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jack: move plugin to gst-plugins-good
https://bugzilla.gnome.org/show_bug.cgi?id=621929
This commit is contained in:
Tim-Philipp Müller
parent
992c05f840
commit
a197901b82
28 changed files with 19 additions and 3154 deletions
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@ -37,8 +37,6 @@ xvid libxvidcore (http://www.xvid.org/)
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Plugins derived from GPL code are as follows:
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Plugins derived from GPL code are as follows:
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dvdreadsrc libdvdread (http://www.dtek.chalmers.se/groups/dvd/)
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dvdreadsrc libdvdread (http://www.dtek.chalmers.se/groups/dvd/)
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jack libjack (http://jackit.sourceforge.net/)
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Note libjack is LGPL, but plugin is GPL.
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monoscope None (Algorithm by Ralph Loader, Joerg Walter,
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monoscope None (Algorithm by Ralph Loader, Joerg Walter,
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Richard Boulton, and Andy Lo A Foe)
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Richard Boulton, and Andy Lo A Foe)
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rtjpeg None (Erik Walthinsen's algorithm)
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rtjpeg None (Erik Walthinsen's algorithm)
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@ -46,6 +46,7 @@ CRUFT_FILES = \
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$(top_builddir)/common/shave-libtool \
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$(top_builddir)/common/shave-libtool \
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$(top_builddir)/ext/alsaspdif/.libs/*.{so,dll,DLL,dylib} \
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$(top_builddir)/ext/alsaspdif/.libs/*.{so,dll,DLL,dylib} \
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$(top_builddir)/ext/ivorbis/.libs/*.{so,dll,DLL,dylib} \
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$(top_builddir)/ext/ivorbis/.libs/*.{so,dll,DLL,dylib} \
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$(top_builddir)/ext/jack/.libs/*.{so,dll,DLL,dylib} \
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$(top_builddir)/gst/aacparse/.libs/*.{so,dll,DLL,dylib} \
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$(top_builddir)/gst/aacparse/.libs/*.{so,dll,DLL,dylib} \
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$(top_builddir)/gst/amrparse/.libs/*.{so,dll,DLL,dylib} \
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$(top_builddir)/gst/amrparse/.libs/*.{so,dll,DLL,dylib} \
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$(top_builddir)/gst/flacparse/.libs/*.{so,dll,DLL,dylib} \
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$(top_builddir)/gst/flacparse/.libs/*.{so,dll,DLL,dylib} \
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@ -60,6 +61,7 @@ CRUFT_FILES = \
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$(top_builddir)/tests/check/elements/selector \
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$(top_builddir)/tests/check/elements/selector \
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$(top_builddir)/tests/check/elements/valve \
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$(top_builddir)/tests/check/elements/valve \
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$(top_builddir)/tests/check/pipelines/metadata \
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$(top_builddir)/tests/check/pipelines/metadata \
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$(top_builddir)/tests/examples/jack/jack_client \
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$(top_builddir)/tests/examples/switch/switcher \
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$(top_builddir)/tests/examples/switch/switcher \
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$(top_builddir)/tests/icles/output-selector-test \
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$(top_builddir)/tests/icles/output-selector-test \
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$(top_builddir)/tests/icles/test-oss4
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$(top_builddir)/tests/icles/test-oss4
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@ -74,8 +76,10 @@ CRUFT_DIRS = \
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$(top_srcdir)/gst/valve \
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$(top_srcdir)/gst/valve \
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$(top_srcdir)/tests/examples/shapewipe \
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$(top_srcdir)/tests/examples/shapewipe \
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$(top_srcdir)/tests/examples/switch \
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$(top_srcdir)/tests/examples/switch \
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$(top_srcdir)/tests/examples/jack \
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$(top_srcdir)/ext/alsaspdif \
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$(top_srcdir)/ext/alsaspdif \
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$(top_srcdir)/ext/ivorbis \
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$(top_srcdir)/ext/ivorbis \
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$(top_srcdir)/ext/jack \
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$(top_srcdir)/ext/metadata
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$(top_srcdir)/ext/metadata
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include $(top_srcdir)/common/cruft.mak
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include $(top_srcdir)/common/cruft.mak
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11
configure.ac
11
configure.ac
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@ -888,14 +888,6 @@ AG_GST_CHECK_FEATURE(GSM, [GSM library], gsmenc gsmdec, [
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AC_SUBST(GSM_LIBS)
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AC_SUBST(GSM_LIBS)
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])
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])
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dnl *** Jack ***
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translit(dnm, m, l) AM_CONDITIONAL(USE_JACK, true)
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AG_GST_CHECK_FEATURE(JACK, Jack, jack, [
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PKG_CHECK_MODULES(JACK, jack >= 0.99.10, HAVE_JACK="yes", HAVE_JACK="no")
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AC_SUBST(JACK_CFLAGS)
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AC_SUBST(JACK_LIBS)
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])
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dnl *** jp2k ***
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dnl *** jp2k ***
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translit(dnm, m, l) AM_CONDITIONAL(USE_JP2K, true)
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translit(dnm, m, l) AM_CONDITIONAL(USE_JP2K, true)
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AG_GST_CHECK_FEATURE(JP2K, [jp2k], jp2kdec jp2kenc, [
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AG_GST_CHECK_FEATURE(JP2K, [jp2k], jp2kdec jp2kenc, [
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@ -1607,7 +1599,6 @@ AM_CONDITIONAL(USE_FAAD, false)
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AM_CONDITIONAL(USE_FBDEV, false)
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AM_CONDITIONAL(USE_FBDEV, false)
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AM_CONDITIONAL(USE_FLITE, false)
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AM_CONDITIONAL(USE_FLITE, false)
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AM_CONDITIONAL(USE_GSM, false)
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AM_CONDITIONAL(USE_GSM, false)
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AM_CONDITIONAL(USE_JACK, false)
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AM_CONDITIONAL(USE_JP2K, false)
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AM_CONDITIONAL(USE_JP2K, false)
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AM_CONDITIONAL(USE_KATE, false)
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AM_CONDITIONAL(USE_KATE, false)
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AM_CONDITIONAL(USE_TIGER, false)
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AM_CONDITIONAL(USE_TIGER, false)
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@ -1812,7 +1803,6 @@ tests/examples/camerabin2/Makefile
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tests/examples/directfb/Makefile
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tests/examples/directfb/Makefile
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tests/examples/mxf/Makefile
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tests/examples/mxf/Makefile
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tests/examples/scaletempo/Makefile
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tests/examples/scaletempo/Makefile
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tests/examples/jack/Makefile
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tests/icles/Makefile
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tests/icles/Makefile
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ext/amrwbenc/Makefile
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ext/amrwbenc/Makefile
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ext/assrender/Makefile
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ext/assrender/Makefile
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@ -1830,7 +1820,6 @@ ext/faac/Makefile
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ext/faad/Makefile
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ext/faad/Makefile
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ext/flite/Makefile
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ext/flite/Makefile
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ext/gsm/Makefile
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ext/gsm/Makefile
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ext/jack/Makefile
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ext/jp2k/Makefile
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ext/jp2k/Makefile
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ext/kate/Makefile
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ext/kate/Makefile
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ext/ladspa/Makefile
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ext/ladspa/Makefile
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@ -99,8 +99,6 @@ EXTRA_HFILES = \
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$(top_srcdir)/ext/dts/gstdtsdec.h \
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$(top_srcdir)/ext/dts/gstdtsdec.h \
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$(top_srcdir)/ext/faac/gstfaac.h \
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$(top_srcdir)/ext/faac/gstfaac.h \
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$(top_srcdir)/ext/faad/gstfaad.h \
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$(top_srcdir)/ext/faad/gstfaad.h \
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$(top_srcdir)/ext/jack/gstjackaudiosrc.h \
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$(top_srcdir)/ext/jack/gstjackaudiosink.h \
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$(top_srcdir)/ext/kate/gstkateenc.h \
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$(top_srcdir)/ext/kate/gstkateenc.h \
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$(top_srcdir)/ext/kate/gstkatedec.h \
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$(top_srcdir)/ext/kate/gstkatedec.h \
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$(top_srcdir)/ext/kate/gstkateparse.h \
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$(top_srcdir)/ext/kate/gstkateparse.h \
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@ -69,8 +69,6 @@
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<xi:include href="xml/element-freeze.xml" />
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<xi:include href="xml/element-freeze.xml" />
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<xi:include href="xml/element-gaussianblur.xml" />
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<xi:include href="xml/element-gaussianblur.xml" />
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<xi:include href="xml/element-ivfparse.xml" />
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<xi:include href="xml/element-ivfparse.xml" />
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<xi:include href="xml/element-jackaudiosrc.xml" />
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<xi:include href="xml/element-jackaudiosink.xml" />
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<xi:include href="xml/element-jpegparse.xml" />
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<xi:include href="xml/element-jpegparse.xml" />
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<xi:include href="xml/element-kaleidoscope.xml" />
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<xi:include href="xml/element-kaleidoscope.xml" />
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<xi:include href="xml/element-kateenc.xml" />
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<xi:include href="xml/element-kateenc.xml" />
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@ -169,7 +167,6 @@
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<xi:include href="xml/plugin-gsm.xml" />
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<xi:include href="xml/plugin-gsm.xml" />
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<xi:include href="xml/plugin-h264parse.xml" />
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<xi:include href="xml/plugin-h264parse.xml" />
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<xi:include href="xml/plugin-ivfparse.xml" />
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<xi:include href="xml/plugin-ivfparse.xml" />
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<xi:include href="xml/plugin-jack.xml" />
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<xi:include href="xml/plugin-jpegformat.xml" />
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<xi:include href="xml/plugin-jpegformat.xml" />
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<xi:include href="xml/plugin-kate.xml" />
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<xi:include href="xml/plugin-kate.xml" />
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<xi:include href="xml/plugin-ladspa.xml" />
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<xi:include href="xml/plugin-ladspa.xml" />
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@ -806,36 +806,6 @@ GST_TYPE_IVF_PARSE
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gst_ivf_parse_get_type
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gst_ivf_parse_get_type
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</SECTION>
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</SECTION>
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<SECTION>
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<FILE>element-jackaudiosrc</FILE>
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<TITLE>jackaudiosrc</TITLE>
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GstJackAudioSrc
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<SUBSECTION Standard>
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GstJackAudioSrcClass
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GST_JACK_AUDIO_SRC
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GST_JACK_AUDIO_SRC_CLASS
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GST_JACK_AUDIO_SRC_GET_CLASS
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GST_IS_JACK_AUDIO_SRC
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GST_IS_JACK_AUDIO_SRC_CLASS
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GST_TYPE_JACK_AUDIO_SRC
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gst_jack_audio_src_get_type
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</SECTION>
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<SECTION>
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<FILE>element-jackaudiosink</FILE>
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<TITLE>jackaudiosink</TITLE>
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GstJackAudioSink
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<SUBSECTION Standard>
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GstJackAudioSinkClass
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GST_JACK_AUDIO_SINK
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GST_JACK_AUDIO_SINK_CLASS
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GST_JACK_AUDIO_SINK_GET_CLASS
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GST_IS_JACK_AUDIO_SINK
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GST_IS_JACK_AUDIO_SINK_CLASS
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GST_TYPE_JACK_AUDIO_SINK
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gst_jack_audio_sink_get_type
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</SECTION>
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<SECTION>
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<SECTION>
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<FILE>element-jpegparse</FILE>
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<FILE>element-jpegparse</FILE>
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<TITLE>jpegparse</TITLE>
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<TITLE>jpegparse</TITLE>
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@ -1538,36 +1538,6 @@
