTER-$(call FILTERDEMDECENCMUX, AFADE, WAV, PCM_S16LE, PCM_S16LE, WAV) += $(FATE_FILTER_AFADE)
fate: fate-filter-acrossfade needs samples Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-14 04:11:24 +02:00
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECENCMUX, ACROSSFADE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-acrossfade
fate: add afade test Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-11 13:30:09 +02:00
fate-filter-acrossfade: tests/data/asynth-44100-2.wav
fate-filter-acrossfade: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-acrossfade: SRC2 = $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
fate-filter-acrossfade: CMD = framecrc -i $(SRC) -i $(SRC2) -filter_complex acrossfade=d=2:c1=log:c2=exp
fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, AGATE ARESAMPLE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-agate
fate: add test for agate Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-16 17:12:29 +02:00
fate-filter-agate: tests/data/asynth-44100-2.wav
fate-filter-agate: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate: disable automatic conversions on many tests. Explicitly insert the scale or aresample filter where it would have been inserted by the negotiation. Re-enable conversions if it cannot be done easily. If a conversion is needed in a test, we want to know about it. If the negotiation changes and makes new conversion necessary, we want to know about it even more.
2020-08-14 18:58:27 +02:00
fate-filter-agate: CMD = framecrc -i $(SRC) -af aresample,agate=level_in=10:range=0:threshold=1:ratio=1:attack=1:knee=1:makeup=4,aresample
fate: add test for agate Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-16 17:12:29 +02:00
fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, ALIMITER ARESAMPLE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-alimiter
fate: add test for alimiter Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-20 19:32:36 +02:00
fate-filter-alimiter: tests/data/asynth-44100-2.wav
fate-filter-alimiter: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate: disable automatic conversions on many tests. Explicitly insert the scale or aresample filter where it would have been inserted by the negotiation. Re-enable conversions if it cannot be done easily. If a conversion is needed in a test, we want to know about it. If the negotiation changes and makes new conversion necessary, we want to know about it even more.
2020-08-14 18:58:27 +02:00
fate-filter-alimiter: CMD = framecrc -i $(SRC) -af aresample,alimiter=level_in=1:level_out=2:limit=0.2,aresample
fate: add test for alimiter Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-20 19:32:36 +02:00
fate: add test for amerge Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-21 14:43:01 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, AMERGE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-amerge
fate-filter-amerge: tests/data/asynth-44100-1.wav
fate-filter-amerge: SRC = $(TARGET_PATH)/tests/data/asynth-44100-1.wav
fate-filter-amerge: CMD = framecrc -i $(SRC) -i $(SRC) -filter_complex "[0:a][1:a]amerge=inputs=2[aout]" -map "[aout]"
fate: add apad test Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-22 15:24:26 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, APAD, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-apad
fate-filter-apad: tests/data/asynth-44100-2.wav
fate-filter-apad: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-apad: CMD = framecrc -i $(SRC) -af apad=pad_len=10
fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, ANEQUALIZER ARESAMPLE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-anequalizer
fate: add anequalizer test Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-22 12:58:14 +02:00
fate-filter-anequalizer: tests/data/asynth-44100-2.wav
fate-filter-anequalizer: tests/data/filtergraphs/anequalizer
fate-filter-anequalizer: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fftools/ffmpeg: deprecate -filter_complex_script It is equivalent to -/filter_complex.
2024-01-12 16:54:11 +02:00
fate-filter-anequalizer: CMD = framecrc -auto_conversion_filters -i $(SRC) -/filter_complex $(TARGET_PATH)/tests/data/filtergraphs/anequalizer
fate: add anequalizer test Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-22 12:58:14 +02:00
fate: add test for asetnsamples filter with padding disabled Adds another test for asetnsamples filter where padding of the last frame is switched off. Renames the existing test to make the difference obvious. Tested-by: Michael Niedermayer <michael@niedermayer.cc> Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
2017-08-07 11:55:39 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, ASETNSAMPLES, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-asetnsamples-pad
fate-filter-asetnsamples-pad: tests/data/asynth-44100-2.wav
fate-filter-asetnsamples-pad: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-asetnsamples-pad: CMD = framecrc -i $(SRC) -af asetnsamples=512:p=1
FATE_AFILTER-$(call FILTERDEMDECENCMUX, ASETNSAMPLES, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-asetnsamples-nopad
fate-filter-asetnsamples-nopad: tests/data/asynth-44100-2.wav
fate-filter-asetnsamples-nopad: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-asetnsamples-nopad: CMD = framecrc -i $(SRC) -af asetnsamples=512:p=0
fate: add test for asetnsamples Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-06-29 16:26:19 +02:00
fate: add test for asetrate Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-07-04 17:51:54 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, ASETRATE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-asetrate
fate-filter-asetrate: tests/data/asynth-44100-2.wav
fate-filter-asetrate: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
Merge commit '043b0b9fb1481053b712d06d2c5b772f1845b72b' * commit '043b0b9fb1481053b712d06d2c5b772f1845b72b': Replace leftover uses of -aframes|-dframes|-vframes with -frames:a|d|v The merge also includes all our own occurences. Merged-by: Clément Bœsch <u@pkh.me>
2017-03-24 12:37:44 +02:00
fate-filter-asetrate: CMD = framecrc -i $(SRC) -frames:a 20 -af asetrate=20000
fate: add test for asetrate Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-07-04 17:51:54 +02:00
fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, CHORUS ARESAMPLE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-chorus
fate: add test for chorus filter Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-07-19 21:18:08 +02:00
fate-filter-chorus: tests/data/asynth-22050-1.wav
fate-filter-chorus: SRC = $(TARGET_PATH)/tests/data/asynth-22050-1.wav
fate: disable automatic conversions on many tests. Explicitly insert the scale or aresample filter where it would have been inserted by the negotiation. Re-enable conversions if it cannot be done easily. If a conversion is needed in a test, we want to know about it. If the negotiation changes and makes new conversion necessary, we want to know about it even more.
2020-08-14 18:58:27 +02:00
fate-filter-chorus: CMD = framecrc -i $(SRC) -frames:a 10 -af aresample,chorus=0.050001:0.050002:64:0.050001:0.025003:2.00004,aresample
fate: add test for chorus filter Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-07-19 21:18:08 +02:00
fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, CRYSTALIZER ARESAMPLE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-crystalizer
fate: add crystalizer test
2022-03-17 13:28:48 +02:00
fate-filter-crystalizer: tests/data/asynth-44100-2.wav
fate-filter-crystalizer: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-crystalizer: CMD = framecrc -i $(SRC) -af aresample,crystalizer=2:0,crystalizer=-2:0,aresample
fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, DCSHIFT ARESAMPLE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-dcshift
fate: add test for dcshift filter Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-07-20 18:32:04 +02:00
fate-filter-dcshift: tests/data/asynth-44100-2.wav
fate-filter-dcshift: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate: disable automatic conversions on many tests. Explicitly insert the scale or aresample filter where it would have been inserted by the negotiation. Re-enable conversions if it cannot be done easily. If a conversion is needed in a test, we want to know about it. If the negotiation changes and makes new conversion necessary, we want to know about it even more.
2020-08-14 18:58:27 +02:00
fate-filter-dcshift: CMD = framecrc -i $(SRC) -frames:a 20 -af aresample,dcshift=shift=0.25:limitergain=0.05,aresample
fate: add test for dcshift filter Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-07-20 18:32:04 +02:00
fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, EARWAX ARESAMPLE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-earwax
fate: add test for earwax filter Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-07-20 18:41:58 +02:00
fate-filter-earwax: tests/data/asynth-44100-2.wav
fate-filter-earwax: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
avfilter/af_earwax: fix filter behavior Previous filter output was incorrect. New one actually follows graph in comments described on side of filter taps.
2020-12-04 18:54:05 +02:00
fate-filter-earwax: CMD = framecrc -i $(SRC) -frames:a 20 -af aresample,earwax,aresample
fate: add test for earwax filter Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-07-20 18:41:58 +02:00
fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, EXTRASTEREO ARESAMPLE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-extrastereo
fate: add test for extrastereo filter Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-07-21 07:21:01 +02:00
fate-filter-extrastereo: tests/data/asynth-44100-2.wav
fate-filter-extrastereo: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate: disable automatic conversions on many tests. Explicitly insert the scale or aresample filter where it would have been inserted by the negotiation. Re-enable conversions if it cannot be done easily. If a conversion is needed in a test, we want to know about it. If the negotiation changes and makes new conversion necessary, we want to know about it even more.
2020-08-14 18:58:27 +02:00
fate-filter-extrastereo: CMD = framecrc -i $(SRC) -frames:a 20 -af aresample,extrastereo=m=2,aresample
fate: add test for extrastereo filter Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-07-21 07:21:01 +02:00
fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, FIREQUALIZER ATRIM VOLUME ARESAMPLE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-firequalizer
fate: add test for firequalizer filter Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
2016-10-13 03:43:50 +02:00
fate-filter-firequalizer: tests/data/asynth-44100-2.wav
fate-filter-firequalizer: tests/data/filtergraphs/firequalizer
fate-filter-firequalizer: REF = tests/data/asynth-44100-2.wav
fftools/ffmpeg: deprecate -filter_script It is equivalent to -/filter.
