gstreamer/gst/audioconvert/Makefile.am

52 lines
1.6 KiB
Makefile
Raw Normal View History

plugin_LTLIBRARIES = libgstaudioconvert.la
ORC_SOURCE=gstaudioconvertorc
include $(top_srcdir)/common/orc.mak
libgstaudioconvert_la_SOURCES = \
gstaudioconvert.c \
audioconvert.c \
gstchannelmix.c \
gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now Original commit message from CVS: * gst/audioconvert/Makefile.am: * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index), (check_default), (audio_convert_prepare_context), (audio_convert_clean_context), (audio_convert_convert): * gst/audioconvert/audioconvert.h: * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dithering_get_type), (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init), (gst_audio_convert_init), (gst_audio_convert_set_caps), (gst_audio_convert_set_property), (gst_audio_convert_get_property): * gst/audioconvert/gstaudioconvert.h: * gst/audioconvert/gstaudioquantize.c: (gst_audio_quantize_setup_noise_shaping), (gst_audio_quantize_free_noise_shaping), (gst_audio_quantize_setup_dither), (gst_audio_quantize_free_dither), (gst_audio_quantize_setup_quantize_func), (gst_audio_quantize_setup), (gst_audio_quantize_free): * gst/audioconvert/gstaudioquantize.h: Implement dithering and noise shaping in audioconvert. By default now TPDF dithering (and no noise shaping) will be used when converting from a higher bit depth to 20 bit depth or smaller, otherwise everything will be as it is now. For the last audioconvert in a pipeline it would make sense to use some kind of noise shaping, enabling it by default for all conversions would give undesired results though. Fixes #360246. * tests/check/elements/audioconvert.c: (setup_audioconvert), (GST_START_TEST): Adjust unit test for the new audioconvert.
2007-06-28 20:37:58 +00:00
gstaudioquantize.c \
plugin.c
nodist_libgstaudioconvert_la_SOURCES = $(ORC_NODIST_SOURCES)
libgstaudioconvert_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(ORC_CFLAGS)
libgstaudioconvert_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstaudioconvert_la_LIBADD = \
$(top_builddir)/gst-libs/gst/audio/libgstaudio-@GST_MAJORMINOR@.la \
$(GST_BASE_LIBS) $(GST_LIBS) $(ORC_LIBS) $(LIBM)
libgstaudioconvert_la_LIBTOOLFLAGS = --tag=disable-static
noinst_HEADERS = \
gstaudioconvert.h \
audioconvert.h \
gstchannelmix.h \
gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now Original commit message from CVS: * gst/audioconvert/Makefile.am: * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index), (check_default), (audio_convert_prepare_context), (audio_convert_clean_context), (audio_convert_convert): * gst/audioconvert/audioconvert.h: * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dithering_get_type), (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init), (gst_audio_convert_init), (gst_audio_convert_set_caps), (gst_audio_convert_set_property), (gst_audio_convert_get_property): * gst/audioconvert/gstaudioconvert.h: * gst/audioconvert/gstaudioquantize.c: (gst_audio_quantize_setup_noise_shaping), (gst_audio_quantize_free_noise_shaping), (gst_audio_quantize_setup_dither), (gst_audio_quantize_free_dither), (gst_audio_quantize_setup_quantize_func), (gst_audio_quantize_setup), (gst_audio_quantize_free): * gst/audioconvert/gstaudioquantize.h: Implement dithering and noise shaping in audioconvert. By default now TPDF dithering (and no noise shaping) will be used when converting from a higher bit depth to 20 bit depth or smaller, otherwise everything will be as it is now. For the last audioconvert in a pipeline it would make sense to use some kind of noise shaping, enabling it by default for all conversions would give undesired results though. Fixes #360246. * tests/check/elements/audioconvert.c: (setup_audioconvert), (GST_START_TEST): Adjust unit test for the new audioconvert.
2007-06-28 20:37:58 +00:00
gstaudioquantize.h \
gstfastrandom.h \
plugin.h
configure.ac: Remove idct and resample libs Original commit message from CVS: * configure.ac: Remove idct and resample libs * gst-libs/gst/Makefile.am: same Remove usage of gst_library_load(): * ext/alsa/gstalsaplugin.c: (plugin_init): * ext/libvisual/visual.c: (plugin_init): * ext/ogg/gstogg.c: (plugin_init): * ext/theora/theora.c: (plugin_init): * ext/vorbis/vorbis.c: (plugin_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (plugin_init): * gst/audioscale/gstaudioscale.c: * gst/adder/gstadder.c: (plugin_init): * gst/audioconvert/plugin.c: (plugin_init): * sys/ximage/ximagesink.c: (plugin_init): * sys/xvimage/xvimagesink.c: (plugin_init): * gst/tcp/gsttcpplugin.c: (plugin_init): Link plugins against libraries: * ext/ogg/Makefile.am: * ext/theora/Makefile.am: * ext/vorbis/Makefile.am: * gst/audioconvert/Makefile.am: Create proper libraries: * gst-libs/gst/riff/Makefile.am: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/video/Makefile.am: Move resample library to audioscale plugin directory: * gst-libs/gst/resample/Makefile.am: * gst-libs/gst/resample/README: * gst-libs/gst/resample/dtof.c: * gst-libs/gst/resample/dtos.c: * gst-libs/gst/resample/functable.c: * gst-libs/gst/resample/private.h: * gst-libs/gst/resample/resample.c: * gst-libs/gst/resample/resample.h: * gst-libs/gst/resample/resample.vcproj: * gst-libs/gst/resample/test.c: * gst/audioscale/Makefile.am: * gst/audioscale/README: * gst/audioscale/dtof.c: * gst/audioscale/dtos.c: * gst/audioscale/functable.c: * gst/audioscale/private.h: * gst/audioscale/resample.c: * gst/audioscale/resample.h: * gst/audioscale/test.c: Move tagedit library to gst-libs: * gst-libs/gst/tag/Makefile.am: * gst-libs/gst/tag/gstid3tag.c: * gst-libs/gst/tag/gsttagediting.c: * gst-libs/gst/tag/gsttageditingprivate.h: * gst-libs/gst/tag/gstvorbistag.c: * gst/tags/Makefile.am: * gst/tags/gstid3tag.c: * gst/tags/gstvorbistag.c: Fix for core changes: * gst/sine/gstsinesrc.c: (gst_sinesrc_class_init), (gst_sinesrc_init), (gst_sinesrc_src_fixate), (gst_sinesrc_link), (gst_sinesrc_getrange):
2005-04-25 00:23:06 +00:00
#TESTS = channelmixtest
#noinst_PROGRAMS = channelmixtest
#channelmixtest_CFLAGS = $(GST_CFLAGS)
#channelmixtest_LDADD = libgstaudioconvert.la \
# $(top_builddir)/gst-libs/gst/audio/libgstaudio-@GST_MAJORMINOR@.la \
# $(GST_LIBS)
Android.mk: Makefile.am $(BUILT_SOURCES)
androgenizer \
-:PROJECT libgstaudioconvert -:SHARED libgstaudioconvert \
-:TAGS eng debug \
-:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
-:SOURCES $(libgstaudioconvert_la_SOURCES) \
$(nodist_libgstaudioconvert_la_SOURCES) \
-:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstaudioconvert_la_CFLAGS) \
-:LDFLAGS $(libgstaudioconvert_la_LDFLAGS) \
$(libgstaudioconvert_la_LIBADD) \
-ldl \
-:PASSTHROUGH LOCAL_ARM_MODE:=arm \
LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
> $@