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<DEFAULT>1</DEFAULT>
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<DEFAULT>1</DEFAULT>
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</ARG>
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</ARG>
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<ARG>
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<NAME>GstJackAudioSink::connect</NAME>
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<TYPE>GstJackConnect</TYPE>
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<RANGE></RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Connect</NICK>
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<BLURB>Specify how the output ports will be connected.</BLURB>
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<DEFAULT>Automatically connect ports to physical ports</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstJackAudioSink::server</NAME>
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<TYPE>gchar*</TYPE>
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<RANGE></RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Server</NICK>
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<BLURB>The Jack server to connect to (NULL = default).</BLURB>
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<DEFAULT>NULL</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstJackAudioSink::client</NAME>
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<TYPE>JackClient*</TYPE>
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<RANGE></RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>JackClient</NICK>
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<BLURB>Handle for jack client.</BLURB>
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<DEFAULT></DEFAULT>
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</ARG>
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<ARG>
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<ARG>
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<NAME>GstDvbSrc::bandwidth</NAME>
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<NAME>GstDvbSrc::bandwidth</NAME>
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<TYPE>GstDvbSrcBandwidth</TYPE>
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<TYPE>GstDvbSrcBandwidth</TYPE>
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@ -21953,36 +21923,6 @@
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<DEFAULT>0</DEFAULT>
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<DEFAULT>0</DEFAULT>
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</ARG>
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</ARG>
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<ARG>
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<NAME>GstJackAudioSrc::connect</NAME>
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<TYPE>GstJackConnect</TYPE>
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<RANGE></RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Connect</NICK>
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<BLURB>Specify how the input ports will be connected.</BLURB>
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<DEFAULT>Automatically connect ports to physical ports</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstJackAudioSrc::server</NAME>
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<TYPE>gchar*</TYPE>
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<RANGE></RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Server</NICK>
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<BLURB>The Jack server to connect to (NULL = default).</BLURB>
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<DEFAULT>NULL</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstJackAudioSrc::client</NAME>
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<TYPE>JackClient*</TYPE>
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<RANGE></RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>JackClient</NICK>
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<BLURB>Handle for jack client.</BLURB>
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<DEFAULT></DEFAULT>
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</ARG>
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<ARG>
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<ARG>
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<NAME>GstDCCPClientSrc::caps</NAME>
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<NAME>GstDCCPClientSrc::caps</NAME>
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<TYPE>GstCaps*</TYPE>
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<TYPE>GstCaps*</TYPE>
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@ -26530,7 +26470,7 @@
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<FLAGS>rw</FLAGS>
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<FLAGS>rw</FLAGS>
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<NICK>physics</NICK>
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<NICK>physics</NICK>
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<BLURB>water density: from 1 to 4.</BLURB>
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<BLURB>water density: from 1 to 4.</BLURB>
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<DEFAULT>7.75038e-304</DEFAULT>
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<DEFAULT>8.09774e-321</DEFAULT>
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</ARG>
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</ARG>
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<ARG>
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<ARG>
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@ -26570,7 +26510,7 @@
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<FLAGS>rw</FLAGS>
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<FLAGS>rw</FLAGS>
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<NICK>splash</NICK>
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<NICK>splash</NICK>
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<BLURB>make a big splash in the center.</BLURB>
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<BLURB>make a big splash in the center.</BLURB>
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<DEFAULT>0</DEFAULT>
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<DEFAULT>2.34994e-310</DEFAULT>
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</ARG>
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</ARG>
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<ARG>
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<ARG>
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@ -26580,7 +26520,7 @@
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<FLAGS>rw</FLAGS>
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<FLAGS>rw</FLAGS>
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<NICK>splash</NICK>
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<NICK>splash</NICK>
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<BLURB>make a big splash in the center.</BLURB>
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<BLURB>make a big splash in the center.</BLURB>
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<DEFAULT>4.77773e-299</DEFAULT>
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<DEFAULT>1.82574e-315</DEFAULT>
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</ARG>
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</ARG>
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<ARG>
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<ARG>
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@ -26610,7 +26550,7 @@
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<FLAGS>rw</FLAGS>
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<FLAGS>rw</FLAGS>
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<NICK>ratiox</NICK>
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<NICK>ratiox</NICK>
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<BLURB>x-ratio.</BLURB>
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<BLURB>x-ratio.</BLURB>
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<DEFAULT>0</DEFAULT>
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<DEFAULT>1.47273e-316</DEFAULT>
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</ARG>
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</ARG>
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<ARG>
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<ARG>
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@ -26620,7 +26560,7 @@
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<FLAGS>rw</FLAGS>
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<FLAGS>rw</FLAGS>
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<NICK>ratioy</NICK>
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<NICK>ratioy</NICK>
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<BLURB>y-ratio.</BLURB>
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<BLURB>y-ratio.</BLURB>
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<DEFAULT>0</DEFAULT>
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<DEFAULT>1.85891e-316</DEFAULT>
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</ARG>
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</ARG>
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<ARG>
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<ARG>
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@ -26630,7 +26570,7 @@
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<FLAGS>rw</FLAGS>
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<FLAGS>rw</FLAGS>
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<NICK>DelayTime</NICK>
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<NICK>DelayTime</NICK>
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<BLURB>the delay time.</BLURB>
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<BLURB>the delay time.</BLURB>
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<DEFAULT>0</DEFAULT>
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<DEFAULT>2.18476e-316</DEFAULT>
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</ARG>
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</ARG>
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<ARG>
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<ARG>
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@ -26660,7 +26600,7 @@
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<FLAGS>rw</FLAGS>
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<FLAGS>rw</FLAGS>
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<NICK>Color</NICK>
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<NICK>Color</NICK>
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<BLURB>the color of the image.</BLURB>
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<BLURB>the color of the image.</BLURB>
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<DEFAULT>0</DEFAULT>
|
<DEFAULT>1.39669e-37</DEFAULT>
|
||||||
</ARG>
|
</ARG>
|
||||||
|
|
||||||
<ARG>
|
<ARG>
|
||||||
|
|
@ -26680,7 +26620,7 @@
|
||||||
<FLAGS>rw</FLAGS>
|
<FLAGS>rw</FLAGS>
|
||||||
<NICK>Color-R</NICK>
|
<NICK>Color-R</NICK>
|
||||||
<BLURB>the color of the image.</BLURB>
|
<BLURB>the color of the image.</BLURB>
|
||||||
<DEFAULT>0</DEFAULT>
|
<DEFAULT>7.30424e-38</DEFAULT>
|
||||||
</ARG>
|
</ARG>
|
||||||
|
|
||||||
<ARG>
|
<ARG>
|
||||||
|
|
@ -27010,7 +26950,7 @@
|
||||||
<FLAGS>rw</FLAGS>
|
<FLAGS>rw</FLAGS>
|
||||||
<NICK>lredscale</NICK>
|
<NICK>lredscale</NICK>
|
||||||
<BLURB>multiplier for downscaling non-edge brightness.</BLURB>
|
<BLURB>multiplier for downscaling non-edge brightness.</BLURB>
|
||||||
<DEFAULT>0</DEFAULT>
|
<DEFAULT>3.40216e-111</DEFAULT>
|
||||||
</ARG>
|
</ARG>
|
||||||
|
|
||||||
<ARG>
|
<ARG>
|
||||||
|
|
@ -27020,7 +26960,7 @@
|
||||||
<FLAGS>rw</FLAGS>
|
<FLAGS>rw</FLAGS>
|
||||||
<NICK>lthresh</NICK>
|
<NICK>lthresh</NICK>
|
||||||
<BLURB>threshold for edge lightening.</BLURB>
|
<BLURB>threshold for edge lightening.</BLURB>
|
||||||
<DEFAULT>0</DEFAULT>
|
<DEFAULT>6.9235e+228</DEFAULT>
|
||||||
</ARG>
|
</ARG>
|
||||||
|
|
||||||
<ARG>
|
<ARG>
|
||||||
|
|
@ -27030,7 +26970,7 @@
|
||||||
<FLAGS>rw</FLAGS>
|
<FLAGS>rw</FLAGS>
|
||||||
<NICK>lupscale</NICK>
|
<NICK>lupscale</NICK>
|
||||||
<BLURB>multiplier for upscaling edge brightness.</BLURB>
|
<BLURB>multiplier for upscaling edge brightness.</BLURB>
|
||||||
<DEFAULT>0</DEFAULT>
|
<DEFAULT>7.54985e-96</DEFAULT>
|
||||||
</ARG>
|
</ARG>
|
||||||
|
|
||||||
<ARG>
|
<ARG>
|
||||||
|
|
@ -27200,7 +27140,7 @@
|
||||||
<FLAGS>rw</FLAGS>
|
<FLAGS>rw</FLAGS>
|
||||||
<NICK>blend</NICK>
|
<NICK>blend</NICK>
|
||||||
<BLURB>blend factor.</BLURB>
|
<BLURB>blend factor.</BLURB>
|
||||||
<DEFAULT>4.74303e-322</DEFAULT>
|
<DEFAULT>4.77773e-299</DEFAULT>
|
||||||
</ARG>
|
</ARG>
|
||||||
|
|
||||||
<ARG>
|
<ARG>
|
||||||
|
|
@ -27390,7 +27330,7 @@
|
||||||
<FLAGS>rw</FLAGS>
|
<FLAGS>rw</FLAGS>
|
||||||
<NICK>HSync</NICK>
|
<NICK>HSync</NICK>
|
||||||
<BLURB>the hsync offset.</BLURB>
|
<BLURB>the hsync offset.</BLURB>
|
||||||
<DEFAULT>0</DEFAULT>
|
<DEFAULT>1.86264e-09</DEFAULT>
|
||||||
</ARG>
|
</ARG>
|
||||||
|
|
||||||
<ARG>
|
<ARG>
|
||||||
|
|
|
||||||
|
|
@ -31,7 +31,6 @@ GObject
|
||||||
GstApExSink
|
GstApExSink
|
||||||
GstNasSink
|
GstNasSink
|
||||||
GstSDLAudioSink
|
GstSDLAudioSink
|
||||||
GstJackAudioSink
|
|
||||||
GstChecksumSink
|
GstChecksumSink
|
||||||
GstDCCPClientSink
|
GstDCCPClientSink
|
||||||
GstDCCPServerSink
|
GstDCCPServerSink
|
||||||
|
|
@ -46,8 +45,6 @@ GObject
|
||||||
GstDTMFSrc
|
GstDTMFSrc
|
||||||
GstDataURISrc
|
GstDataURISrc
|
||||||
GstPushSrc
|
GstPushSrc
|
||||||
GstBaseAudioSrc
|
|
||||||
GstJackAudioSrc
|
|
||||||
GstDCCPClientSrc
|
GstDCCPClientSrc
|
||||||
GstDCCPServerSrc
|
GstDCCPServerSrc
|
||||||
GstDc1394
|
GstDc1394
|
||||||
|
|
@ -490,8 +487,6 @@ GObject
|
||||||
GstRegistry
|
GstRegistry
|
||||||
GstRingBuffer
|
GstRingBuffer
|
||||||
GstAudioSinkRingBuffer
|
GstAudioSinkRingBuffer
|
||||||
GstJackAudioSinkRingBuffer
|
|
||||||
GstJackAudioSrcRingBuffer
|
|
||||||
GstTask
|
GstTask
|
||||||
GstTaskPool
|
GstTaskPool
|
||||||
GstSignalObject
|
GstSignalObject
|
||||||
|
|
|
||||||
|
|
@ -1,43 +0,0 @@
|
||||||
<plugin>
|
|
||||||
<name>jack</name>
|
|
||||||
<description>Jack elements</description>
|
|
||||||
<filename>../../ext/jack/.libs/libgstjack.so</filename>
|
|
||||||
<basename>libgstjack.so</basename>
|
|
||||||
<version>0.10.20.1</version>
|
|
||||||
<license>LGPL</license>
|
|
||||||
<source>gst-plugins-bad</source>
|
|
||||||
<package>GStreamer Bad Plug-ins git</package>
|
|
||||||
<origin>Unknown package origin</origin>
|
|
||||||
<elements>
|
|
||||||
<element>
|
|
||||||
<name>jackaudiosink</name>
|
|
||||||
<longname>Audio Sink (Jack)</longname>
|
|
||||||
<class>Sink/Audio</class>
|
|
||||||
<description>Output to Jack</description>
|
|
||||||
<author>Wim Taymans <wim@fluendo.com></author>
|
|
||||||
<pads>
|
|
||||||
<caps>
|
|
||||||
<name>sink</name>
|
|
||||||
<direction>sink</direction>
|
|
||||||
<presence>always</presence>
|
|
||||||
<details>audio/x-raw-float, endianness=(int){ 1234 }, width=(int)32, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
|
|
||||||
</caps>
|
|
||||||
</pads>
|
|
||||||
</element>
|
|
||||||
<element>
|
|
||||||
<name>jackaudiosrc</name>
|
|
||||||
<longname>Audio Source (Jack)</longname>
|
|
||||||
<class>Source/Audio</class>
|
|
||||||
<description>Input from Jack</description>
|
|
||||||
<author>Tristan Matthews <tristan@sat.qc.ca></author>
|
|
||||||
<pads>
|
|
||||||
<caps>
|
|
||||||
<name>src</name>
|
|
||||||
<direction>source</direction>
|
|
||||||
<presence>always</presence>
|
|
||||||
<details>audio/x-raw-float, endianness=(int){ 1234 }, width=(int)32, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
|
|
||||||
</caps>
|
|
||||||
</pads>
|
|
||||||
</element>
|
|
||||||
</elements>
|
|
||||||
</plugin>
|
|
||||||
|
|
@ -136,12 +136,6 @@ endif
|
||||||
HERMES_DIR=
|
HERMES_DIR=
|
||||||
# endif
|
# endif
|
||||||
|
|
||||||
if USE_JACK
|
|
||||||
JACK_DIR=jack
|
|
||||||
else
|
|
||||||
JACK_DIR=
|
|
||||||
endif
|
|
||||||
|
|
||||||
if USE_JP2K
|
if USE_JP2K
|
||||||
JP2K_DIR = jp2k
|
JP2K_DIR = jp2k
|
||||||
else
|
else
|
||||||
|
|
@ -398,7 +392,6 @@ SUBDIRS=\
|
||||||
$(GSM_DIR) \
|
$(GSM_DIR) \
|
||||||
$(G729_DIR) \
|
$(G729_DIR) \
|
||||||
$(HERMES_DIR) \
|
$(HERMES_DIR) \
|
||||||
$(JACK_DIR) \
|
|
||||||
$(JP2K_DIR) \
|
$(JP2K_DIR) \
|
||||||
$(KATE_DIR) \
|
$(KATE_DIR) \
|
||||||
$(LADSPA_DIR) \
|
$(LADSPA_DIR) \
|
||||||
|
|
@ -453,7 +446,6 @@ DIST_SUBDIRS = \
|
||||||
gsettings \
|
gsettings \
|
||||||
gsm \
|
gsm \
|
||||||
ladspa \
|
ladspa \
|
||||||
jack \
|
|
||||||
jp2k \
|
jp2k \
|
||||||
kate \
|
kate \
|
||||||
libmms \
|
libmms \
|
||||||
|
|
|
||||||
1
ext/jack/.gitignore
vendored
1
ext/jack/.gitignore
vendored
|
|
@ -1 +0,0 @@
|
||||||
*.loT
|
|
||||||
|
|
@ -1,12 +0,0 @@
|
||||||
|
|
||||||
plugin_LTLIBRARIES = libgstjack.la
|
|
||||||
|
|
||||||
libgstjack_la_SOURCES = gstjackutil.c gstjack.c gstjackaudiosrc.c gstjackaudiosink.c gstjackaudioclient.c
|
|
||||||
libgstjack_la_CFLAGS = $(GST_PLUGINS_BAD_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(JACK_CFLAGS)
|
|
||||||
libgstjack_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(JACK_LIBS)
|
|
||||||
libgstjack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
|
|
||||||
libgstjack_la_LIBTOOLFLAGS = --tag=disable-static
|
|
||||||
|
|
||||||
noinst_HEADERS = gstjackutil.h gstjackaudiosrc.h gstjackaudiosink.h gstjackaudioclient.h gstjack.h gstjackringbuffer.h
|
|
||||||
|
|
||||||
EXTRA_DIST = README
|
|
||||||
|
|
@ -1,4 +0,0 @@
|
||||||
to be written, la dee da
|
|
||||||
|
|
||||||
jackit.sf.net
|
|
||||||
|
|
||||||
|
|
@ -1,95 +0,0 @@
|
||||||
/* GStreamer Jack plugins
|
|
||||||
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
|
|
||||||
*
|
|
||||||
* This library is free software; you can redistribute it and/or
|
|
||||||
* modify it under the terms of the GNU Library General Public
|
|
||||||
* License as published by the Free Software Foundation; either
|
|
||||||
* version 2 of the License, or (at your option) any later version.