2024-01-12 16:54:11 +02:00
fate-filter-firequalizer: CMD = ffmpeg -auto_conversion_filters -i $(TARGET_PATH)/tests/data/asynth-44100-2.wav -/filter $(TARGET_PATH)/tests/data/filtergraphs/firequalizer -f wav -c:a pcm_s16le -
fate: add test for firequalizer filter Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
2016-10-13 03:43:50 +02:00
fate-filter-firequalizer: CMP = oneoff
fate-filter-firequalizer: CMP_UNIT = s16
fate-filter-firequalizer: SIZE_TOLERANCE = 1058400 - 1097208
fate: add tests for pan audio filter Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
2018-02-15 09:21:24 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, PAN, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-pan-mono1
fate-filter-pan-mono1: tests/data/asynth-44100-2.wav
fate-filter-pan-mono1: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-pan-mono1: CMD = framecrc -ss 3.14 -i $(SRC) -frames:a 20 -filter:a "pan=mono|FC=FL"
FATE_AFILTER-$(call FILTERDEMDECENCMUX, PAN, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-pan-mono2
fate-filter-pan-mono2: tests/data/asynth-44100-2.wav
fate-filter-pan-mono2: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-pan-mono2: CMD = framecrc -ss 3.14 -i $(SRC) -frames:a 20 -filter:a "pan=1C|c0=c0+c1"
FATE_AFILTER-$(call FILTERDEMDECENCMUX, PAN, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-pan-stereo1
fate-filter-pan-stereo1: tests/data/asynth-44100-3.wav
fate-filter-pan-stereo1: SRC = $(TARGET_PATH)/tests/data/asynth-44100-3.wav
fate-filter-pan-stereo1: CMD = framecrc -ss 3.14 -i $(SRC) -frames:a 20 -filter:a "pan=2c|FL=FR|FR=FL"
FATE_AFILTER-$(call FILTERDEMDECENCMUX, PAN, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-pan-stereo2
fate-filter-pan-stereo2: tests/data/asynth-44100-3.wav
fate-filter-pan-stereo2: SRC = $(TARGET_PATH)/tests/data/asynth-44100-3.wav
fate-filter-pan-stereo2: CMD = framecrc -ss 3.14 -i $(SRC) -frames:a 20 -filter:a "pan=stereo|c0=c0-c2|c1=c1-c2"
FATE_AFILTER-$(call FILTERDEMDECENCMUX, PAN, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-pan-stereo3
fate-filter-pan-stereo3: tests/data/asynth-44100-2.wav
fate-filter-pan-stereo3: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-pan-stereo3: CMD = framecrc -ss 3.14 -i $(SRC) -frames:a 20 -filter:a "pan=FL+FR|FL<3*c0+2*c1|FR<2*c0+3*c1"
FATE_AFILTER-$(call FILTERDEMDECENCMUX, PAN, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-pan-stereo4
fate-filter-pan-stereo4: tests/data/asynth-44100-2.wav
fate-filter-pan-stereo4: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate: add tests for audio channel up-/downmixing with pan filter Add tests for upmixing and downmixing with audio channel counts that have a corresponding default layout and also tests where there is no default layout. Update the existing "stereo4" test so it actually outputs stereo like the other stereo tests. Rename the previous "stereo4" test into "upmix1". Reviewed-by: Nicolas George <george@nsup.org> Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
2018-07-19 13:59:37 +02:00
fate-filter-pan-stereo4: CMD = framecrc -ss 3.14 -guess_layout_max 0 -i $(SRC) -frames:a 20 -filter:a "pan=2C|c0=c0-0.5*c1|c1=c1+0.5*c0"
FATE_AFILTER-$(call FILTERDEMDECENCMUX, PAN, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-pan-upmix1
fate-filter-pan-upmix1: tests/data/asynth-44100-2.wav
fate-filter-pan-upmix1: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-pan-upmix1: CMD = framecrc -ss 3.14 -guess_layout_max 0 -i $(SRC) -frames:a 20 -filter:a "pan=4C|c0=c0-0.5*c1|c1=c1+0.5*c0|c2=0*c0|c3=0*c0"
FATE_AFILTER-$(call FILTERDEMDECENCMUX, PAN, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-pan-upmix2
fate-filter-pan-upmix2: tests/data/asynth-44100-4.wav
fate-filter-pan-upmix2: SRC = $(TARGET_PATH)/tests/data/asynth-44100-4.wav
fate-filter-pan-upmix2: CMD = framecrc -ss 3.14 -i $(SRC) -frames:a 20 -filter:a "pan=9C|c0=c0-c1|c1=c2+c3|c2=c0+c1|c3=c2-c3|c4=c1-c0|c5=c3+c2|c6=c1+c0|c7=c3-c2|c8=c0-c3"
FATE_AFILTER-$(call FILTERDEMDECENCMUX, PAN, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-pan-downmix1
fate-filter-pan-downmix1: tests/data/asynth-44100-4.wav
fate-filter-pan-downmix1: SRC = $(TARGET_PATH)/tests/data/asynth-44100-4.wav
fate-filter-pan-downmix1: CMD = framecrc -ss 3.14 -i $(SRC) -frames:a 20 -filter:a "pan=2c|FL<FL+0.5*FC+0.6*BL+0.6*SL|FR<FR+0.5*FC+0.6*BR+0.6*SR"
FATE_AFILTER-$(call FILTERDEMDECENCMUX, PAN, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-pan-downmix2
fate-filter-pan-downmix2: tests/data/asynth-44100-11.wav
fate-filter-pan-downmix2: SRC = $(TARGET_PATH)/tests/data/asynth-44100-11.wav
fate-filter-pan-downmix2: CMD = framecrc -ss 3.14 -i $(SRC) -frames:a 20 -filter:a "pan=5C|c0=0.7*c0+0.7*c10|c1=c9|c2=c8|c3=c7|c4=c6"
fate: add tests for pan audio filter Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
2018-02-15 09:21:24 +02:00
fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER-$(call ALLYES, LAVFI_INDEV AEVALSRC_FILTER SILENCEREMOVE_FILTER ARESAMPLE_FILTER) += fate-filter-silenceremove
avfilter/af_silenceremove: add real peak detector Rename old peak detector to more correct name one.
2023-05-25 23:06:50 +02:00
fate-filter-silenceremove: CMD = framecrc -auto_conversion_filters -f lavfi -i "aevalsrc=between(t\,1\,2)+between(t\,4\,5)+between(t\,7\,9):d=10:n=8192,silenceremove=start_periods=0:start_duration=0:start_threshold=0:stop_periods=-1:stop_duration=0:stop_threshold=-90dB:window=0:detection=avg"
fate: add test for silenceremove filter Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-07-28 19:31:55 +02:00
fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECENCMUX, STEREOTOOLS ARESAMPLE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-stereotools
fate: add test for stereotools filter Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-07-28 20:01:40 +02:00
fate-filter-stereotools: SRC = $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
fate: disable automatic conversions on many tests. Explicitly insert the scale or aresample filter where it would have been inserted by the negotiation. Re-enable conversions if it cannot be done easily. If a conversion is needed in a test, we want to know about it. If the negotiation changes and makes new conversion necessary, we want to know about it even more.
2020-08-14 18:58:27 +02:00
fate-filter-stereotools: CMD = framecrc -i $(SRC) -frames:a 20 -af aresample,stereotools=mlev=0.015625,aresample
fate: add test for stereotools filter Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-07-28 20:01:40 +02:00
fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECENCMUX, TREMOLO ATRIM ARESAMPLE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-tremolo
fate: add test for tremolo filter Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-02 09:38:17 +02:00
fate-filter-tremolo: tests/data/asynth-44100-2.wav
fate-filter-tremolo: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate: use atrim filter instead of -frames:a 20 for fate-filter-tremolo To make it independent of incoming wav demuxer packet size. Signed-off-by: Marton Balint <cus@passwd.hu>
2024-03-06 00:04:51 +02:00
fate-filter-tremolo: CMD = ffmpeg -auto_conversion_filters -i $(SRC) -af tremolo,atrim=end_sample=20480 -f wav -f s16le -
fate: Use a oneoff test for the tremolo filter The tremolo filter uses floating point internally, and uses multiplication factors derived from sin(fmod()), neither of which is bitexact for use with framecrc. This fixes running this test when built with for mingw/x86_32 with clang. In this case, a 1 ulp difference in the output from fmod() would end up in an output from the filter that differs by 1 ulp, but which makes the lrint() in swresample/audioconvert.c round in a different direction. Signed-off-by: Martin Storsjö <martin@martin.st>
2019-12-09 22:57:44 +02:00
fate-filter-tremolo: REF = $(SAMPLES)/filter/tremolo.pcm
fate-filter-tremolo: CMP = oneoff
fate-filter-tremolo: CMP_UNIT = s16
fate: add test for tremolo filter Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-02 09:38:17 +02:00
fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, COMPAND ARESAMPLE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-compand
fate: add test for compand filter Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-12 19:00:09 +02:00
fate-filter-compand: tests/data/asynth-44100-2.wav
fate-filter-compand: tests/data/filtergraphs/compand
fate-filter-compand: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fftools/ffmpeg: deprecate -filter_complex_script It is equivalent to -/filter_complex.
