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72b81d5b64
Convert DSD to PCM with the same 96-tap symmetric lowpass filter the libavcodec DSD decoders use. The implementation is copied from libavcodec/dsd.c, minus the LSBF table variants, since AV_SAMPLE_FMT_DSD is defined as MSBF. The libavcodec copy will be removed in future commits as decoders are transitioned to SWR. The conversion is stateful (per-channel filter history), so the audio conversion functions gain a per-channel state argument. DSD silence is the 0x69 bit pattern instead of the 0x80 used for PCM. Only a direct DSD to float conversion function is provided. For format pairs without a direct conversion function (e.g. DSD to s16 with equal rates), swr_init() now falls back from the full_convert shortcut to the generic two-stage path instead of failing, and reports unsupported conversions with a proper error message instead of ENOMEM. Signed-off-by: Kacper Michajłow <kasper93@gmail.com>
31 lines
1002 B
Makefile
31 lines
1002 B
Makefile
NAME = swresample
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DESC = FFmpeg audio resampling library
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FFLIBS = avutil
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HEADERS = swresample.h \
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version.h \
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version_major.h \
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OBJS = audioconvert.o \
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dsd2pcm.o \
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dither.o \
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options.o \
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rematrix.o \
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resample.o \
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resample_dsp.o \
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swresample.o \
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swresample_frame.o \
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version.o \
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OBJS-$(CONFIG_LIBSOXR) += soxr_resample.o
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# Objects duplicated from other libraries for shared builds
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SHLIBOBJS += log2_tab.o reverse.o
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# Windows resource file
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SHLIBOBJS-$(HAVE_GNU_WINDRES) += swresampleres.o
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TESTPROGS = rematrix \
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swresample \
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swresample_resample_realloc \
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