|
|
||||||
*
|
|
||||||
* This library is distributed in the hope that it will be useful,
|
|
||||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
||||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
||||||
* Library General Public License for more details.
|
|
||||||
*
|
|
||||||
* You should have received a copy of the GNU Library General Public
|
|
||||||
* License along with this library; if not, write to the
|
|
||||||
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
|
||||||
* Boston, MA 02111-1307, USA.
|
|
||||||
*/
|
|
||||||
|
|
||||||
#ifdef HAVE_CONFIG_H
|
|
||||||
#include "config.h"
|
|
||||||
#endif
|
|
||||||
|
|
||||||
#include "gstjackaudiosrc.h"
|
|
||||||
#include "gstjackaudiosink.h"
|
|
||||||
|
|
||||||
GType
|
|
||||||
gst_jack_connect_get_type (void)
|
|
||||||
{
|
|
||||||
static GType jack_connect_type = 0;
|
|
||||||
static const GEnumValue jack_connect[] = {
|
|
||||||
{GST_JACK_CONNECT_NONE,
|
|
||||||
"Don't automatically connect ports to physical ports", "none"},
|
|
||||||
{GST_JACK_CONNECT_AUTO,
|
|
||||||
"Automatically connect ports to physical ports", "auto"},
|
|
||||||
{GST_JACK_CONNECT_AUTO_FORCED,
|
|
||||||
"Automatically connect ports to as many physical ports as possible",
|
|
||||||
"auto-forced"},
|
|
||||||
{0, NULL, NULL},
|
|
||||||
};
|
|
||||||
|
|
||||||
if (!jack_connect_type) {
|
|
||||||
jack_connect_type = g_enum_register_static ("GstJackConnect", jack_connect);
|
|
||||||
}
|
|
||||||
return jack_connect_type;
|
|
||||||
}
|
|
||||||
|
|
||||||
|
|
||||||
static gpointer
|
|
||||||
gst_jack_client_copy (gpointer jclient)
|
|
||||||
{
|
|
||||||
return jclient;
|
|
||||||
}
|
|
||||||
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_jack_client_free (gpointer jclient)
|
|
||||||
{
|
|
||||||
return;
|
|
||||||
}
|
|
||||||
|
|
||||||
|
|
||||||
GType
|
|
||||||
gst_jack_client_get_type (void)
|
|
||||||
{
|
|
||||||
static GType type; /* 0 */
|
|
||||||
|
|
||||||
if (type == 0) {
|
|
||||||
/* hackish, but makes it show up nicely in gst-inspect */
|
|
||||||
type = g_boxed_type_register_static ("JackClient",
|
|
||||||
(GBoxedCopyFunc) gst_jack_client_copy,
|
|
||||||
(GBoxedFreeFunc) gst_jack_client_free);
|
|
||||||
}
|
|
||||||
|
|
||||||
return type;
|
|
||||||
}
|
|
||||||
|
|
||||||
static gboolean
|
|
||||||
plugin_init (GstPlugin * plugin)
|
|
||||||
{
|
|
||||||
if (!gst_element_register (plugin, "jackaudiosrc", GST_RANK_PRIMARY,
|
|
||||||
GST_TYPE_JACK_AUDIO_SRC))
|
|
||||||
return FALSE;
|
|
||||||
if (!gst_element_register (plugin, "jackaudiosink", GST_RANK_PRIMARY,
|
|
||||||
GST_TYPE_JACK_AUDIO_SINK))
|
|
||||||
return FALSE;
|
|
||||||
|
|
||||||
return TRUE;
|
|
||||||
}
|
|
||||||
|
|
||||||
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
|
||||||
GST_VERSION_MINOR,
|
|
||||||
"jack",
|
|
||||||
"Jack elements",
|
|
||||||
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
|
|
||||||
|
|
@ -1,55 +0,0 @@
|
||||||
/* GStreamer
|
|
||||||
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
|
|
||||||
*
|
|
||||||
* gstjack.h:
|
|
||||||
*
|
|
||||||
* This library is free software; you can redistribute it and/or
|
|
||||||
* modify it under the terms of the GNU Library General Public
|
|
||||||
* License as published by the Free Software Foundation; either
|
|
||||||
* version 2 of the License, or (at your option) any later version.
|
|
||||||
*
|
|
||||||
* This library is distributed in the hope that it will be useful,
|
|
||||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
||||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
||||||
* Library General Public License for more details.
|
|
||||||
*
|
|
||||||
* You should have received a copy of the GNU Library General Public
|
|
||||||
* License along with this library; if not, write to the
|
|
||||||
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
|
||||||
* Boston, MA 02111-1307, USA.
|
|
||||||
*/
|
|
||||||
|
|
||||||
#ifndef _GST_JACK_H_
|
|
||||||
#define _GST_JACK_H_
|
|
||||||
|
|
||||||
|
|
||||||
/**
|
|
||||||
* GstJackConnect:
|
|
||||||
* @GST_JACK_CONNECT_NONE: Don't automatically connect to physical ports.
|
|
||||||
* In this mode, the element will accept any number of input channels and will
|
|
||||||
* create (but not connect) an output port for each channel.
|
|
||||||
* @GST_JACK_CONNECT_AUTO: In this mode, the element will try to connect each
|
|
||||||
* output port to a random physical jack input pin. The sink will
|
|
||||||
* expose the number of physical channels on its pad caps.
|
|
||||||
* @GST_JACK_CONNECT_AUTO_FORCED: In this mode, the element will try to connect each
|
|
||||||
* output port to a random physical jack input pin. The element will accept any number
|
|
||||||
* of input channels.
|
|
||||||
*
|
|
||||||
* Specify how the output ports will be connected.
|
|
||||||
*/
|
|
||||||
|
|
||||||
typedef enum {
|
|
||||||
GST_JACK_CONNECT_NONE,
|
|
||||||
GST_JACK_CONNECT_AUTO,
|
|
||||||
GST_JACK_CONNECT_AUTO_FORCED
|
|
||||||
} GstJackConnect;
|
|
||||||
|
|
||||||
typedef jack_default_audio_sample_t sample_t;
|
|
||||||
|
|
||||||
#define GST_TYPE_JACK_CONNECT (gst_jack_connect_get_type())
|
|
||||||
#define GST_TYPE_JACK_CLIENT (gst_jack_client_get_type ())
|
|
||||||
|
|
||||||
GType gst_jack_client_get_type(void);
|
|
||||||
GType gst_jack_connect_get_type(void);
|
|
||||||
|
|
||||||
#endif // _GST_JACK_H_
|
|
||||||
|
|
@ -1,525 +0,0 @@
|
||||||
/* GStreamer
|
|
||||||
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
|
|
||||||
*
|
|
||||||
* gstjackaudioclient.c: jack audio client implementation
|
|
||||||
*
|
|
||||||
* This library is free software; you can redistribute it and/or
|
|
||||||
* modify it under the terms of the GNU Library General Public
|
|
||||||
* License as published by the Free Software Foundation; either
|
|
||||||
* version 2 of the License, or (at your option) any later version.
|
|
||||||
*
|
|
||||||
* This library is distributed in the hope that it will be useful,
|
|
||||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
||||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
||||||
* Library General Public License for more details.
|
|
||||||
*
|
|
||||||
* You should have received a copy of the GNU Library General Public
|
|
||||||
* License along with this library; if not, write to the
|
|
||||||
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
|
||||||
* Boston, MA 02111-1307, USA.