2024-01-12 16:54:11 +02:00
fate-filter-compand: CMD = framecrc -auto_conversion_filters -i $(SRC) -frames:a 20 -/filter_complex $(TARGET_PATH)/tests/data/filtergraphs/compand
fate: add test for compand filter Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-08-12 19:00:09 +02:00
fate: fix filter-hls tests dependencies Reviewed-by: Michael Niedermayer <michael@niedermayer.cc> Signed-off-by: James Almer <jamrial@gmail.com>
2016-03-03 05:46:07 +02:00
tests/data/hls-list.m3u8: TAG = GEN
tests/data/hls-list.m3u8: ffmpeg$(PROGSSUF)$(EXESUF) | tests/data
FATE: always pass -nostdin to ffmpeg This avoids making terminal config changes that may not be reverted properly during parallel testing.
2021-12-17 03:04:11 +02:00
$(M)$(TARGET_EXEC) $(TARGET_PATH)/$< -nostdin \
fate: stop using deprecated filter syntax in hls tests Signed-off-by: James Almer <jamrial@gmail.com>
2017-08-30 19:14:54 +02:00
-f lavfi -i "aevalsrc=cos(2*PI*t)*sin(2*PI*(440+4*t)*t):d=20" -f segment -segment_time 10 -map 0 -flags +bitexact -codec:a mp2fixed \
fate: fix filter-hls tests dependencies Reviewed-by: Michael Niedermayer <michael@niedermayer.cc> Signed-off-by: James Almer <jamrial@gmail.com>
2016-03-03 05:46:07 +02:00
-segment_list $(TARGET_PATH)/$@ -y $(TARGET_PATH)/tests/data/hls-out-%03d.ts 2>/dev/null
fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER-$(call ALLYES, HLS_DEMUXER MPEGTS_MUXER MPEGTS_DEMUXER AEVALSRC_FILTER ARESAMPLE_FILTER LAVFI_INDEV MP2FIXED_ENCODER) += fate-filter-hls
fate: fix filter-hls tests dependencies Reviewed-by: Michael Niedermayer <michael@niedermayer.cc> Signed-off-by: James Almer <jamrial@gmail.com>
2016-03-03 05:46:07 +02:00
fate-filter-hls: tests/data/hls-list.m3u8
fate: disable automatic conversions on many tests. Explicitly insert the scale or aresample filter where it would have been inserted by the negotiation. Re-enable conversions if it cannot be done easily. If a conversion is needed in a test, we want to know about it. If the negotiation changes and makes new conversion necessary, we want to know about it even more.
2020-08-14 18:58:27 +02:00
fate-filter-hls: CMD = framecrc -flags +bitexact -i $(TARGET_PATH)/tests/data/hls-list.m3u8 -af aresample
fate: add filter-hls
2016-03-01 03:30:05 +02:00
tests/fate:Add FATE for hls_flags append option add tests/ref/fate/filter-hls-append for FATE add hls-list-append fate use filter make audio data and test hls_flags append options Signed-off-by: Steven Liu <lingjiujianke@gmail.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-07 17:04:46 +02:00
tests/data/hls-list-append.m3u8: TAG = GEN
tests/data/hls-list-append.m3u8: ffmpeg$(PROGSSUF)$(EXESUF) | tests/data
FATE: always pass -nostdin to ffmpeg This avoids making terminal config changes that may not be reverted properly during parallel testing.
2021-12-17 03:04:11 +02:00
$(M)$(TARGET_EXEC) $(TARGET_PATH)/$< -nostdin \
fate: stop using deprecated filter syntax in hls tests Signed-off-by: James Almer <jamrial@gmail.com>
2017-08-30 19:14:54 +02:00
-f lavfi -i "aevalsrc=cos(2*PI*t)*sin(2*PI*(440+4*t)*t):d=20" -f segment -segment_time 10 -map 0 -flags +bitexact -codec:a mp2fixed \
tests/fate:Add FATE for hls_flags append option add tests/ref/fate/filter-hls-append for FATE add hls-list-append fate use filter make audio data and test hls_flags append options Signed-off-by: Steven Liu <lingjiujianke@gmail.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-07 17:04:46 +02:00
-segment_list $(TARGET_PATH)/$@ -y $(TARGET_PATH)/tests/data/hls-append-out-%03d.ts 2>/dev/null; \
FATE: always pass -nostdin to ffmpeg This avoids making terminal config changes that may not be reverted properly during parallel testing.
2021-12-17 03:04:11 +02:00
$(TARGET_EXEC) $(TARGET_PATH)/$< -nostdin \
fate: stop using deprecated filter syntax in hls tests Signed-off-by: James Almer <jamrial@gmail.com>
2017-08-30 19:14:54 +02:00
-f lavfi -i "aevalsrc=cos(2*PI*t)*sin(2*PI*(440+4*t)*t):d=20" -f hls -hls_time 10 -map 0 -flags +bitexact \
tests/fate:Add FATE for hls_flags append option add tests/ref/fate/filter-hls-append for FATE add hls-list-append fate use filter make audio data and test hls_flags append options Signed-off-by: Steven Liu <lingjiujianke@gmail.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-07 17:04:46 +02:00
-hls_flags append_list -codec:a mp2fixed -hls_segment_filename $(TARGET_PATH)/tests/data/hls-append-out-%03d.ts \
$(TARGET_PATH)/tests/data/hls-list-append.m3u8 2>/dev/null
fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER-$(call ALLYES, HLS_DEMUXER MPEGTS_MUXER MPEGTS_DEMUXER AEVALSRC_FILTER ARESAMPLE_FILTER LAVFI_INDEV MP2FIXED_ENCODER) += fate-filter-hls-append
tests/fate:Add FATE for hls_flags append option add tests/ref/fate/filter-hls-append for FATE add hls-list-append fate use filter make audio data and test hls_flags append options Signed-off-by: Steven Liu <lingjiujianke@gmail.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-07 17:04:46 +02:00
fate-filter-hls-append: tests/data/hls-list-append.m3u8
fate: disable automatic conversions on many tests. Explicitly insert the scale or aresample filter where it would have been inserted by the negotiation. Re-enable conversions if it cannot be done easily. If a conversion is needed in a test, we want to know about it. If the negotiation changes and makes new conversion necessary, we want to know about it even more.
2020-08-14 18:58:27 +02:00
fate-filter-hls-append: CMD = framecrc -flags +bitexact -i $(TARGET_PATH)/tests/data/hls-list-append.m3u8 -af asetpts=N*23,aresample
tests/fate:Add FATE for hls_flags append option add tests/ref/fate/filter-hls-append for FATE add hls-list-append fate use filter make audio data and test hls_flags append options Signed-off-by: Steven Liu <lingjiujianke@gmail.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-09-07 17:04:46 +02:00
fate: Split audio filters into their own separate file
2013-04-16 17:38:02 +03:00
FATE_AMIX += fate-filter-amix-simple
fate: use a fixed wav demux packet size for amix tests The dropout transition feature of the amix filter depends on the incoming packet size. Signed-off-by: Marton Balint <cus@passwd.hu>
2024-03-05 23:46:34 +02:00
fate-filter-amix-simple: CMD = ffmpeg -auto_conversion_filters -filter_complex amix -max_size 4096 -i $(SRC) -ss 3 -max_size 4096 -i $(SRC1) -f f32le -
fate: Split audio filters into their own separate file
2013-04-16 17:38:02 +03:00
fate-filter-amix-simple: REF = $(SAMPLES)/filter/amix_simple.pcm
FATE_AMIX += fate-filter-amix-first
fate: use a fixed wav demux packet size for amix tests The dropout transition feature of the amix filter depends on the incoming packet size. Signed-off-by: Marton Balint <cus@passwd.hu>
2024-03-05 23:46:34 +02:00
fate-filter-amix-first: CMD = ffmpeg -auto_conversion_filters -filter_complex amix=duration=first -ss 4 -max_size 4096 -i $(SRC) -max_size 4096 -i $(SRC1) -f f32le -
fate: Split audio filters into their own separate file
2013-04-16 17:38:02 +03:00
fate-filter-amix-first: REF = $(SAMPLES)/filter/amix_first.pcm
FATE_AMIX += fate-filter-amix-transition
fate-filter-amix-transition: tests/data/asynth-44100-2-3.wav
fate-filter-amix-transition: SRC2 = $(TARGET_PATH)/tests/data/asynth-44100-2-3.wav
fate: use a fixed wav demux packet size for amix tests The dropout transition feature of the amix filter depends on the incoming packet size. Signed-off-by: Marton Balint <cus@passwd.hu>
2024-03-05 23:46:34 +02:00
fate-filter-amix-transition: CMD = ffmpeg -auto_conversion_filters -filter_complex amix=inputs=3:dropout_transition=0.5 -max_size 4096 -i $(SRC) -ss 2 -max_size 4096 -i $(SRC1) -ss 4 -max_size 4096 -i $(SRC2) -f f32le -
fate: Split audio filters into their own separate file
2013-04-16 17:38:02 +03:00
fate-filter-amix-transition: REF = $(SAMPLES)/filter/amix_transition.pcm
fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECENCMUX, AMIX ARESAMPLE, WAV, PCM_S16LE, PCM_F32LE, PCM_F32LE) += $(FATE_AMIX)
fate: Split audio filters into their own separate file
2013-04-16 17:38:02 +03:00
$(FATE_AMIX): tests/data/asynth-44100-2.wav tests/data/asynth-44100-2-2.wav
$(FATE_AMIX): SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
$(FATE_AMIX): SRC1 = $(TARGET_PATH)/tests/data/asynth-44100-2-2.wav
$(FATE_AMIX): CMP = oneoff
$(FATE_AMIX): CMP_UNIT = f32
fate: fix sample dependencies for fate-{a,v}filter tests
2016-04-10 23:31:05 +02:00
FATE_AFILTER_SAMPLES-$(CONFIG_ARESAMPLE_FILTER) += fate-filter-aresample
Fix TARGET_SAMPLES and TARGET_PATH for some fate tests. This allows running more fate tests on remote targets without the source tree.