|
|
||||||
*/
|
|
||||||
|
|
||||||
#include <string.h>
|
|
||||||
|
|
||||||
#include "gstjackaudioclient.h"
|
|
||||||
|
|
||||||
GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_client_debug);
|
|
||||||
#define GST_CAT_DEFAULT gst_jack_audio_client_debug
|
|
||||||
|
|
||||||
void
|
|
||||||
gst_jack_audio_client_init (void)
|
|
||||||
{
|
|
||||||
GST_DEBUG_CATEGORY_INIT (gst_jack_audio_client_debug, "jackclient", 0,
|
|
||||||
"jackclient helpers");
|
|
||||||
}
|
|
||||||
|
|
||||||
/* a list of global connections indexed by id and server. */
|
|
||||||
G_LOCK_DEFINE_STATIC (connections_lock);
|
|
||||||
static GList *connections;
|
|
||||||
|
|
||||||
/* the connection to a server */
|
|
||||||
typedef struct
|
|
||||||
{
|
|
||||||
gint refcount;
|
|
||||||
GMutex *lock;
|
|
||||||
GCond *flush_cond;
|
|
||||||
|
|
||||||
/* id/server pair and the connection */
|
|
||||||
gchar *id;
|
|
||||||
gchar *server;
|
|
||||||
jack_client_t *client;
|
|
||||||
|
|
||||||
/* lists of GstJackAudioClients */
|
|
||||||
gint n_clients;
|
|
||||||
GList *src_clients;
|
|
||||||
GList *sink_clients;
|
|
||||||
} GstJackAudioConnection;
|
|
||||||
|
|
||||||
/* an object sharing a jack_client_t connection. */
|
|
||||||
struct _GstJackAudioClient
|
|
||||||
{
|
|
||||||
GstJackAudioConnection *conn;
|
|
||||||
|
|
||||||
GstJackClientType type;
|
|
||||||
gboolean active;
|
|
||||||
gboolean deactivate;
|
|
||||||
|
|
||||||
void (*shutdown) (void *arg);
|
|
||||||
JackProcessCallback process;
|
|
||||||
JackBufferSizeCallback buffer_size;
|
|
||||||
JackSampleRateCallback sample_rate;
|
|
||||||
gpointer user_data;
|
|
||||||
};
|
|
||||||
|
|
||||||
typedef jack_default_audio_sample_t sample_t;
|
|
||||||
|
|
||||||
typedef struct
|
|
||||||
{
|
|
||||||
jack_nframes_t nframes;
|
|
||||||
gpointer user_data;
|
|
||||||
} JackCB;
|
|
||||||
|
|
||||||
static int
|
|
||||||
jack_process_cb (jack_nframes_t nframes, void *arg)
|
|
||||||
{
|
|
||||||
GstJackAudioConnection *conn = (GstJackAudioConnection *) arg;
|
|
||||||
GList *walk;
|
|
||||||
int res = 0;
|
|
||||||
|
|
||||||
g_mutex_lock (conn->lock);
|
|
||||||
/* call sources first, then sinks. Sources will either push data into the
|
|
||||||
* ringbuffer of the sinks, which will then pull the data out of it, or
|
|
||||||
* sinks will pull the data from the sources. */
|
|
||||||
for (walk = conn->src_clients; walk; walk = g_list_next (walk)) {
|
|
||||||
GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
|
|
||||||
|
|
||||||
/* only call active clients */
|
|
||||||
if ((client->active || client->deactivate) && client->process) {
|
|
||||||
res = client->process (nframes, client->user_data);
|
|
||||||
if (client->deactivate) {
|
|
||||||
client->deactivate = FALSE;
|
|
||||||
g_cond_signal (conn->flush_cond);
|
|
||||||
}
|
|
||||||
}
|
|
||||||
}
|
|
||||||
for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) {
|
|
||||||
GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
|
|
||||||
|
|
||||||
/* only call active clients */
|
|
||||||
if ((client->active || client->deactivate) && client->process) {
|
|
||||||
res = client->process (nframes, client->user_data);
|
|
||||||
if (client->deactivate) {
|
|
||||||
client->deactivate = FALSE;
|
|
||||||
g_cond_signal (conn->flush_cond);
|
|
||||||
}
|
|
||||||
}
|
|
||||||
}
|
|
||||||
g_mutex_unlock (conn->lock);
|
|
||||||
|
|
||||||
return res;
|
|
||||||
}
|
|
||||||
|
|
||||||
/* we error out */
|
|
||||||
static int
|
|
||||||
jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
|
|
||||||
{
|
|
||||||
return 0;
|
|
||||||
}
|
|
||||||
|
|
||||||
/* we error out */
|
|
||||||
static int
|
|
||||||
jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
|
|
||||||
{
|
|
||||||
return 0;
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
jack_shutdown_cb (void *arg)
|
|
||||||
{
|
|
||||||
GstJackAudioConnection *conn = (GstJackAudioConnection *) arg;
|
|
||||||
GList *walk;
|
|
||||||
|
|
||||||
GST_DEBUG ("disconnect client %s from server %s", conn->id,
|
|
||||||
GST_STR_NULL (conn->server));
|
|
||||||
|
|
||||||
g_mutex_lock (conn->lock);
|
|
||||||
for (walk = conn->src_clients; walk; walk = g_list_next (walk)) {
|
|
||||||
GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
|
|
||||||
|
|
||||||
if (client->shutdown)
|
|
||||||
client->shutdown (client->user_data);
|
|
||||||
}
|
|
||||||
for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) {
|
|
||||||
GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
|
|
||||||
|
|
||||||
if (client->shutdown)
|
|
||||||
client->shutdown (client->user_data);
|
|
||||||
}
|
|
||||||
g_mutex_unlock (conn->lock);
|
|
||||||
}
|
|
||||||
|
|
||||||
typedef struct
|
|
||||||
{
|
|
||||||
const gchar *id;
|
|
||||||
const gchar *server;
|
|
||||||
} FindData;
|
|
||||||
|
|
||||||
static gint
|
|
||||||
connection_find (GstJackAudioConnection * conn, FindData * data)
|
|
||||||
{
|
|
||||||
/* id's must match */
|
|
||||||
if (strcmp (conn->id, data->id))
|
|
||||||
return 1;
|
|
||||||
|
|
||||||
/* both the same or NULL */
|
|
||||||
if (conn->server == data->server)
|
|
||||||
return 0;
|
|
||||||
|
|
||||||
/* we cannot compare NULL */
|
|
||||||
if (conn->server == NULL || data->server == NULL)
|
|
||||||
return 1;
|
|
||||||
|
|
||||||
if (strcmp (conn->server, data->server))
|
|
||||||
return 1;
|
|
||||||
|
|
||||||
return 0;
|
|
||||||
}
|
|
||||||
|
|
||||||
/* make a connection with @id and @server. Returns NULL on failure with the
|
|
||||||
* status set. */
|
|
||||||
static GstJackAudioConnection *
|
|
||||||
gst_jack_audio_make_connection (const gchar * id, const gchar * server,
|
|
||||||
jack_client_t * jclient, jack_status_t * status)
|
|
||||||
{
|
|
||||||
GstJackAudioConnection *conn;
|
|
||||||
jack_options_t options;
|
|
||||||
gint res;
|
|
||||||
|
|
||||||
*status = 0;
|
|
||||||
|
|
||||||
GST_DEBUG ("new client %s, connecting to server %s", id,
|
|
||||||
GST_STR_NULL (server));
|
|
||||||
|
|
||||||
/* never start a server */
|
|
||||||
options = JackNoStartServer;
|
|
||||||
/* if we have a servername, use it */
|
|
||||||
if (server != NULL)
|
|
||||||
options |= JackServerName;
|
|
||||||
/* open the client */
|
|
||||||
if (jclient == NULL)
|
|
||||||
jclient = jack_client_open (id, options, status, server);
|
|
||||||
if (jclient == NULL)
|
|
||||||
goto could_not_open;
|
|
||||||
|
|
||||||
/* now create object */
|
|
||||||
conn = g_new (GstJackAudioConnection, 1);
|
|
||||||
conn->refcount = 1;
|
|
||||||
conn->lock = g_mutex_new ();
|
|
||||||
conn->flush_cond = g_cond_new ();
|
|
||||||
conn->id = g_strdup (id);
|
|
||||||
conn->server = g_strdup (server);
|
|
||||||
conn->client = jclient;
|
|
||||||
conn->n_clients = 0;
|
|
||||||
conn->src_clients = NULL;
|
|
||||||
conn->sink_clients = NULL;
|
|
||||||
|
|
||||||
/* set our callbacks */
|
|
||||||
jack_set_process_callback (jclient, jack_process_cb, conn);
|
|
||||||
/* these callbacks cause us to error */
|
|
||||||
jack_set_buffer_size_callback (jclient, jack_buffer_size_cb, conn);
|
|
||||||
jack_set_sample_rate_callback (jclient, jack_sample_rate_cb, conn);
|
|
||||||
jack_on_shutdown (jclient, jack_shutdown_cb, conn);
|
|
||||||
|
|
||||||
/* all callbacks are set, activate the client */
|
|
||||||
if ((res = jack_activate (jclient)))
|
|
||||||
goto could_not_activate;
|
|
||||||
|
|
||||||
GST_DEBUG ("opened connection %p", conn);
|
|
||||||
|
|
||||||
return conn;
|
|
||||||
|
|
||||||
/* ERRORS */
|
|
||||||
could_not_open:
|
|
||||||
{
|
|
||||||
GST_DEBUG ("failed to open jack client, %d", *status);
|
|
||||||
return NULL;
|
|
||||||
}
|
|
||||||
could_not_activate:
|
|
||||||
{
|
|
||||||
GST_ERROR ("Could not activate client (%d)", res);
|
|
||||||
*status = JackFailure;
|
|
||||||
g_mutex_free (conn->lock);
|
|
||||||
g_free (conn->id);
|
|
||||||
g_free (conn->server);
|
|
||||||
g_free (conn);
|
|
||||||
return NULL;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
static GstJackAudioConnection *
|
|
||||||
gst_jack_audio_get_connection (const gchar * id, const gchar * server,
|
|
||||||
jack_client_t * jclient, jack_status_t * status)
|
|
||||||
{
|
|
||||||
GstJackAudioConnection *conn;
|
|
||||||
GList *found;
|
|
||||||
FindData data;
|
|
||||||
|
|
||||||
GST_DEBUG ("getting connection for id %s, server %s", id,
|
|
||||||
GST_STR_NULL (server));
|
|
||||||
|
|
||||||
data.id = id;
|
|
||||||
data.server = server;
|
|
||||||
|
|
||||||
G_LOCK (connections_lock);
|
|
||||||
found =
|
|
||||||
g_list_find_custom (connections, &data, (GCompareFunc) connection_find);
|
|
||||||
if (found != NULL && jclient != NULL) {
|
|
||||||
/* we found it, increase refcount and return it */
|
|
||||||
conn = (GstJackAudioConnection *) found->data;
|
|
||||||
conn->refcount++;
|
|
||||||
|
|
||||||
GST_DEBUG ("found connection %p", conn);
|
|
||||||
} else {
|
|
||||||
/* make new connection */
|
|
||||||
conn = gst_jack_audio_make_connection (id, server, jclient, status);
|
|
||||||
if (conn != NULL) {
|
|
||||||
GST_DEBUG ("created connection %p", conn);
|
|
||||||
/* add to list on success */
|
|
||||||
connections = g_list_prepend (connections, conn);
|
|
||||||
} else {
|
|
||||||
GST_WARNING ("could not create connection");
|
|
||||||
}
|
|
||||||
}
|
|
||||||
G_UNLOCK (connections_lock);
|
|
||||||
|
|
||||||
return conn;
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_jack_audio_unref_connection (GstJackAudioConnection * conn)
|
|
||||||
{
|
|
||||||
gint res;
|
|
||||||
gboolean zero;
|
|
||||||
|
|
||||||
GST_DEBUG ("unref connection %p refcnt %d", conn, conn->refcount);
|
|
||||||
|
|
||||||
G_LOCK (connections_lock);
|
|
||||||
conn->refcount--;
|
|
||||||
if ((zero = (conn->refcount == 0))) {
|
|
||||||
GST_DEBUG ("closing connection %p", conn);
|
|
||||||
/* remove from list, we can release the mutex after removing the connection
|
|
||||||
* from the list because after that, nobody can access the connection anymore. */
|
|
||||||
connections = g_list_remove (connections, conn);
|
|
||||||
}
|
|
||||||
G_UNLOCK (connections_lock);
|
|
||||||
|
|
||||||
/* if we are zero, close and cleanup the connection */
|
|
||||||
if (zero) {
|
|
||||||
/* don't use conn->lock here. two reasons:
|
|
||||||
*
|
|
||||||
* 1) its not necessary: jack_deactivate() will not return until the JACK thread
|
|
||||||
* associated with this connection is cleaned up by a thread join, hence
|
|
||||||
* no more callbacks can occur or be in progress.
|
|
||||||
*
|
|
||||||
* 2) it would deadlock anyway, because jack_deactivate() will sleep
|
|
||||||
* waiting for the JACK thread, and can thus cause deadlock in
|
|
||||||
* jack_process_cb()
|
|
||||||
*/
|
|
||||||
if ((res = jack_deactivate (conn->client))) {
|
|
||||||
/* we only warn, this means the server is probably shut down and the client
|
|
||||||
* is gone anyway. */
|
|
||||||
GST_WARNING ("Could not deactivate Jack client (%d)", res);
|
|
||||||
}
|
|
||||||
/* close connection */
|
|
||||||
if ((res = jack_client_close (conn->client))) {
|
|
||||||
/* we assume the client is gone. */
|
|
||||||
GST_WARNING ("close failed (%d)", res);
|
|
||||||
}
|
|
||||||
|
|
||||||
/* free resources */
|
|
||||||
g_mutex_free (conn->lock);
|
|
||||||
g_cond_free (conn->flush_cond);
|
|
||||||
g_free (conn->id);
|
|
||||||
g_free (conn->server);
|
|
||||||
g_free (conn);
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_jack_audio_connection_add_client (GstJackAudioConnection * conn,
|
|
||||||
GstJackAudioClient * client)
|
|
||||||
{
|
|
||||||
g_mutex_lock (conn->lock);
|
|
||||||
switch (client->type) {
|
|
||||||
case GST_JACK_CLIENT_SOURCE:
|
|
||||||
conn->src_clients = g_list_append (conn->src_clients, client);
|
|
||||||
conn->n_clients++;
|
|
||||||
break;
|
|
||||||
case GST_JACK_CLIENT_SINK:
|
|
||||||
conn->sink_clients = g_list_append (conn->sink_clients, client);
|
|
||||||
conn->n_clients++;
|
|
||||||
break;
|
|
||||||
default:
|
|
||||||
g_warning ("trying to add unknown client type");
|
|
||||||
break;
|
|
||||||
}
|
|
||||||
g_mutex_unlock (conn->lock);
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_jack_audio_connection_remove_client (GstJackAudioConnection * conn,
|
|
||||||
GstJackAudioClient * client)
|
|
||||||
{
|
|
||||||
g_mutex_lock (conn->lock);
|
|
||||||
switch (client->type) {
|
|
||||||
case GST_JACK_CLIENT_SOURCE:
|
|
||||||
conn->src_clients = g_list_remove (conn->src_clients, client);
|
|
||||||
conn->n_clients--;
|
|
||||||
break;
|
|
||||||
case GST_JACK_CLIENT_SINK:
|
|
||||||
conn->sink_clients = g_list_remove (conn->sink_clients, client);
|
|
||||||
conn->n_clients--;
|
|
||||||
break;
|
|
||||||
default:
|
|
||||||
g_warning ("trying to remove unknown client type");
|
|
||||||
break;
|
|
||||||
}
|
|
||||||
g_mutex_unlock (conn->lock);
|
|
||||||
}
|
|
||||||
|
|
||||||
/**
|
|
||||||
* gst_jack_audio_client_get:
|
|
||||||
* @id: the client id
|
|
||||||
* @server: the server to connect to or NULL for the default server
|
|
||||||
* @type: the client type
|
|
||||||
* @shutdown: a callback when the jack server shuts down
|
|
||||||
* @process: a callback when samples are available
|
|
||||||
* @buffer_size: a callback when the buffer_size changes
|
|
||||||
* @sample_rate: a callback when the sample_rate changes
|
|
||||||
* @user_data: user data passed to the callbacks
|
|
||||||
* @status: pointer to hold the jack status code in case of errors
|
|
||||||
*
|
|
||||||
* Get the jack client connection for @id and @server. Connections to the same
|
|
||||||
* @id and @server will receive the same physical Jack client connection and
|
|
||||||
* will therefore be scheduled in the same process callback.
|
|
||||||
*
|
|
||||||
* Returns: a #GstJackAudioClient.
|
|
||||||
*/
|
|
||||||
GstJackAudioClient *
|
|
||||||
gst_jack_audio_client_new (const gchar * id, const gchar * server,
|
|
||||||
jack_client_t * jclient, GstJackClientType type,
|
|
||||||
void (*shutdown) (void *arg), JackProcessCallback process,
|
|
||||||
JackBufferSizeCallback buffer_size, JackSampleRateCallback sample_rate,
|
|
||||||
gpointer user_data, jack_status_t * status)
|
|
||||||
{
|
|
||||||
GstJackAudioClient *client;
|
|
||||||
GstJackAudioConnection *conn;
|
|
||||||
|
|
||||||
g_return_val_if_fail (id != NULL, NULL);
|
|
||||||
g_return_val_if_fail (status != NULL, NULL);
|
|
||||||
|
|
||||||
/* first get a connection for the id/server pair */
|
|
||||||
conn = gst_jack_audio_get_connection (id, server, jclient, status);
|
|
||||||
if (conn == NULL)
|
|
||||||
goto no_connection;
|
|
||||||
|
|
||||||
GST_INFO ("new client %s", id);
|
|
||||||
|
|
||||||
/* make new client using the connection */
|
|
||||||
client = g_new (GstJackAudioClient, 1);
|
|
||||||
client->active = client->deactivate = FALSE;
|
|
||||||
client->conn = conn;
|
|
||||||
client->type = type;
|
|
||||||
client->shutdown = shutdown;
|
|
||||||
client->process = process;
|
|
||||||
client->buffer_size = buffer_size;
|
|
||||||
client->sample_rate = sample_rate;
|
|
||||||
client->user_data = user_data;
|
|
||||||
|
|
||||||
/* add the client to the connection */
|
|
||||||
gst_jack_audio_connection_add_client (conn, client);
|
|
||||||
|
|
||||||
return client;
|
|
||||||
|
|
||||||
/* ERRORS */
|
|
||||||
no_connection:
|
|
||||||
{
|
|
||||||
GST_DEBUG ("Could not get server connection (%d)", *status);
|
|
||||||
return NULL;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
/**
|
|
||||||
* gst_jack_audio_client_free:
|
|
||||||
* @client: a #GstJackAudioClient
|
|
||||||
*
|
|
||||||
* Free the resources used by @client.
|
|
||||||
*/
|
|
||||||
void
|
|
||||||
gst_jack_audio_client_free (GstJackAudioClient * client)
|
|
||||||
{
|
|
||||||
GstJackAudioConnection *conn;
|
|
||||||
|
|
||||||
g_return_if_fail (client != NULL);
|
|
||||||
|
|
||||||
GST_INFO ("free client");
|
|
||||||
|
|
||||||
conn = client->conn;
|
|
||||||
|
|
||||||
/* remove from connection first so that it's not scheduled anymore after this
|
|
||||||
* call */
|
|
||||||
gst_jack_audio_connection_remove_client (conn, client);
|
|
||||||
gst_jack_audio_unref_connection (conn);
|
|
||||||
|
|
||||||
g_free (client);
|
|
||||||
}
|
|
||||||
|
|
||||||
/**
|
|
||||||
* gst_jack_audio_client_get_client:
|
|
||||||
* @client: a #GstJackAudioClient
|
|
||||||
*
|
|
||||||
* Get the jack audio client for @client. This function is used to perform
|
|
||||||
* operations on the jack server from this client.
|
|
||||||
*
|
|
||||||
* Returns: The jack audio client.
|
|
||||||
*/
|
|
||||||
jack_client_t *
|
|
||||||
gst_jack_audio_client_get_client (GstJackAudioClient * client)
|
|
||||||
{
|
|
||||||
g_return_val_if_fail (client != NULL, NULL);
|
|
||||||
|
|
||||||
/* no lock needed, the connection and the client does not change
|
|
||||||
* once the client is created. */
|
|
||||||
return client->conn->client;
|
|
||||||
}
|
|
||||||
|
|
||||||
/**
|
|
||||||
* gst_jack_audio_client_set_active:
|
|
||||||
* @client: a #GstJackAudioClient
|
|
||||||
* @active: new mode for the client
|
|
||||||
*
|
|
||||||
* Activate or deactive @client. When a client is activated it will receive
|
|
||||||
* callbacks when data should be processed.
|
|
||||||
*
|
|
||||||
* Returns: 0 if all ok.
|
|
||||||
*/
|
|
||||||
gint
|
|
||||||
gst_jack_audio_client_set_active (GstJackAudioClient * client, gboolean active)
|
|
||||||
{
|
|
||||||
g_return_val_if_fail (client != NULL, -1);
|
|
||||||
|
|
||||||
/* make sure that we are not dispatching the client */
|
|
||||||
g_mutex_lock (client->conn->lock);
|
|
||||||
if (client->active && !active) {
|
|
||||||
/* we need to process once more to flush the port */
|
|
||||||
client->deactivate = TRUE;
|
|
||||||
|
|
||||||
/* need to wait for process_cb run once more */
|
|
||||||
while (client->deactivate)
|
|
||||||
g_cond_wait (client->conn->flush_cond, client->conn->lock);
|
|
||||||
}
|
|
||||||
client->active = active;
|
|
||||||
g_mutex_unlock (client->conn->lock);
|
|
||||||
|
|
||||||
return 0;
|
|
||||||
}
|
|
||||||
|
|
@ -1,59 +0,0 @@
|
||||||
/* GStreamer
|
|
||||||
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
|
|
||||||
*
|
|
||||||
* gstjackaudioclient.h:
|
|
||||||
*
|
|
||||||
* This library is free software; you can redistribute it and/or
|
|
||||||
* modify it under the terms of the GNU Library General Public
|
|
||||||
* License as published by the Free Software Foundation; either
|
|
||||||
* version 2 of the License, or (at your option) any later version.
|
|
||||||
*
|
|
||||||
* This library is distributed in the hope that it will be useful,
|
|
||||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
||||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
||||||
* Library General Public License for more details.
|
|
||||||
*
|
|
||||||
* You should have received a copy of the GNU Library General Public
|
|
||||||
* License along with this library; if not, write to the
|
|
||||||
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
|
||||||
* Boston, MA 02111-1307, USA.
|
|
||||||
*/
|
|
||||||
|
|
||||||
#ifndef __GST_JACK_AUDIO_CLIENT_H__
|
|
||||||
#define __GST_JACK_AUDIO_CLIENT_H__
|
|
||||||
|
|
||||||
#include <jack/jack.h>
|
|
||||||
|
|
||||||
#include <gst/gst.h>
|
|
||||||
|
|
||||||
G_BEGIN_DECLS
|
|
||||||
|
|
||||||
typedef enum
|
|
||||||
{
|
|
||||||
GST_JACK_CLIENT_SOURCE,
|
|
||||||
GST_JACK_CLIENT_SINK
|
|
||||||
} GstJackClientType;
|
|
||||||
|
|
||||||
typedef struct _GstJackAudioClient GstJackAudioClient;
|
|
||||||
|
|
||||||
void gst_jack_audio_client_init (void);
|
|
||||||
|
|
||||||
|
|
||||||
GstJackAudioClient * gst_jack_audio_client_new (const gchar *id, const gchar *server,
|
|
||||||
jack_client_t *jclient,
|
|
||||||
GstJackClientType type,
|
|
||||||
void (*shutdown) (void *arg),
|
|
||||||
JackProcessCallback process,
|
|
||||||
JackBufferSizeCallback buffer_size,
|
|
||||||
JackSampleRateCallback sample_rate,
|
|
||||||
gpointer user_data,
|
|
||||||
jack_status_t *status);
|
|
||||||
void gst_jack_audio_client_free (GstJackAudioClient *client);
|
|
||||||
|
|
||||||
jack_client_t * gst_jack_audio_client_get_client (GstJackAudioClient *client);
|
|
||||||
|
|
||||||
gboolean gst_jack_audio_client_set_active (GstJackAudioClient *client, gboolean active);
|
|
||||||
|
|
||||||
G_END_DECLS
|
|
||||||
|
|
||||||
#endif /* __GST_JACK_AUDIO_CLIENT_H__ */
|
|
||||||
|
|
@ -1,852 +0,0 @@
|
||||||
/* GStreamer
|
|
||||||
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
|
|
||||||
*
|
|
||||||
* gstjackaudiosink.c: jack audio sink implementation
|
|
||||||
*
|
|
||||||
* This library is free software; you can redistribute it and/or
|
|
||||||
* modify it under the terms of the GNU Library General Public
|
|
||||||
* License as published by the Free Software Foundation; either
|
|
||||||
* version 2 of the License, or (at your option) any later version.
|
|
||||||
*
|
|
||||||
* This library is distributed in the hope that it will be useful,
|
|
||||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
||||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
||||||
* Library General Public License for more details.
|
|
||||||
*
|
|
||||||
* You should have received a copy of the GNU Library General Public
|
|
||||||
* License along with this library; if not, write to the
|
|
||||||
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
|
||||||
* Boston, MA 02111-1307, USA.
|
|
||||||
*/
|
|
||||||
|
|
||||||
/**
|
|
||||||
* SECTION:element-jackaudiosink
|
|
||||||
* @see_also: #GstBaseAudioSink, #GstRingBuffer
|
|
||||||
*
|
|
||||||
* A Sink that outputs data to Jack ports.
|
|
||||||
*
|
|
||||||
* It will create N Jack ports named out_<name>_<num> where
|
|
||||||
* <name> is the element name and <num> is starting from 1.
|
|
||||||
* Each port corresponds to a gstreamer channel.
|
|
||||||
*
|
|
||||||
* The samplerate as exposed on the caps is always the same as the samplerate of
|
|
||||||
* the jack server.
|
|
||||||
*
|
|
||||||
* When the #GstJackAudioSink:connect property is set to auto, this element
|
|
||||||
* will try to connect each output port to a random physical jack input pin. In
|
|
||||||
* this mode, the sink will expose the number of physical channels on its pad
|
|
||||||
* caps.
|
|
||||||
*
|
|
||||||
* When the #GstJackAudioSink:connect property is set to none, the element will
|
|
||||||
* accept any number of input channels and will create (but not connect) an
|
|
||||||
* output port for each channel.
|
|
||||||
*
|
|
||||||
* The element will generate an error when the Jack server is shut down when it
|
|
||||||
* was PAUSED or PLAYING. This element does not support dynamic rate and buffer
|
|
||||||
* size changes at runtime.
|
|
||||||
*
|
|
||||||
* <refsect2>
|
|
||||||
* <title>Example launch line</title>
|
|
||||||
* |[
|
|
||||||
* gst-launch audiotestsrc ! jackaudiosink
|
|
||||||
* ]| Play a sine wave to using jack.