2014-03-18 03:29:50 +03:00
fate-filter-aresample: SRC = $(TARGET_SAMPLES)/nellymoser/nellymoser-discont.flv
avformat_find_stream_info: fallback to timestamps in analyzeduration check in more cases This fixes speex in rtmp Fixes Ticket2409 the nellymoser in flv case actually needs larger analyzeduration. The code previously just failed to calculate the duration If this causes any problems, like premature analyze/probe end, please report! Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-07-08 03:19:28 +03:00
fate-filter-aresample: CMD = pcm -analyzeduration 10000000 -i $(SRC) -af aresample=min_comp=0.001:min_hard_comp=0.1:first_pts=0
Merge commit 'b357f1d1d3dd1d6672df6a397a31be425d31b6d1' * commit 'b357f1d1d3dd1d6672df6a397a31be425d31b6d1': fate: Split audio filters into their own separate file Conflicts: tests/Makefile tests/fate/filter.mak Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-04-20 00:11:44 +03:00
fate-filter-aresample: CMP = oneoff
fate-filter-aresample: REF = $(SAMPLES)/nellymoser/nellymoser-discont.pcm
fate: Split audio filters into their own separate file
2013-04-16 17:38:02 +03:00
lavfi: add trim and atrim filters.
2013-04-02 20:30:11 +03:00
FATE_ATRIM += fate-filter-atrim-duration
fate-filter-atrim-duration: CMD = framecrc -i $(SRC) -af atrim=start=0.1:duration=0.01
FATE_ATRIM += fate-filter-atrim-mixed
fate-filter-atrim-mixed: CMD = framecrc -i $(SRC) -af atrim=start=0.05:start_sample=1025:end=0.1:end_sample=4411
FATE_ATRIM += fate-filter-atrim-samples
fate-filter-atrim-samples: CMD = framecrc -i $(SRC) -af atrim=start_sample=26:end_sample=80
FATE_ATRIM += fate-filter-atrim-time
fate-filter-atrim-time: CMD = framecrc -i $(SRC) -af atrim=0.1:0.2
$(FATE_ATRIM): tests/data/asynth-44100-2.wav
$(FATE_ATRIM): SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate: fix sample dependencies for fate-{a,v}filter tests
2016-04-10 23:31:05 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, ATRIM, WAV, PCM_S16LE, PCM_S16LE, WAV) += $(FATE_ATRIM)
lavfi: add trim and atrim filters.
2013-04-02 20:30:11 +03:00
tests/fate: add asegment filter tests
2023-12-04 17:06:02 +02:00
FATE_ASEGMENT += fate-filter-asegment-samples-absolute
fate-filter-asegment-samples-absolute: CMD = framecrc -i $(SRC) -lavfi asegment=samples="10000|20000"
FATE_ASEGMENT += fate-filter-asegment-samples-relative
fate-filter-asegment-samples-relative: CMD = framecrc -i $(SRC) -lavfi asegment=samples="10000|+10000"
FATE_ASEGMENT += fate-filter-asegment-timestamps-absolute
fate-filter-asegment-timestamps-absolute: CMD = framecrc -i $(SRC) -lavfi asegment=timestamps="1|3"
FATE_ASEGMENT += fate-filter-asegment-timestamps-relative
fate-filter-asegment-timestamps-relative: CMD = framecrc -i $(SRC) -lavfi asegment=timestamps="1|+2"
$(FATE_ASEGMENT): tests/data/asynth-44100-2.wav
$(FATE_ASEGMENT): SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
FATE_AFILTER-$(call FILTERDEMDECENCMUX, ASEGMENT, WAV, PCM_S16LE, PCM_S16LE, WAV) += $(FATE_ASEGMENT)
FATE: add a test for the ONE_STR mapping mode of the channelmap filter
2014-03-04 09:31:47 +03:00
FATE_FILTER_CHANNELMAP += fate-filter-channelmap-one-int
fate-filter-channelmap-one-int: tests/data/filtergraphs/channelmap_one_int
fate-filter-channelmap-one-int: SRC = $(TARGET_PATH)/tests/data/asynth-44100-6.wav
fate-filter-channelmap-one-int: tests/data/asynth-44100-6.wav
fftools/ffmpeg: deprecate -filter_complex_script It is equivalent to -/filter_complex.
2024-01-12 16:54:11 +02:00
fate-filter-channelmap-one-int: CMD = md5 -auto_conversion_filters -i $(SRC) -/filter_complex $(TARGET_PATH)/tests/data/filtergraphs/channelmap_one_int -f wav -fflags +bitexact
FATE: add a test for the ONE_STR mapping mode of the channelmap filter
2014-03-04 09:31:47 +03:00
fate-filter-channelmap-one-int: CMP = oneline
fate: use do_md5sum instead of the md5 protocol for most md5 fate tests The md5 protocol has no seek support, but some tests use seeks. This changes the fate tests to actually create the output files and calculate the md5 on the written files, which also makes the tests independent of the size of the output buffers and output buffering in general. A new md5pipe fate test method is also introduced to keep the old functionality for tests where using a non-seekable output was intentional, and matroska md5 tests are changed to use that. Signed-off-by: Marton Balint <cus@passwd.hu>
2017-06-16 22:08:02 +02:00
fate-filter-channelmap-one-int: REF = 8cfe553d65ed4696756d8c1b824fcdd3
FATE: add a test for the ONE_STR mapping mode of the channelmap filter
2014-03-04 09:31:47 +03:00
FATE_FILTER_CHANNELMAP += fate-filter-channelmap-one-str
fate-filter-channelmap-one-str: tests/data/filtergraphs/channelmap_one_str
fate-filter-channelmap-one-str: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-channelmap-one-str: tests/data/asynth-44100-2.wav
fftools/ffmpeg: deprecate -filter_complex_script It is equivalent to -/filter_complex.
2024-01-12 16:54:11 +02:00
fate-filter-channelmap-one-str: CMD = md5 -auto_conversion_filters -i $(SRC) -/filter_complex $(TARGET_PATH)/tests/data/filtergraphs/channelmap_one_str -f wav -fflags +bitexact
FATE: add a test for the ONE_STR mapping mode of the channelmap filter
2014-03-04 09:31:47 +03:00
fate-filter-channelmap-one-str: CMP = oneline
avfilter/af_channelmap: fix mapping if in_channel was a string but out_channel was not specified In this case in_channel_idx was never set and the default 0 was used. Suprisingly no one noticed that the respective fate test output was wrong. Signed-off-by: Marton Balint <cus@passwd.hu>
2024-03-18 23:08:54 +02:00
fate-filter-channelmap-one-str: REF = e18791f65ce5861e130b2c3e472ab90a
FATE: add a test for the ONE_STR mapping mode of the channelmap filter
2014-03-04 09:31:47 +03:00
fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, CHANNELMAP ARESAMPLE, WAV, PCM_S16LE, PCM_S16LE, WAV) += $(FATE_FILTER_CHANNELMAP)
fate: Split audio filters into their own separate file
2013-04-16 17:38:02 +03:00
fate/filter-audio: Don't use pcm output for channelsplit test This test muxes two streams into a single pcm file, although the two streams are of course not recoverable from the output (unless one has extra information). So use the streamhash muxer instead (which also provides coverage for it; it was surprisingly unused in FATE so far). This is in preparation for actually enforcing a limit of one stream for the PCM muxers. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2024-03-19 22:08:28 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, CHANNELSPLIT ASETNSAMPLES ARESAMPLE, WAV, PCM_S16LE, PCM_S16LE, STREAMHASH) += fate-filter-channelsplit
fate: Split audio filters into their own separate file
2013-04-16 17:38:02 +03:00
fate-filter-channelsplit: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-channelsplit: tests/data/asynth-44100-2.wav
fate/filter-audio: Don't use pcm output for channelsplit test This test muxes two streams into a single pcm file, although the two streams are of course not recoverable from the output (unless one has extra information). So use the streamhash muxer instead (which also provides coverage for it; it was surprisingly unused in FATE so far). This is in preparation for actually enforcing a limit of one stream for the PCM muxers. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2024-03-19 22:08:28 +02:00
fate-filter-channelsplit: CMD = fmtstdout streamhash -auto_conversion_filters -i $(SRC) -filter_complex asetnsamples=n=1024:p=0,channelsplit
fate: Split audio filters into their own separate file
2013-04-16 17:38:02 +03:00
fate/filter-audio: add missing aresample filter dependency Signed-off-by: James Almer <jamrial@gmail.com>
2024-03-15 00:30:12 +02:00
FATE_AFILTER-$(call FILTERDEMDECENCMUX, JOIN ARESAMPLE, WAV, PCM_S16LE, PCM_S16LE, PCM_S16LE) += fate-filter-join
fate: Split audio filters into their own separate file
2013-04-16 17:38:02 +03:00
fate-filter-join: SRC1 = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-join: SRC2 = $(TARGET_PATH)/tests/data/asynth-44100-3.wav
fate-filter-join: tests/data/asynth-44100-2.wav tests/data/asynth-44100-3.wav
fate: disable automatic conversions on many tests. Explicitly insert the scale or aresample filter where it would have been inserted by the negotiation. Re-enable conversions if it cannot be done easily. If a conversion is needed in a test, we want to know about it. If the negotiation changes and makes new conversion necessary, we want to know about it even more.