|
|
||||||
* </refsect2>
|
|
||||||
*
|
|
||||||
* Last reviewed on 2006-11-30 (0.10.4)
|
|
||||||
*/
|
|
||||||
|
|
||||||
#ifdef HAVE_CONFIG_H
|
|
||||||
#include "config.h"
|
|
||||||
#endif
|
|
||||||
|
|
||||||
#include <gst/gst-i18n-plugin.h>
|
|
||||||
#include <stdlib.h>
|
|
||||||
#include <string.h>
|
|
||||||
|
|
||||||
#include "gstjackaudiosink.h"
|
|
||||||
#include "gstjackringbuffer.h"
|
|
||||||
|
|
||||||
GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_sink_debug);
|
|
||||||
#define GST_CAT_DEFAULT gst_jack_audio_sink_debug
|
|
||||||
|
|
||||||
static gboolean
|
|
||||||
gst_jack_audio_sink_allocate_channels (GstJackAudioSink * sink, gint channels)
|
|
||||||
{
|
|
||||||
jack_client_t *client;
|
|
||||||
|
|
||||||
client = gst_jack_audio_client_get_client (sink->client);
|
|
||||||
|
|
||||||
/* remove ports we don't need */
|
|
||||||
while (sink->port_count > channels) {
|
|
||||||
jack_port_unregister (client, sink->ports[--sink->port_count]);
|
|
||||||
}
|
|
||||||
|
|
||||||
/* alloc enough output ports */
|
|
||||||
sink->ports = g_realloc (sink->ports, sizeof (jack_port_t *) * channels);
|
|
||||||
|
|
||||||
/* create an output port for each channel */
|
|
||||||
while (sink->port_count < channels) {
|
|
||||||
gchar *name;
|
|
||||||
|
|
||||||
/* port names start from 1 and are local to the element */
|
|
||||||
name =
|
|
||||||
g_strdup_printf ("out_%s_%d", GST_ELEMENT_NAME (sink),
|
|
||||||
sink->port_count + 1);
|
|
||||||
sink->ports[sink->port_count] =
|
|
||||||
jack_port_register (client, name, JACK_DEFAULT_AUDIO_TYPE,
|
|
||||||
JackPortIsOutput, 0);
|
|
||||||
if (sink->ports[sink->port_count] == NULL)
|
|
||||||
return FALSE;
|
|
||||||
|
|
||||||
sink->port_count++;
|
|
||||||
|
|
||||||
g_free (name);
|
|
||||||
}
|
|
||||||
return TRUE;
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_jack_audio_sink_free_channels (GstJackAudioSink * sink)
|
|
||||||
{
|
|
||||||
gint res, i = 0;
|
|
||||||
jack_client_t *client;
|
|
||||||
|
|
||||||
client = gst_jack_audio_client_get_client (sink->client);
|
|
||||||
|
|
||||||
/* get rid of all ports */
|
|
||||||
while (sink->port_count) {
|
|
||||||
GST_LOG_OBJECT (sink, "unregister port %d", i);
|
|
||||||
if ((res = jack_port_unregister (client, sink->ports[i++]))) {
|
|
||||||
GST_DEBUG_OBJECT (sink, "unregister of port failed (%d)", res);
|
|
||||||
}
|
|
||||||
sink->port_count--;
|
|
||||||
}
|
|
||||||
g_free (sink->ports);
|
|
||||||
sink->ports = NULL;
|
|
||||||
}
|
|
||||||
|
|
||||||
/* ringbuffer abstract base class */
|
|
||||||
static GType
|
|
||||||
gst_jack_ring_buffer_get_type (void)
|
|
||||||
{
|
|
||||||
static GType ringbuffer_type = 0;
|
|
||||||
|
|
||||||
if (!ringbuffer_type) {
|
|
||||||
static const GTypeInfo ringbuffer_info = {
|
|
||||||
sizeof (GstJackRingBufferClass),
|
|
||||||
NULL,
|
|
||||||
NULL,
|
|
||||||
(GClassInitFunc) gst_jack_ring_buffer_class_init,
|
|
||||||
NULL,
|
|
||||||
NULL,
|
|
||||||
sizeof (GstJackRingBuffer),
|
|
||||||
0,
|
|
||||||
(GInstanceInitFunc) gst_jack_ring_buffer_init,
|
|
||||||
NULL
|
|
||||||
};
|
|
||||||
|
|
||||||
ringbuffer_type =
|
|
||||||
g_type_register_static (GST_TYPE_RING_BUFFER,
|
|
||||||
"GstJackAudioSinkRingBuffer", &ringbuffer_info, 0);
|
|
||||||
}
|
|
||||||
return ringbuffer_type;
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass)
|
|
||||||
{
|
|
||||||
GObjectClass *gobject_class;
|
|
||||||
GstObjectClass *gstobject_class;
|
|
||||||
GstRingBufferClass *gstringbuffer_class;
|
|
||||||
|
|
||||||
gobject_class = (GObjectClass *) klass;
|
|
||||||
gstobject_class = (GstObjectClass *) klass;
|
|
||||||
gstringbuffer_class = (GstRingBufferClass *) klass;
|
|
||||||
|
|
||||||
ring_parent_class = g_type_class_peek_parent (klass);
|
|
||||||
|
|
||||||
gstringbuffer_class->open_device =
|
|
||||||
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_open_device);
|
|
||||||
gstringbuffer_class->close_device =
|
|
||||||
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_close_device);
|
|
||||||
gstringbuffer_class->acquire =
|
|
||||||
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_acquire);
|
|
||||||
gstringbuffer_class->release =
|
|
||||||
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_release);
|
|
||||||
gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
|
|
||||||
gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_pause);
|
|
||||||
gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
|
|
||||||
gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_stop);
|
|
||||||
|
|
||||||
gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_delay);
|
|
||||||
}
|
|
||||||
|
|
||||||
/* this is the callback of jack. This should RT-safe.
|
|
||||||
*/
|
|
||||||
static int
|
|
||||||
jack_process_cb (jack_nframes_t nframes, void *arg)
|
|
||||||
{
|
|
||||||
GstJackAudioSink *sink;
|
|
||||||
GstRingBuffer *buf;
|
|
||||||
GstJackRingBuffer *abuf;
|
|
||||||
gint readseg, len;
|
|
||||||
guint8 *readptr;
|
|
||||||
gint i, j, flen, channels;
|
|
||||||
sample_t **buffers, *data;
|
|
||||||
|
|
||||||
buf = GST_RING_BUFFER_CAST (arg);
|
|
||||||
abuf = GST_JACK_RING_BUFFER_CAST (arg);
|
|
||||||
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
|
|
||||||
|
|
||||||
channels = buf->spec.channels;
|
|
||||||
|
|
||||||
/* alloc pointers to samples */
|
|
||||||
buffers = g_alloca (sizeof (sample_t *) * channels);
|
|
||||||
|
|
||||||
/* get target buffers */
|
|
||||||
for (i = 0; i < channels; i++) {
|
|
||||||
buffers[i] = (sample_t *) jack_port_get_buffer (sink->ports[i], nframes);
|
|
||||||
}
|
|
||||||
|
|
||||||
if (gst_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
|
|
||||||
flen = len / channels;
|
|
||||||
|
|
||||||
/* the number of samples must be exactly the segment size */
|
|
||||||
if (nframes * sizeof (sample_t) != flen)
|
|
||||||
goto wrong_size;
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (sink, "copy %d frames: %p, %d bytes, %d channels",
|
|
||||||
nframes, readptr, flen, channels);
|
|
||||||
data = (sample_t *) readptr;
|
|
||||||
|
|
||||||
/* the samples in the ringbuffer have the channels interleaved, we need to
|
|
||||||
* deinterleave into the jack target buffers */
|
|
||||||
for (i = 0; i < nframes; i++) {
|
|
||||||
for (j = 0; j < channels; j++) {
|
|
||||||
buffers[j][i] = *data++;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
/* clear written samples in the ringbuffer */
|
|
||||||
gst_ring_buffer_clear (buf, readseg);
|
|
||||||
|
|
||||||
/* we wrote one segment */
|
|
||||||
gst_ring_buffer_advance (buf, 1);
|
|
||||||
} else {
|
|
||||||
GST_DEBUG_OBJECT (sink, "write %d frames silence", nframes);
|
|
||||||
/* We are not allowed to read from the ringbuffer, write silence to all
|
|
||||||
* jack output buffers */
|
|
||||||
for (i = 0; i < channels; i++) {
|
|
||||||
memset (buffers[i], 0, nframes * sizeof (sample_t));
|
|
||||||
}
|
|
||||||
}
|
|
||||||
return 0;
|
|
||||||
|
|
||||||
/* ERRORS */
|
|
||||||
wrong_size:
|
|
||||||
{
|
|
||||||
GST_ERROR_OBJECT (sink, "nbytes (%d) != flen (%d)",
|
|
||||||
(gint) (nframes * sizeof (sample_t)), flen);
|
|
||||||
return 1;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
/* we error out */
|
|
||||||
static int
|
|
||||||
jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
|
|
||||||
{
|
|
||||||
GstJackAudioSink *sink;
|
|
||||||
GstJackRingBuffer *abuf;
|
|
||||||
|
|
||||||
abuf = GST_JACK_RING_BUFFER_CAST (arg);
|
|
||||||
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
|
|
||||||
|
|
||||||
if (abuf->sample_rate != -1 && abuf->sample_rate != nframes)
|
|
||||||
goto not_supported;
|
|
||||||
|
|
||||||
return 0;
|
|
||||||
|
|
||||||
/* ERRORS */
|
|
||||||
not_supported:
|
|
||||||
{
|
|
||||||
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS,
|
|
||||||
(NULL), ("Jack changed the sample rate, which is not supported"));
|
|
||||||
return 1;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
/* we error out */
|
|
||||||
static int
|
|
||||||
jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
|
|
||||||
{
|
|
||||||
GstJackAudioSink *sink;
|
|
||||||
GstJackRingBuffer *abuf;
|
|
||||||
|
|
||||||
abuf = GST_JACK_RING_BUFFER_CAST (arg);
|
|
||||||
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
|
|
||||||
|
|
||||||
if (abuf->buffer_size != -1 && abuf->buffer_size != nframes)
|
|
||||||
goto not_supported;
|
|
||||||
|
|
||||||
return 0;
|
|
||||||
|
|
||||||
/* ERRORS */
|
|
||||||
not_supported:
|
|
||||||
{
|
|
||||||
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS,
|
|
||||||
(NULL), ("Jack changed the buffer size, which is not supported"));
|
|
||||||
return 1;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
jack_shutdown_cb (void *arg)
|
|
||||||
{
|
|
||||||
GstJackAudioSink *sink;
|
|
||||||
|
|
||||||
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (sink, "shutdown");
|
|
||||||
|
|
||||||
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
|
|
||||||
(NULL), ("Jack server shutdown"));
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_jack_ring_buffer_init (GstJackRingBuffer * buf,
|
|
||||||
GstJackRingBufferClass * g_class)
|
|
||||||
{
|
|
||||||
buf->channels = -1;
|
|
||||||
buf->buffer_size = -1;
|
|
||||||
buf->sample_rate = -1;
|
|
||||||
}
|
|
||||||
|
|
||||||
/* the _open_device method should make a connection with the server
|
|
||||||
*/
|
|
||||||
static gboolean
|
|
||||||
gst_jack_ring_buffer_open_device (GstRingBuffer * buf)
|
|
||||||
{
|
|
||||||
GstJackAudioSink *sink;
|
|
||||||
jack_status_t status = 0;
|
|
||||||
const gchar *name;
|
|
||||||
|
|
||||||
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (sink, "open");
|
|
||||||
|
|
||||||
name = g_get_application_name ();
|
|
||||||
if (!name)
|
|
||||||
name = "GStreamer";
|
|
||||||
|
|
||||||
sink->client = gst_jack_audio_client_new (name, sink->server,
|
|
||||||
sink->jclient,
|
|
||||||
GST_JACK_CLIENT_SINK,
|
|
||||||
jack_shutdown_cb,
|
|
||||||
jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status);
|
|
||||||
if (sink->client == NULL)
|
|
||||||
goto could_not_open;
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (sink, "opened");
|
|
||||||
|
|
||||||
return TRUE;
|
|
||||||
|
|
||||||
/* ERRORS */
|
|
||||||
could_not_open:
|
|
||||||
{
|
|
||||||
if (status & JackServerFailed) {
|
|
||||||
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
|
|
||||||
(_("Jack server not found")),
|
|
||||||
("Cannot connect to the Jack server (status %d)", status));
|
|
||||||
} else {
|
|
||||||
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
|
|
||||||
(NULL), ("Jack client open error (status %d)", status));
|
|
||||||
}
|
|
||||||
return FALSE;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
/* close the connection with the server
|
|
||||||
*/
|
|
||||||
static gboolean
|
|
||||||
gst_jack_ring_buffer_close_device (GstRingBuffer * buf)
|
|
||||||
{
|
|
||||||
GstJackAudioSink *sink;
|
|
||||||
|
|
||||||
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (sink, "close");
|
|
||||||
|
|
||||||
gst_jack_audio_sink_free_channels (sink);
|
|
||||||
gst_jack_audio_client_free (sink->client);
|
|
||||||
sink->client = NULL;
|
|
||||||
|
|
||||||
return TRUE;
|
|
||||||
}
|
|
||||||
|
|
||||||
/* allocate a buffer and setup resources to process the audio samples of
|
|
||||||
* the format as specified in @spec.
|
|
||||||
*
|
|
||||||
* We allocate N jack ports, one for each channel. If we are asked to
|
|
||||||
* automatically make a connection with physical ports, we connect as many
|
|
||||||
* ports as there are physical ports, leaving leftover ports unconnected.
|
|
||||||
*
|
|
||||||
* It is assumed that samplerate and number of channels are acceptable since our
|
|
||||||
* getcaps method will always provide correct values. If unacceptable caps are
|
|
||||||
* received for some reason, we fail here.
|
|
||||||
*/
|
|
||||||
static gboolean
|
|
||||||
gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
|
|
||||||
{
|
|
||||||
GstJackAudioSink *sink;
|
|
||||||
GstJackRingBuffer *abuf;
|
|
||||||
const char **ports;
|
|
||||||
gint sample_rate, buffer_size;
|
|
||||||
gint i, channels, res;
|
|
||||||
jack_client_t *client;
|
|
||||||
|
|
||||||
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
|
|
||||||
abuf = GST_JACK_RING_BUFFER_CAST (buf);
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (sink, "acquire");
|
|
||||||
|
|
||||||
client = gst_jack_audio_client_get_client (sink->client);
|
|
||||||
|
|
||||||
/* sample rate must be that of the server */
|
|
||||||
sample_rate = jack_get_sample_rate (client);
|
|
||||||
if (sample_rate != spec->rate)
|
|
||||||
goto wrong_samplerate;
|
|
||||||
|
|
||||||
channels = spec->channels;
|
|
||||||
|
|
||||||
if (!gst_jack_audio_sink_allocate_channels (sink, channels))
|
|
||||||
goto out_of_ports;
|
|
||||||
|
|
||||||
buffer_size = jack_get_buffer_size (client);
|
|
||||||
|
|
||||||
/* the segment size in bytes, this is large enough to hold a buffer of 32bit floats
|
|
||||||
* for all channels */
|
|
||||||
spec->segsize = buffer_size * sizeof (gfloat) * channels;
|
|
||||||
spec->latency_time = gst_util_uint64_scale (spec->segsize,
|
|
||||||
(GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample);
|
|
||||||
/* segtotal based on buffer-time latency */
|
|
||||||
spec->segtotal = spec->buffer_time / spec->latency_time;
|
|
||||||
if (spec->segtotal < 2) {
|
|
||||||
spec->segtotal = 2;
|
|
||||||
spec->buffer_time = spec->latency_time * spec->segtotal;
|
|
||||||
}
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (sink, "buffer time: %" G_GINT64_FORMAT " usec",
|
|
||||||
spec->buffer_time);
|
|
||||||
GST_DEBUG_OBJECT (sink, "latency time: %" G_GINT64_FORMAT " usec",
|
|
||||||
spec->latency_time);
|
|
||||||
GST_DEBUG_OBJECT (sink, "buffer_size %d, segsize %d, segtotal %d",
|
|
||||||
buffer_size, spec->segsize, spec->segtotal);
|
|
||||||
|
|
||||||
/* allocate the ringbuffer memory now */
|
|
||||||
buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
|
|
||||||
memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
|
|
||||||
|
|
||||||
if ((res = gst_jack_audio_client_set_active (sink->client, TRUE)))
|
|
||||||
goto could_not_activate;
|
|
||||||
|
|
||||||
/* if we need to automatically connect the ports, do so now. We must do this
|
|
||||||
* after activating the client. */
|
|
||||||
if (sink->connect == GST_JACK_CONNECT_AUTO
|
|
||||||
|| sink->connect == GST_JACK_CONNECT_AUTO_FORCED) {
|
|
||||||
/* find all the physical input ports. A physical input port is a port
|
|
||||||
* associated with a hardware device. Someone needs connect to a physical
|
|
||||||
* port in order to hear something. */
|
|
||||||
ports = jack_get_ports (client, NULL, NULL,
|
|
||||||
JackPortIsPhysical | JackPortIsInput);
|
|
||||||
if (ports == NULL) {
|
|
||||||
/* no ports? fine then we don't do anything except for posting a warning
|
|
||||||
* message. */
|
|
||||||
GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL),
|
|
||||||
("No physical input ports found, leaving ports unconnected"));
|
|
||||||
goto done;
|
|
||||||
}
|
|
||||||
|
|
||||||
for (i = 0; i < channels; i++) {
|
|
||||||
/* stop when all input ports are exhausted */
|
|
||||||
if (ports[i] == NULL) {
|
|
||||||
/* post a warning that we could not connect all ports */
|
|
||||||
GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL),
|
|
||||||
("No more physical ports, leaving some ports unconnected"));
|
|
||||||
break;
|
|
||||||
}
|
|
||||||
GST_DEBUG_OBJECT (sink, "try connecting to %s",
|
|
||||||
jack_port_name (sink->ports[i]));
|
|
||||||
/* connect the port to a physical port */
|
|
||||||
res = jack_connect (client, jack_port_name (sink->ports[i]), ports[i]);
|
|
||||||
if (res != 0 && res != EEXIST)
|
|
||||||
goto cannot_connect;
|
|
||||||
}
|
|
||||||
free (ports);
|
|
||||||
}
|
|
||||||
done:
|
|
||||||
|
|
||||||
abuf->sample_rate = sample_rate;
|
|
||||||
abuf->buffer_size = buffer_size;
|
|
||||||
abuf->channels = spec->channels;
|
|
||||||
|
|
||||||
return TRUE;
|
|
||||||
|
|
||||||
/* ERRORS */
|
|
||||||
wrong_samplerate:
|
|
||||||
{
|
|
||||||
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
|
|
||||||
("Wrong samplerate, server is running at %d and we received %d",
|
|
||||||
sample_rate, spec->rate));
|
|
||||||
return FALSE;
|
|
||||||
}
|
|
||||||
out_of_ports:
|
|
||||||
{
|
|
||||||
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
|
|
||||||
("Cannot allocate more Jack ports"));
|
|
||||||
return FALSE;
|
|
||||||
}
|
|
||||||
could_not_activate:
|
|
||||||
{
|
|
||||||
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
|
|
||||||
("Could not activate client (%d:%s)", res, g_strerror (res)));
|
|
||||||
return FALSE;
|
|
||||||
}
|
|
||||||
cannot_connect:
|
|
||||||
{
|
|
||||||
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
|
|
||||||
("Could not connect output ports to physical ports (%d:%s)",
|
|
||||||
res, g_strerror (res)));
|
|
||||||
free (ports);
|
|
||||||
return FALSE;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
/* function is called with LOCK */
|
|
||||||
static gboolean
|
|
||||||
gst_jack_ring_buffer_release (GstRingBuffer * buf)
|
|
||||||
{
|
|
||||||
GstJackAudioSink *sink;
|
|
||||||
GstJackRingBuffer *abuf;
|
|
||||||
gint res;
|
|
||||||
|
|
||||||
abuf = GST_JACK_RING_BUFFER_CAST (buf);
|
|
||||||
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (sink, "release");
|
|
||||||
|
|
||||||
if ((res = gst_jack_audio_client_set_active (sink->client, FALSE))) {
|
|
||||||
/* we only warn, this means the server is probably shut down and the client
|
|
||||||
* is gone anyway. */
|
|
||||||
GST_ELEMENT_WARNING (sink, RESOURCE, CLOSE, (NULL),
|
|
||||||
("Could not deactivate Jack client (%d)", res));
|
|
||||||
}
|
|
||||||
|
|
||||||
abuf->channels = -1;
|
|
||||||
abuf->buffer_size = -1;
|
|
||||||
abuf->sample_rate = -1;
|
|
||||||
|
|
||||||
/* free the buffer */
|
|
||||||
gst_buffer_unref (buf->data);
|
|
||||||
buf->data = NULL;
|
|
||||||
|
|
||||||
return TRUE;
|
|
||||||
}
|
|
||||||
|
|
||||||
static gboolean
|
|
||||||
gst_jack_ring_buffer_start (GstRingBuffer * buf)
|
|
||||||
{
|
|
||||||
GstJackAudioSink *sink;
|
|
||||||
|
|
||||||
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (sink, "start");
|
|
||||||
|
|
||||||
return TRUE;
|
|
||||||
}
|
|
||||||
|
|
||||||
static gboolean
|
|
||||||
gst_jack_ring_buffer_pause (GstRingBuffer * buf)
|
|
||||||
{
|
|
||||||
GstJackAudioSink *sink;
|
|
||||||
|
|
||||||
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (sink, "pause");
|
|
||||||
|
|
||||||
return TRUE;
|
|
||||||
}
|
|
||||||
|
|
||||||
static gboolean
|
|
||||||
gst_jack_ring_buffer_stop (GstRingBuffer * buf)
|
|
||||||
{
|
|
||||||
GstJackAudioSink *sink;
|
|
||||||
|
|
||||||
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (sink, "stop");
|
|
||||||
|
|
||||||
return TRUE;
|
|
||||||
}
|
|
||||||
|
|
||||||
static guint
|
|
||||||
gst_jack_ring_buffer_delay (GstRingBuffer * buf)
|
|
||||||
{
|
|
||||||
GstJackAudioSink *sink;
|
|
||||||
guint i, res = 0, latency;
|
|
||||||
jack_client_t *client;
|
|
||||||
|
|
||||||
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
|
|
||||||
client = gst_jack_audio_client_get_client (sink->client);
|
|
||||||
|
|
||||||
for (i = 0; i < sink->port_count; i++) {
|
|
||||||
latency = jack_port_get_total_latency (client, sink->ports[i]);
|
|
||||||
if (latency > res)
|
|
||||||
res = latency;
|
|
||||||
}
|
|
||||||
|
|
||||||
GST_LOG_OBJECT (sink, "delay %u", res);
|
|
||||||
|
|
||||||
return res;
|
|
||||||
}
|
|
||||||
|
|
||||||
static GstStaticPadTemplate jackaudiosink_sink_factory =
|
|
||||||
GST_STATIC_PAD_TEMPLATE ("sink",
|
|
||||||
GST_PAD_SINK,
|
|
||||||
GST_PAD_ALWAYS,
|
|
||||||
GST_STATIC_CAPS ("audio/x-raw-float, "
|
|
||||||
"endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
|
|
||||||
"width = (int) 32, "
|
|
||||||
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
|
|
||||||
);
|
|
||||||
|
|
||||||
/* AudioSink signals and args */
|
|
||||||
enum
|
|
||||||
{
|
|
||||||
/* FILL ME */
|
|
||||||
SIGNAL_LAST
|
|
||||||
};
|
|
||||||
|
|
||||||
#define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO
|
|
||||||
#define DEFAULT_PROP_SERVER NULL
|
|
||||||
|
|
||||||
enum
|
|
||||||
{
|
|
||||||
PROP_0,
|
|
||||||
PROP_CONNECT,
|
|
||||||
PROP_SERVER,
|
|
||||||
PROP_CLIENT,
|
|
||||||
PROP_LAST
|
|
||||||
};
|
|
||||||
|
|
||||||
#define _do_init(bla) \
|
|
||||||
GST_DEBUG_CATEGORY_INIT (gst_jack_audio_sink_debug, "jacksink", 0, "jacksink element");
|
|
||||||
|
|
||||||
GST_BOILERPLATE_FULL (GstJackAudioSink, gst_jack_audio_sink, GstBaseAudioSink,
|
|
||||||
GST_TYPE_BASE_AUDIO_SINK, _do_init);
|
|
||||||
|
|
||||||
static void gst_jack_audio_sink_dispose (GObject * object);
|
|
||||||
static void gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
|
|
||||||
const GValue * value, GParamSpec * pspec);
|
|
||||||
static void gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
|
|
||||||
GValue * value, GParamSpec * pspec);
|
|
||||||
|
|
||||||
static GstCaps *gst_jack_audio_sink_getcaps (GstBaseSink * bsink);
|
|
||||||
static GstRingBuffer *gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink *
|
|
||||||
sink);
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_jack_audio_sink_base_init (gpointer g_class)
|
|
||||||
{
|
|
||||||
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
|
|
||||||
|
|
||||||
gst_element_class_set_details_simple (element_class, "Audio Sink (Jack)",
|
|
||||||
"Sink/Audio", "Output to Jack", "Wim Taymans <wim@fluendo.com>");
|
|
||||||
|
|
||||||
gst_element_class_add_pad_template (element_class,
|
|
||||||
gst_static_pad_template_get (&jackaudiosink_sink_factory));
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
|
|
||||||
{
|
|
||||||
GObjectClass *gobject_class;
|
|
||||||
GstElementClass *gstelement_class;
|
|
||||||
GstBaseSinkClass *gstbasesink_class;
|
|
||||||
GstBaseAudioSinkClass *gstbaseaudiosink_class;
|
|
||||||
|
|
||||||
gobject_class = (GObjectClass *) klass;
|
|
||||||
gstelement_class = (GstElementClass *) klass;
|
|
||||||
gstbasesink_class = (GstBaseSinkClass *) klass;
|
|
||||||
gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
|
|
||||||
|
|
||||||
gobject_class->dispose = gst_jack_audio_sink_dispose;
|
|
||||||
gobject_class->get_property = gst_jack_audio_sink_get_property;
|
|
||||||
gobject_class->set_property = gst_jack_audio_sink_set_property;
|
|
||||||
|
|
||||||
g_object_class_install_property (gobject_class, PROP_CONNECT,
|
|
||||||
g_param_spec_enum ("connect", "Connect",
|
|
||||||
"Specify how the output ports will be connected",
|
|
||||||
GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT,
|
|
||||||
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
||||||
|
|
||||||
g_object_class_install_property (gobject_class, PROP_SERVER,
|
|
||||||
g_param_spec_string ("server", "Server",
|
|
||||||
"The Jack server to connect to (NULL = default)",
|
|
||||||
DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
||||||
|
|
||||||
g_object_class_install_property (gobject_class, PROP_CLIENT,
|
|
||||||
g_param_spec_boxed ("client", "JackClient", "Handle for jack client",
|
|
||||||
GST_TYPE_JACK_CLIENT,
|
|
||||||
GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
|
|
||||||
G_PARAM_STATIC_STRINGS));
|
|
||||||
|
|
||||||
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_sink_getcaps);
|
|
||||||
|
|
||||||
gstbaseaudiosink_class->create_ringbuffer =
|
|
||||||
GST_DEBUG_FUNCPTR (gst_jack_audio_sink_create_ringbuffer);
|
|
||||||
|
|
||||||
/* ref class from a thread-safe context to work around missing bit of
|
|
||||||
* thread-safety in GObject */