2020-08-14 18:58:27 +02:00
fate-filter-join: CMD = md5 -auto_conversion_filters -i $(SRC1) -i $(SRC2) -filter_complex join=channel_layout=5c -f s16le
fate: Split audio filters into their own separate file
2013-04-16 17:38:02 +03:00
fate-filter-join: CMP = oneline
fate: re-enable af_join test Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-04-22 14:13:03 +03:00
fate-filter-join: REF = 88b0d24a64717ba8635b29e8dac6ecd8
fate: Split audio filters into their own separate file
2013-04-16 17:38:02 +03:00
Merge commit 'b357f1d1d3dd1d6672df6a397a31be425d31b6d1' * commit 'b357f1d1d3dd1d6672df6a397a31be425d31b6d1': fate: Split audio filters into their own separate file Conflicts: tests/Makefile tests/fate/filter.mak Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-04-20 00:11:44 +03:00
FATE_AFILTER-$(call ALLYES, WAV_DEMUXER PCM_S16LE_DECODER PCM_S16LE_ENCODER PCM_S16LE_MUXER APERMS_FILTER VOLUME_FILTER) += fate-filter-volume
fate: Split audio filters into their own separate file
2013-04-16 17:38:02 +03:00
fate-filter-volume: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
fate-filter-volume: tests/data/asynth-44100-2.wav
Merge commit 'b357f1d1d3dd1d6672df6a397a31be425d31b6d1' * commit 'b357f1d1d3dd1d6672df6a397a31be425d31b6d1': fate: Split audio filters into their own separate file Conflicts: tests/Makefile tests/fate/filter.mak Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-04-20 00:11:44 +03:00
fate-filter-volume: CMD = md5 -i $(SRC) -af aperms=random,volume=precision=fixed:volume=0.5 -f s16le
fate: Split audio filters into their own separate file
2013-04-16 17:38:02 +03:00
fate-filter-volume: CMP = oneline
fate-filter-volume: REF = 4d6ba75ef3e32d305d066b9bc771d6f4
af_hdcd: more FATE tests Additional/Modified FATE tests improve code coverage from 63.7% to 98.1%. Changed fate-suite sample files: * filter/hdcd-mix.flac (958K) added. It is a much better test than filter/hdcd.flac (910K), which is now unused, but can't be removed. * filter/hdcd-fake20bit.flac (168K) added. It is the first second of filter/hdcd.flac, with the 16-bit LSB copied into bit 20 of a 24-bit stream. There isn't an actual non-16-bit HDCD sample available to test. Signed-off-by: Burt P <pburt0@gmail.com>
2016-12-18 08:20:48 +02:00
# hdcd-mix.flac is a mix of three different sources which are interesting for various reasons:
# first 5 seconds uses packet format A and max LLE of -7.0db
# second 5 seconds uses packet format B and has a gain mismatch between channels
# last 10 seconds is not HDCD but has a coincidental HDCD packet, it needs to be 10 seconds because it also tests the cdt expiration
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECENCMUX, HDCD, FLAC, FLAC, PCM_S24LE, PCM_S24LE) += fate-filter-hdcd-mix
fate-filter-hdcd-mix: SRC = $(TARGET_SAMPLES)/filter/hdcd-mix.flac
fate-filter-hdcd-mix: CMD = md5 -i $(SRC) -af hdcd -f s24le
fate-filter-hdcd-mix: CMP = oneline
avcodec/flac_parser: Fix off-by-one error The flac parser uses a fifo to buffer its data. Consequently, when searching for sync codes of flac packets, one needs to take care of the possibility of wraparound. This is done by using an optimized start code search that works on each of the continuous buffers separately and by explicitly checking whether the last pre-wrap byte and the first post-wrap byte constitute a valid sync code. Moreover, the last MAX_FRAME_HEADER_SIZE - 1 bytes ought not to be searched for (the start of) a sync code because a header that might be found in this region might not be completely available. These bytes ought to be searched lateron when more data is available or when flushing. Unfortunately there was an off-by-one error in the calculation of the length to search of the post-wrap buffer: It was too large, because the calculation was based on the amount of bytes available in the fifo from the last pre-wrap byte onwards. This meant that a header might be parsed twice (once prematurely and once regularly when more data is available); it could also mean that an invalid header will be treated as valid (namely if the length of said invalid header is MAX_FRAME_HEADER_SIZE and the invalid byte that will be treated as the last byte of this potential header happens to be the right CRC-8). Should a header be parsed twice, the second instance will be the best child of the first instance; the first instance's score will be FLAC_HEADER_BASE_SCORE - FLAC_HEADER_CHANGED_PENALTY ( = 3) higher than the second instance's score. So the frame belonging to the first instance will be output and it will be done as a zero length frame (the difference of the header's offset and the child's offset). This has serious consequences when flushing, as returning a zero length buffer signals to the caller that no more data will be output; consequently the last frames not yet output will be dropped. Furthermore, a "sample/frame number mismatch in adjacent frames" warning got output when returning the zero-length frame belonging to the first header, because the child's sample/frame number of course didn't match the expected sample frame/number given its parent. filter/hdcd-mix.flac from the FATE-suite was affected by this (the last frame was omitted) which is the reason why several FATE-tests needed to be updated. Fixes ticket #5937. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2019-10-06 07:01:15 +02:00
fate-filter-hdcd-mix: REF = 77443573e0bd3532de52a8bc0e825da7
af_hdcd: more FATE tests Additional/Modified FATE tests improve code coverage from 63.7% to 98.1%. Changed fate-suite sample files: * filter/hdcd-mix.flac (958K) added. It is a much better test than filter/hdcd.flac (910K), which is now unused, but can't be removed. * filter/hdcd-fake20bit.flac (168K) added. It is the first second of filter/hdcd.flac, with the 16-bit LSB copied into bit 20 of a 24-bit stream. There isn't an actual non-16-bit HDCD sample available to test. Signed-off-by: Burt P <pburt0@gmail.com>
2016-12-18 08:20:48 +02:00
# output will be different because of the gain mismatch in the second and third parts
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECENCMUX, HDCD, FLAC, FLAC, PCM_S24LE, PCM_S24LE) += fate-filter-hdcd-mix-psoff
fate-filter-hdcd-mix-psoff: SRC = $(TARGET_SAMPLES)/filter/hdcd-mix.flac
fate-filter-hdcd-mix-psoff: CMD = md5 -i $(SRC) -af hdcd=process_stereo=false -f s24le
fate-filter-hdcd-mix-psoff: CMP = oneline
avcodec/flac_parser: Fix off-by-one error The flac parser uses a fifo to buffer its data. Consequently, when searching for sync codes of flac packets, one needs to take care of the possibility of wraparound. This is done by using an optimized start code search that works on each of the continuous buffers separately and by explicitly checking whether the last pre-wrap byte and the first post-wrap byte constitute a valid sync code. Moreover, the last MAX_FRAME_HEADER_SIZE - 1 bytes ought not to be searched for (the start of) a sync code because a header that might be found in this region might not be completely available. These bytes ought to be searched lateron when more data is available or when flushing. Unfortunately there was an off-by-one error in the calculation of the length to search of the post-wrap buffer: It was too large, because the calculation was based on the amount of bytes available in the fifo from the last pre-wrap byte onwards. This meant that a header might be parsed twice (once prematurely and once regularly when more data is available); it could also mean that an invalid header will be treated as valid (namely if the length of said invalid header is MAX_FRAME_HEADER_SIZE and the invalid byte that will be treated as the last byte of this potential header happens to be the right CRC-8). Should a header be parsed twice, the second instance will be the best child of the first instance; the first instance's score will be FLAC_HEADER_BASE_SCORE - FLAC_HEADER_CHANGED_PENALTY ( = 3) higher than the second instance's score. So the frame belonging to the first instance will be output and it will be done as a zero length frame (the difference of the header's offset and the child's offset). This has serious consequences when flushing, as returning a zero length buffer signals to the caller that no more data will be output; consequently the last frames not yet output will be dropped. Furthermore, a "sample/frame number mismatch in adjacent frames" warning got output when returning the zero-length frame belonging to the first header, because the child's sample/frame number of course didn't match the expected sample frame/number given its parent. filter/hdcd-mix.flac from the FATE-suite was affected by this (the last frame was omitted) which is the reason why several FATE-tests needed to be updated. Fixes ticket #5937. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2019-10-06 07:01:15 +02:00
fate-filter-hdcd-mix-psoff: REF = 89e57885917a436b30855db4d478cefb
af_hdcd: more FATE tests Additional/Modified FATE tests improve code coverage from 63.7% to 98.1%. Changed fate-suite sample files: * filter/hdcd-mix.flac (958K) added. It is a much better test than filter/hdcd.flac (910K), which is now unused, but can't be removed. * filter/hdcd-fake20bit.flac (168K) added. It is the first second of filter/hdcd.flac, with the 16-bit LSB copied into bit 20 of a 24-bit stream. There isn't an actual non-16-bit HDCD sample available to test. Signed-off-by: Burt P <pburt0@gmail.com>
2016-12-18 08:20:48 +02:00
# test the different analyze modes
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECENCMUX, HDCD, FLAC, FLAC, PCM_S24LE, PCM_S24LE) += fate-filter-hdcd-analyze-pe
fate-filter-hdcd-analyze-pe: SRC = $(TARGET_SAMPLES)/filter/hdcd-mix.flac
fate-filter-hdcd-analyze-pe: CMD = md5 -i $(SRC) -af hdcd=analyze_mode=pe -f s24le
fate-filter-hdcd-analyze-pe: CMP = oneline
avcodec/flac_parser: Fix off-by-one error The flac parser uses a fifo to buffer its data. Consequently, when searching for sync codes of flac packets, one needs to take care of the possibility of wraparound. This is done by using an optimized start code search that works on each of the continuous buffers separately and by explicitly checking whether the last pre-wrap byte and the first post-wrap byte constitute a valid sync code. Moreover, the last MAX_FRAME_HEADER_SIZE - 1 bytes ought not to be searched for (the start of) a sync code because a header that might be found in this region might not be completely available. These bytes ought to be searched lateron when more data is available or when flushing. Unfortunately there was an off-by-one error in the calculation of the length to search of the post-wrap buffer: It was too large, because the calculation was based on the amount of bytes available in the fifo from the last pre-wrap byte onwards. This meant that a header might be parsed twice (once prematurely and once regularly when more data is available); it could also mean that an invalid header will be treated as valid (namely if the length of said invalid header is MAX_FRAME_HEADER_SIZE and the invalid byte that will be treated as the last byte of this potential header happens to be the right CRC-8). Should a header be parsed twice, the second instance will be the best child of the first instance; the first instance's score will be FLAC_HEADER_BASE_SCORE - FLAC_HEADER_CHANGED_PENALTY ( = 3) higher than the second instance's score. So the frame belonging to the first instance will be output and it will be done as a zero length frame (the difference of the header's offset and the child's offset). This has serious consequences when flushing, as returning a zero length buffer signals to the caller that no more data will be output; consequently the last frames not yet output will be dropped. Furthermore, a "sample/frame number mismatch in adjacent frames" warning got output when returning the zero-length frame belonging to the first header, because the child's sample/frame number of course didn't match the expected sample frame/number given its parent. filter/hdcd-mix.flac from the FATE-suite was affected by this (the last frame was omitted) which is the reason why several FATE-tests needed to be updated. Fixes ticket #5937. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2019-10-06 07:01:15 +02:00
fate-filter-hdcd-analyze-pe: REF = 2d839d8a1cf73b10a566ce3d4cfaa79e
af_hdcd: more FATE tests Additional/Modified FATE tests improve code coverage from 63.7% to 98.1%. Changed fate-suite sample files: * filter/hdcd-mix.flac (958K) added. It is a much better test than filter/hdcd.flac (910K), which is now unused, but can't be removed. * filter/hdcd-fake20bit.flac (168K) added. It is the first second of filter/hdcd.flac, with the 16-bit LSB copied into bit 20 of a 24-bit stream. There isn't an actual non-16-bit HDCD sample available to test. Signed-off-by: Burt P <pburt0@gmail.com>
2016-12-18 08:20:48 +02:00
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECENCMUX, HDCD, FLAC, FLAC, PCM_S24LE, PCM_S24LE) += fate-filter-hdcd-analyze-lle
fate-filter-hdcd-analyze-lle: SRC = $(TARGET_SAMPLES)/filter/hdcd-mix.flac
fate-filter-hdcd-analyze-lle: CMD = md5 -i $(SRC) -af hdcd=analyze_mode=lle -f s24le
fate-filter-hdcd-analyze-lle: CMP = oneline
avcodec/flac_parser: Fix off-by-one error The flac parser uses a fifo to buffer its data. Consequently, when searching for sync codes of flac packets, one needs to take care of the possibility of wraparound. This is done by using an optimized start code search that works on each of the continuous buffers separately and by explicitly checking whether the last pre-wrap byte and the first post-wrap byte constitute a valid sync code. Moreover, the last MAX_FRAME_HEADER_SIZE - 1 bytes ought not to be searched for (the start of) a sync code because a header that might be found in this region might not be completely available. These bytes ought to be searched lateron when more data is available or when flushing. Unfortunately there was an off-by-one error in the calculation of the length to search of the post-wrap buffer: It was too large, because the calculation was based on the amount of bytes available in the fifo from the last pre-wrap byte onwards. This meant that a header might be parsed twice (once prematurely and once regularly when more data is available); it could also mean that an invalid header will be treated as valid (namely if the length of said invalid header is MAX_FRAME_HEADER_SIZE and the invalid byte that will be treated as the last byte of this potential header happens to be the right CRC-8). Should a header be parsed twice, the second instance will be the best child of the first instance; the first instance's score will be FLAC_HEADER_BASE_SCORE - FLAC_HEADER_CHANGED_PENALTY ( = 3) higher than the second instance's score. So the frame belonging to the first instance will be output and it will be done as a zero length frame (the difference of the header's offset and the child's offset). This has serious consequences when flushing, as returning a zero length buffer signals to the caller that no more data will be output; consequently the last frames not yet output will be dropped. Furthermore, a "sample/frame number mismatch in adjacent frames" warning got output when returning the zero-length frame belonging to the first header, because the child's sample/frame number of course didn't match the expected sample frame/number given its parent. filter/hdcd-mix.flac from the FATE-suite was affected by this (the last frame was omitted) which is the reason why several FATE-tests needed to be updated. Fixes ticket #5937. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2019-10-06 07:01:15 +02:00
fate-filter-hdcd-analyze-lle: REF = b4b185332b7025c191062f49a2c015f1
af_hdcd: more FATE tests Additional/Modified FATE tests improve code coverage from 63.7% to 98.1%. Changed fate-suite sample files: * filter/hdcd-mix.flac (958K) added. It is a much better test than filter/hdcd.flac (910K), which is now unused, but can't be removed. * filter/hdcd-fake20bit.flac (168K) added. It is the first second of filter/hdcd.flac, with the 16-bit LSB copied into bit 20 of a 24-bit stream. There isn't an actual non-16-bit HDCD sample available to test. Signed-off-by: Burt P <pburt0@gmail.com>
2016-12-18 08:20:48 +02:00
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECENCMUX, HDCD, FLAC, FLAC, PCM_S24LE, PCM_S24LE) += fate-filter-hdcd-analyze-cdt
fate-filter-hdcd-analyze-cdt: SRC = $(TARGET_SAMPLES)/filter/hdcd-mix.flac
fate-filter-hdcd-analyze-cdt: CMD = md5 -i $(SRC) -af hdcd=analyze_mode=cdt -f s24le
fate-filter-hdcd-analyze-cdt: CMP = oneline
avcodec/flac_parser: Fix off-by-one error The flac parser uses a fifo to buffer its data. Consequently, when searching for sync codes of flac packets, one needs to take care of the possibility of wraparound. This is done by using an optimized start code search that works on each of the continuous buffers separately and by explicitly checking whether the last pre-wrap byte and the first post-wrap byte constitute a valid sync code. Moreover, the last MAX_FRAME_HEADER_SIZE - 1 bytes ought not to be searched for (the start of) a sync code because a header that might be found in this region might not be completely available. These bytes ought to be searched lateron when more data is available or when flushing. Unfortunately there was an off-by-one error in the calculation of the length to search of the post-wrap buffer: It was too large, because the calculation was based on the amount of bytes available in the fifo from the last pre-wrap byte onwards. This meant that a header might be parsed twice (once prematurely and once regularly when more data is available); it could also mean that an invalid header will be treated as valid (namely if the length of said invalid header is MAX_FRAME_HEADER_SIZE and the invalid byte that will be treated as the last byte of this potential header happens to be the right CRC-8). Should a header be parsed twice, the second instance will be the best child of the first instance; the first instance's score will be FLAC_HEADER_BASE_SCORE - FLAC_HEADER_CHANGED_PENALTY ( = 3) higher than the second instance's score. So the frame belonging to the first instance will be output and it will be done as a zero length frame (the difference of the header's offset and the child's offset). This has serious consequences when flushing, as returning a zero length buffer signals to the caller that no more data will be output; consequently the last frames not yet output will be dropped. Furthermore, a "sample/frame number mismatch in adjacent frames" warning got output when returning the zero-length frame belonging to the first header, because the child's sample/frame number of course didn't match the expected sample frame/number given its parent. filter/hdcd-mix.flac from the FATE-suite was affected by this (the last frame was omitted) which is the reason why several FATE-tests needed to be updated. Fixes ticket #5937. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2019-10-06 07:01:15 +02:00
fate-filter-hdcd-analyze-cdt: REF = afa6577675c63e87da3edbd442b7b6e2
af_hdcd: more FATE tests Additional/Modified FATE tests improve code coverage from 63.7% to 98.1%. Changed fate-suite sample files: * filter/hdcd-mix.flac (958K) added. It is a much better test than filter/hdcd.flac (910K), which is now unused, but can't be removed. * filter/hdcd-fake20bit.flac (168K) added. It is the first second of filter/hdcd.flac, with the 16-bit LSB copied into bit 20 of a 24-bit stream. There isn't an actual non-16-bit HDCD sample available to test. Signed-off-by: Burt P <pburt0@gmail.com>
2016-12-18 08:20:48 +02:00
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECENCMUX, HDCD, FLAC, FLAC, PCM_S24LE, PCM_S24LE) += fate-filter-hdcd-analyze-tgm
fate-filter-hdcd-analyze-tgm: SRC = $(TARGET_SAMPLES)/filter/hdcd-mix.flac
fate-filter-hdcd-analyze-tgm: CMD = md5 -i $(SRC) -af hdcd=analyze_mode=tgm -f s24le
fate-filter-hdcd-analyze-tgm: CMP = oneline
avcodec/flac_parser: Fix off-by-one error The flac parser uses a fifo to buffer its data. Consequently, when searching for sync codes of flac packets, one needs to take care of the possibility of wraparound. This is done by using an optimized start code search that works on each of the continuous buffers separately and by explicitly checking whether the last pre-wrap byte and the first post-wrap byte constitute a valid sync code. Moreover, the last MAX_FRAME_HEADER_SIZE - 1 bytes ought not to be searched for (the start of) a sync code because a header that might be found in this region might not be completely available. These bytes ought to be searched lateron when more data is available or when flushing. Unfortunately there was an off-by-one error in the calculation of the length to search of the post-wrap buffer: It was too large, because the calculation was based on the amount of bytes available in the fifo from the last pre-wrap byte onwards. This meant that a header might be parsed twice (once prematurely and once regularly when more data is available); it could also mean that an invalid header will be treated as valid (namely if the length of said invalid header is MAX_FRAME_HEADER_SIZE and the invalid byte that will be treated as the last byte of this potential header happens to be the right CRC-8). Should a header be parsed twice, the second instance will be the best child of the first instance; the first instance's score will be FLAC_HEADER_BASE_SCORE - FLAC_HEADER_CHANGED_PENALTY ( = 3) higher than the second instance's score. So the frame belonging to the first instance will be output and it will be done as a zero length frame (the difference of the header's offset and the child's offset). This has serious consequences when flushing, as returning a zero length buffer signals to the caller that no more data will be output; consequently the last frames not yet output will be dropped. Furthermore, a "sample/frame number mismatch in adjacent frames" warning got output when returning the zero-length frame belonging to the first header, because the child's sample/frame number of course didn't match the expected sample frame/number given its parent. filter/hdcd-mix.flac from the FATE-suite was affected by this (the last frame was omitted) which is the reason why several FATE-tests needed to be updated. Fixes ticket #5937. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2019-10-06 07:01:15 +02:00
fate-filter-hdcd-analyze-tgm: REF = 285f0fd2249b4903cd5e1ad5ce004219
af_hdcd: more FATE tests Additional/Modified FATE tests improve code coverage from 63.7% to 98.1%. Changed fate-suite sample files: * filter/hdcd-mix.flac (958K) added. It is a much better test than filter/hdcd.flac (910K), which is now unused, but can't be removed. * filter/hdcd-fake20bit.flac (168K) added. It is the first second of filter/hdcd.flac, with the 16-bit LSB copied into bit 20 of a 24-bit stream. There isn't an actual non-16-bit HDCD sample available to test. Signed-off-by: Burt P <pburt0@gmail.com>
2016-12-18 08:20:48 +02:00
# the two additional analyze modes from libhdcd
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECENCMUX, HDCD, FLAC, FLAC, PCM_S24LE, PCM_S24LE) += fate-filter-hdcd-analyze-ltgm
fate-filter-hdcd-analyze-ltgm: SRC = $(TARGET_SAMPLES)/filter/hdcd-mix.flac
fate-filter-hdcd-analyze-ltgm: CMD = md5 -i $(SRC) -af hdcd=analyze_mode=lle:process_stereo=false -f s24le
fate-filter-hdcd-analyze-ltgm: CMP = oneline
avcodec/flac_parser: Fix off-by-one error The flac parser uses a fifo to buffer its data. Consequently, when searching for sync codes of flac packets, one needs to take care of the possibility of wraparound. This is done by using an optimized start code search that works on each of the continuous buffers separately and by explicitly checking whether the last pre-wrap byte and the first post-wrap byte constitute a valid sync code. Moreover, the last MAX_FRAME_HEADER_SIZE - 1 bytes ought not to be searched for (the start of) a sync code because a header that might be found in this region might not be completely available. These bytes ought to be searched lateron when more data is available or when flushing. Unfortunately there was an off-by-one error in the calculation of the length to search of the post-wrap buffer: It was too large, because the calculation was based on the amount of bytes available in the fifo from the last pre-wrap byte onwards. This meant that a header might be parsed twice (once prematurely and once regularly when more data is available); it could also mean that an invalid header will be treated as valid (namely if the length of said invalid header is MAX_FRAME_HEADER_SIZE and the invalid byte that will be treated as the last byte of this potential header happens to be the right CRC-8). Should a header be parsed twice, the second instance will be the best child of the first instance; the first instance's score will be FLAC_HEADER_BASE_SCORE - FLAC_HEADER_CHANGED_PENALTY ( = 3) higher than the second instance's score. So the frame belonging to the first instance will be output and it will be done as a zero length frame (the difference of the header's offset and the child's offset). This has serious consequences when flushing, as returning a zero length buffer signals to the caller that no more data will be output; consequently the last frames not yet output will be dropped. Furthermore, a "sample/frame number mismatch in adjacent frames" warning got output when returning the zero-length frame belonging to the first header, because the child's sample/frame number of course didn't match the expected sample frame/number given its parent. filter/hdcd-mix.flac from the FATE-suite was affected by this (the last frame was omitted) which is the reason why several FATE-tests needed to be updated. Fixes ticket #5937. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2019-10-06 07:01:15 +02:00
fate-filter-hdcd-analyze-ltgm: REF = 404dc2301ea97e9f96c3d6d2ebcfeaa5
af_hdcd: more FATE tests Additional/Modified FATE tests improve code coverage from 63.7% to 98.1%. Changed fate-suite sample files: * filter/hdcd-mix.flac (958K) added. It is a much better test than filter/hdcd.flac (910K), which is now unused, but can't be removed. * filter/hdcd-fake20bit.flac (168K) added. It is the first second of filter/hdcd.flac, with the 16-bit LSB copied into bit 20 of a 24-bit stream. There isn't an actual non-16-bit HDCD sample available to test. Signed-off-by: Burt P <pburt0@gmail.com>
2016-12-18 08:20:48 +02:00
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECENCMUX, HDCD, FLAC, FLAC, PCM_S24LE, PCM_S24LE) += fate-filter-hdcd-analyze-pel
fate-filter-hdcd-analyze-pel: SRC = $(TARGET_SAMPLES)/filter/hdcd-mix.flac
fate-filter-hdcd-analyze-pel: CMD = md5 -i $(SRC) -af hdcd=analyze_mode=pe:force_pe=true -f s24le
fate-filter-hdcd-analyze-pel: CMP = oneline
avcodec/flac_parser: Fix off-by-one error The flac parser uses a fifo to buffer its data. Consequently, when searching for sync codes of flac packets, one needs to take care of the possibility of wraparound. This is done by using an optimized start code search that works on each of the continuous buffers separately and by explicitly checking whether the last pre-wrap byte and the first post-wrap byte constitute a valid sync code. Moreover, the last MAX_FRAME_HEADER_SIZE - 1 bytes ought not to be searched for (the start of) a sync code because a header that might be found in this region might not be completely available. These bytes ought to be searched lateron when more data is available or when flushing. Unfortunately there was an off-by-one error in the calculation of the length to search of the post-wrap buffer: It was too large, because the calculation was based on the amount of bytes available in the fifo from the last pre-wrap byte onwards. This meant that a header might be parsed twice (once prematurely and once regularly when more data is available); it could also mean that an invalid header will be treated as valid (namely if the length of said invalid header is MAX_FRAME_HEADER_SIZE and the invalid byte that will be treated as the last byte of this potential header happens to be the right CRC-8). Should a header be parsed twice, the second instance will be the best child of the first instance; the first instance's score will be FLAC_HEADER_BASE_SCORE - FLAC_HEADER_CHANGED_PENALTY ( = 3) higher than the second instance's score. So the frame belonging to the first instance will be output and it will be done as a zero length frame (the difference of the header's offset and the child's offset). This has serious consequences when flushing, as returning a zero length buffer signals to the caller that no more data will be output; consequently the last frames not yet output will be dropped. Furthermore, a "sample/frame number mismatch in adjacent frames" warning got output when returning the zero-length frame belonging to the first header, because the child's sample/frame number of course didn't match the expected sample frame/number given its parent. filter/hdcd-mix.flac from the FATE-suite was affected by this (the last frame was omitted) which is the reason why several FATE-tests needed to be updated. Fixes ticket #5937. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2019-10-06 07:01:15 +02:00
fate-filter-hdcd-analyze-pel: REF = 9342983208ec1a7f2b3e332ac4dcb723
fate: add test for af_hdcd analyze mode Signed-off-by: Burt P <pburt0@gmail.com>
2016-08-23 00:11:42 +02:00
fate: Add HDCD filter tests for false positive and error detection Signed-off-by: Burt P <pburt0@gmail.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-07-14 19:54:34 +02:00
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECENCMUX, HDCD, FLAC, FLAC, PCM_S24LE, PCM_S24LE) += fate-filter-hdcd-false-positive
fate-filter-hdcd-false-positive: SRC = $(TARGET_SAMPLES)/filter/hdcd-false-positive.flac
fate-filter-hdcd-false-positive: CMD = md5 -i $(SRC) -af hdcd -f s24le
fate-filter-hdcd-false-positive: CMP = grep
fate-filter-hdcd-false-positive: REF = HDCD detected: no
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECENCMUX, HDCD, FLAC, FLAC, PCM_S24LE, PCM_S24LE) += fate-filter-hdcd-detect-errors
fate-filter-hdcd-detect-errors: SRC = $(TARGET_SAMPLES)/filter/hdcd-encoding-errors.flac
fate-filter-hdcd-detect-errors: CMD = md5 -i $(SRC) -af hdcd -f s24le
fate-filter-hdcd-detect-errors: CMP = grep
fate-filter-hdcd-detect-errors: REF = detectable errors: [1-9]
af_hdcd: more FATE tests Additional/Modified FATE tests improve code coverage from 63.7% to 98.1%. Changed fate-suite sample files: * filter/hdcd-mix.flac (958K) added. It is a much better test than filter/hdcd.flac (910K), which is now unused, but can't be removed. * filter/hdcd-fake20bit.flac (168K) added. It is the first second of filter/hdcd.flac, with the 16-bit LSB copied into bit 20 of a 24-bit stream. There isn't an actual non-16-bit HDCD sample available to test. Signed-off-by: Burt P <pburt0@gmail.com>
2016-12-18 08:20:48 +02:00
# 20bit HDCD
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECENCMUX, HDCD, FLAC, FLAC, PCM_S32LE, PCM_S32LE) += fate-filter-hdcd-20bit
fate-filter-hdcd-20bit: SRC = $(TARGET_SAMPLES)/filter/hdcd-fake20bit.flac
fate-filter-hdcd-20bit: CMD = md5 -i $(SRC) -af hdcd=bits_per_sample=20 -f s32le
fate-filter-hdcd-20bit: CMP = oneline
fate-filter-hdcd-20bit: REF = 365ded883a4a92483b15b69babc81390
# non-hdcd tests of different input formats for code coverage
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECENCMUX, HDCD, WAV, PCM_S16LE, PCM_S24LE, PCM_S24LE) += fate-filter-hdcd-mono
fate-filter-hdcd-mono: SRC = $(TARGET_SAMPLES)/audiomatch/tones_44100_mono.wav
fate-filter-hdcd-mono: CMD = md5 -i $(SRC) -af hdcd -f s24le
fate-filter-hdcd-mono: CMP = oneline
fate-filter-hdcd-mono: REF = f51b114b20728e6a463a9491c643d166
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECENCMUX, HDCD, WV, WAVPACK, PCM_S32LE, PCM_S32LE) += fate-filter-hdcd-s16p
fate-filter-hdcd-s16p: SRC = $(TARGET_SAMPLES)/wavpack/lossless/16bit-partial.wv
fate-filter-hdcd-s16p: CMD = md5 -i $(SRC) -af hdcd -f s32le
fate-filter-hdcd-s16p: CMP = oneline
fate-filter-hdcd-s16p: REF = 4e767f436b891ac59810a8b2b1d7e96b
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECENCMUX, HDCD, WV, WAVPACK, PCM_S32LE, PCM_S32LE) += fate-filter-hdcd-s32p
fate-filter-hdcd-s32p: SRC = $(TARGET_SAMPLES)/wavpack/lossless/24bit-partial.wv
fate-filter-hdcd-s32p: CMD = md5 -i $(SRC) -af hdcd -f s32le
fate-filter-hdcd-s32p: CMP = oneline
fate-filter-hdcd-s32p: REF = 0c5513e83eedaa10ab6fac9ddc173cf5
tests/fate/filter-audio.mak: add test for atempo audio filter Signed-off-by: Rajiv Harlalka <rajivharlalka009@gmail.com> Signed-off-by: Anton Khirnov <anton@khirnov.net>
2024-03-21 10:05:29 +02:00
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECENCMUX, ATEMPO, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-atempo
tests: Add a missing dependency for the filter-atempo test Signed-off-by: Martin Storsjö <martin@martin.st>
2024-07-04 10:10:12 +02:00
fate-filter-atempo: tests/data/asynth-44100-1.wav
tests/fate/filter-audio: convert atempo test to oneoff Filter output is not bitexact.
2024-07-04 21:02:25 +02:00
fate-filter-atempo: CMD = pcm -i $(TARGET_PATH)/tests/data/asynth-44100-1.wav -af "atempo=2.0"
fate-filter-atempo: CMP = oneoff
fate-filter-atempo: REF = $(SAMPLES)/filter-reference/atempo.pcm
tests/fate/filter-audio.mak: add test for atempo audio filter Signed-off-by: Rajiv Harlalka <rajivharlalka009@gmail.com> Signed-off-by: Anton Khirnov <anton@khirnov.net>
2024-03-21 10:05:29 +02:00
tests/fate/filter-audio: add a test for handling more than 64 channels The test constructs 66 sine waves of different frequencies and joins them into a 7th order ambisonic layout with extra non-diegetic stereo. The channels are then shifted circularly with channelmap and separated into individual streams for muxing with channelsplit. Note that the last step is necessary because swr is not capable of interleaving planar to packed for more than 64 channels, which would be necessary to mux such a stream as PCM.
2024-09-03 10:19:36 +02:00
fate-filter-crazychannels: tests/data/filtergraphs/crazychannels
fate-filter-crazychannels: CMD = framecrc -auto_conversion_filters -/filter_complex $(TARGET_PATH)/tests/data/filtergraphs/crazychannels
fate/filter-audio: add missing aresample filter dependency to fate-filter-crazychannels Signed-off-by: James Almer <jamrial@gmail.com>
2024-09-29 14:01:21 +02:00
FATE_AFILTER-$(call FILTERFRAMECRC, ARESAMPLE SINE JOIN ATRIM CHANNELMAP CHANNELSPLIT) += fate-filter-crazychannels
tests/fate/filter-audio: add a test for handling more than 64 channels The test constructs 66 sine waves of different frequencies and joins them into a 7th order ambisonic layout with extra non-diegetic stereo. The channels are then shifted circularly with channelmap and separated into individual streams for muxing with channelsplit. Note that the last step is necessary because swr is not capable of interleaving planar to packed for more than 64 channels, which would be necessary to mux such a stream as PCM.
2024-09-03 10:19:36 +02:00
fate: add avfilter/formats-test Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-06-20 14:07:44 +02:00
FATE_AFILTER-yes += fate-filter-formats
Merge commit 'a6a750c7ef240b72ce01e9653343a0ddf247d196' * commit 'a6a750c7ef240b72ce01e9653343a0ddf247d196': tests: Move all test programs to a subdirectory Merged-by: Clément Bœsch <clement@stupeflix.com>
2016-06-22 13:44:29 +02:00
fate-filter-formats: libavfilter/tests/formats$(EXESUF)
tests: Add EXESUF to program calls. Fixes fate in Windows subsystem for Linux.
2019-04-18 03:24:35 +02:00
fate-filter-formats: CMD = run libavfilter/tests/formats$(EXESUF)
fate: add avfilter/formats-test Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-06-20 14:07:44 +02:00
fate: fix sample dependencies for fate-{a,v}filter tests
2016-04-10 23:31:05 +02:00
FATE_SAMPLES_AVCONV += $(FATE_AFILTER_SAMPLES-yes)
FATE_FFMPEG += $(FATE_AFILTER-yes)
fate-afilter: $(FATE_AFILTER-yes) $(FATE_AFILTER_SAMPLES-yes)
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