|
|
||||||
g_type_class_ref (GST_TYPE_JACK_RING_BUFFER);
|
|
||||||
|
|
||||||
gst_jack_audio_client_init ();
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_jack_audio_sink_init (GstJackAudioSink * sink,
|
|
||||||
GstJackAudioSinkClass * g_class)
|
|
||||||
{
|
|
||||||
sink->connect = DEFAULT_PROP_CONNECT;
|
|
||||||
sink->server = g_strdup (DEFAULT_PROP_SERVER);
|
|
||||||
sink->jclient = NULL;
|
|
||||||
sink->ports = NULL;
|
|
||||||
sink->port_count = 0;
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_jack_audio_sink_dispose (GObject * object)
|
|
||||||
{
|
|
||||||
GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (object);
|
|
||||||
|
|
||||||
gst_caps_replace (&sink->caps, NULL);
|
|
||||||
G_OBJECT_CLASS (parent_class)->dispose (object);
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
|
|
||||||
const GValue * value, GParamSpec * pspec)
|
|
||||||
{
|
|
||||||
GstJackAudioSink *sink;
|
|
||||||
|
|
||||||
sink = GST_JACK_AUDIO_SINK (object);
|
|
||||||
|
|
||||||
switch (prop_id) {
|
|
||||||
case PROP_CONNECT:
|
|
||||||
sink->connect = g_value_get_enum (value);
|
|
||||||
break;
|
|
||||||
case PROP_SERVER:
|
|
||||||
g_free (sink->server);
|
|
||||||
sink->server = g_value_dup_string (value);
|
|
||||||
break;
|
|
||||||
case PROP_CLIENT:
|
|
||||||
if (GST_STATE (sink) == GST_STATE_NULL ||
|
|
||||||
GST_STATE (sink) == GST_STATE_READY) {
|
|
||||||
sink->jclient = g_value_get_boxed (value);
|
|
||||||
}
|
|
||||||
break;
|
|
||||||
default:
|
|
||||||
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
||||||
break;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
|
|
||||||
GValue * value, GParamSpec * pspec)
|
|
||||||
{
|
|
||||||
GstJackAudioSink *sink;
|
|
||||||
|
|
||||||
sink = GST_JACK_AUDIO_SINK (object);
|
|
||||||
|
|
||||||
switch (prop_id) {
|
|
||||||
case PROP_CONNECT:
|
|
||||||
g_value_set_enum (value, sink->connect);
|
|
||||||
break;
|
|
||||||
case PROP_SERVER:
|
|
||||||
g_value_set_string (value, sink->server);
|
|
||||||
break;
|
|
||||||
case PROP_CLIENT:
|
|
||||||
g_value_set_boxed (value, sink->jclient);
|
|
||||||
break;
|
|
||||||
default:
|
|
||||||
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
||||||
break;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
static GstCaps *
|
|
||||||
gst_jack_audio_sink_getcaps (GstBaseSink * bsink)
|
|
||||||
{
|
|
||||||
GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (bsink);
|
|
||||||
const char **ports;
|
|
||||||
gint min, max;
|
|
||||||
gint rate;
|
|
||||||
jack_client_t *client;
|
|
||||||
|
|
||||||
if (sink->client == NULL)
|
|
||||||
goto no_client;
|
|
||||||
|
|
||||||
client = gst_jack_audio_client_get_client (sink->client);
|
|
||||||
|
|
||||||
if (sink->connect == GST_JACK_CONNECT_AUTO) {
|
|
||||||
/* get a port count, this is the number of channels we can automatically
|
|
||||||
* connect. */
|
|
||||||
ports = jack_get_ports (client, NULL, NULL,
|
|
||||||
JackPortIsPhysical | JackPortIsInput);
|
|
||||||
max = 0;
|
|
||||||
if (ports != NULL) {
|
|
||||||
for (; ports[max]; max++);
|
|
||||||
free (ports);
|
|
||||||
} else
|
|
||||||
max = 0;
|
|
||||||
} else {
|
|
||||||
/* we allow any number of pads, something else is going to connect the
|
|
||||||
* pads. */
|
|
||||||
max = G_MAXINT;
|
|
||||||
}
|
|
||||||
min = MIN (1, max);
|
|
||||||
|
|
||||||
rate = jack_get_sample_rate (client);
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (sink, "got %d-%d ports, samplerate: %d", min, max, rate);
|
|
||||||
|
|
||||||
if (!sink->caps) {
|
|
||||||
sink->caps = gst_caps_new_simple ("audio/x-raw-float",
|
|
||||||
"endianness", G_TYPE_INT, G_BYTE_ORDER,
|
|
||||||
"width", G_TYPE_INT, 32,
|
|
||||||
"rate", G_TYPE_INT, rate,
|
|
||||||
"channels", GST_TYPE_INT_RANGE, min, max, NULL);
|
|
||||||
}
|
|
||||||
GST_INFO_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, sink->caps);
|
|
||||||
|
|
||||||
return gst_caps_ref (sink->caps);
|
|
||||||
|
|
||||||
/* ERRORS */
|
|
||||||
no_client:
|
|
||||||
{
|
|
||||||
GST_DEBUG_OBJECT (sink, "device not open, using template caps");
|
|
||||||
/* base class will get template caps for us when we return NULL */
|
|
||||||
return NULL;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
static GstRingBuffer *
|
|
||||||
gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
|
|
||||||
{
|
|
||||||
GstRingBuffer *buffer;
|
|
||||||
|
|
||||||
buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL);
|
|
||||||
GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer);
|
|
||||||
|
|
||||||
return buffer;
|
|
||||||
}
|
|
||||||
|
|
@ -1,78 +0,0 @@
|
||||||
/* GStreamer
|
|
||||||
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
|
|
||||||
*
|
|
||||||
* gstjacksink.h:
|
|
||||||
*
|
|
||||||
* This library is free software; you can redistribute it and/or
|
|
||||||
* modify it under the terms of the GNU Library General Public
|
|
||||||
* License as published by the Free Software Foundation; either
|
|
||||||
* version 2 of the License, or (at your option) any later version.
|
|
||||||
*
|
|
||||||
* This library is distributed in the hope that it will be useful,
|
|
||||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
||||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
||||||
* Library General Public License for more details.
|
|
||||||
*
|
|
||||||
* You should have received a copy of the GNU Library General Public
|
|
||||||
* License along with this library; if not, write to the
|
|
||||||
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
|
||||||
* Boston, MA 02111-1307, USA.
|
|
||||||
*/
|
|
||||||
|
|
||||||
#ifndef __GST_JACK_AUDIO_SINK_H__
|
|
||||||
#define __GST_JACK_AUDIO_SINK_H__
|
|
||||||
|
|
||||||
#include <jack/jack.h>
|
|
||||||
|
|
||||||
#include <gst/gst.h>
|
|
||||||
#include <gst/audio/gstbaseaudiosink.h>
|
|
||||||
|
|
||||||
#include "gstjack.h"
|
|
||||||
#include "gstjackaudioclient.h"
|
|
||||||
|
|
||||||
G_BEGIN_DECLS
|
|
||||||
|
|
||||||
#define GST_TYPE_JACK_AUDIO_SINK (gst_jack_audio_sink_get_type())
|
|
||||||
#define GST_JACK_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_AUDIO_SINK,GstJackAudioSink))
|
|
||||||
#define GST_JACK_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_AUDIO_SINK,GstJackAudioSinkClass))
|
|
||||||
#define GST_JACK_AUDIO_SINK_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_JACK_AUDIO_SINK,GstJackAudioSinkClass))
|
|
||||||
#define GST_IS_JACK_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_AUDIO_SINK))
|
|
||||||
#define GST_IS_JACK_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_AUDIO_SINK))
|
|
||||||
|
|
||||||
typedef struct _GstJackAudioSink GstJackAudioSink;
|
|
||||||
typedef struct _GstJackAudioSinkClass GstJackAudioSinkClass;
|
|
||||||
|
|
||||||
/**
|
|
||||||
* GstJackAudioSink:
|
|
||||||
*
|
|
||||||
* Opaque #GstJackAudioSink.
|
|
||||||
*/
|
|
||||||
struct _GstJackAudioSink {
|
|
||||||
GstBaseAudioSink element;
|
|
||||||
|
|
||||||
/*< private >*/
|
|
||||||
/* cached caps */
|
|
||||||
GstCaps *caps;
|
|
||||||
|
|
||||||
/* properties */
|
|
||||||
GstJackConnect connect;
|
|
||||||
gchar *server;
|
|
||||||
jack_client_t *jclient;
|
|
||||||
|
|
||||||
/* our client */
|
|
||||||
GstJackAudioClient *client;
|
|
||||||
|
|
||||||
/* our ports */
|
|
||||||
jack_port_t **ports;
|
|
||||||
int port_count;
|
|
||||||
};
|
|
||||||
|
|
||||||
struct _GstJackAudioSinkClass {
|
|
||||||
GstBaseAudioSinkClass parent_class;
|
|
||||||
};
|
|
||||||
|
|
||||||
GType gst_jack_audio_sink_get_type (void);
|
|
||||||
|
|
||||||
G_END_DECLS
|
|
||||||
|
|
||||||
#endif /* __GST_JACK_AUDIO_SINK_H__ */
|
|
||||||
|
|
@ -1,874 +0,0 @@
|
||||||
/* GStreamer
|
|
||||||
* Copyright (C) 2008 Tristan Matthews <tristan@sat.qc.ca>
|
|
||||||
*
|
|
||||||
* Permission is hereby granted, free of charge, to any person obtaining a
|
|
||||||
* copy of this software and associated documentation files (the "Software"),
|
|
||||||
* to deal in the Software without restriction, including without limitation
|
|
||||||
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
|
|
||||||
* and/or sell copies of the Software, and to permit persons to whom the
|
|
||||||
* Software is furnished to do so, subject to the following conditions:
|
|
||||||
*
|
|
||||||
* The above copyright notice and this permission notice shall be included in
|
|
||||||
* all copies or substantial portions of the Software.
|
|
||||||
*
|
|
||||||
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
|
|
||||||
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
|
|
||||||
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
|
|
||||||
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
|
|
||||||
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
|
|
||||||
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
|
|
||||||
* DEALINGS IN THE SOFTWARE.
|
|
||||||
*
|
|
||||||
* Alternatively, the contents of this file may be used under the
|
|
||||||
* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
|
|
||||||
* which case the following provisions apply instead of the ones
|
|
||||||
* mentioned above:
|
|
||||||
*
|
|
||||||
* This library is free software; you can redistribute it and/or
|
|
||||||
* modify it under the terms of the GNU Library General Public
|
|
||||||
* License as published by the Free Software Foundation; either
|
|
||||||
* version 2 of the License, or (at your option) any later version.
|
|
||||||
*
|
|
||||||
* This library is distributed in the hope that it will be useful,
|
|
||||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
||||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
||||||
* Library General Public License for more details.
|
|
||||||
*
|
|
||||||
* You should have received a copy of the GNU Library General Public
|
|
||||||
* License along with this library; if not, write to the
|
|
||||||
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
|
||||||
* Boston, MA 02111-1307, USA.
|
|
||||||
*/
|
|
||||||
|
|
||||||
/**
|
|
||||||
* SECTION:element-jackaudiosrc
|
|
||||||
* @see_also: #GstBaseAudioSrc, #GstRingBuffer
|
|
||||||
*
|
|
||||||
* A Src that inputs data from Jack ports.
|
|
||||||
*
|
|
||||||
* It will create N Jack ports named in_<name>_<num> where
|
|
||||||
* <name> is the element name and <num> is starting from 1.
|
|
||||||
* Each port corresponds to a gstreamer channel.
|
|
||||||
*
|
|
||||||
* The samplerate as exposed on the caps is always the same as the samplerate of
|
|
||||||
* the jack server.
|
|
||||||
*
|
|
||||||
* When the #GstJackAudioSrc:connect property is set to auto, this element
|
|
||||||
* will try to connect each input port to a random physical jack output pin.
|
|
||||||
*
|
|
||||||
* When the #GstJackAudioSrc:connect property is set to none, the element will
|
|
||||||
* accept any number of output channels and will create (but not connect) an
|
|
||||||
* input port for each channel.
|
|
||||||
*
|
|
||||||
* The element will generate an error when the Jack server is shut down when it
|
|
||||||
* was PAUSED or PLAYING. This element does not support dynamic rate and buffer
|
|
||||||
* size changes at runtime.
|
|
||||||
*
|
|
||||||
* <refsect2>
|
|
||||||
* <title>Example launch line</title>
|
|
||||||
* |[
|
|
||||||
* gst-launch jackaudiosrc connect=0 ! jackaudiosink connect=0
|
|
||||||
* ]| Get audio input into gstreamer from jack.
|
|
||||||
* </refsect2>
|
|
||||||
*
|
|
||||||
* Last reviewed on 2008-07-22 (0.10.4)
|
|
||||||
*/
|
|
||||||
|
|
||||||
#ifdef HAVE_CONFIG_H
|
|
||||||
#include "config.h"
|
|
||||||
#endif
|
|
||||||
|
|
||||||
#include <gst/gst-i18n-plugin.h>
|
|
||||||
#include <stdlib.h>
|
|
||||||
#include <string.h>
|
|
||||||
|
|
||||||
#include "gstjackaudiosrc.h"
|
|
||||||
#include "gstjackringbuffer.h"
|
|
||||||
#include "gstjackutil.h"
|
|
||||||
|
|
||||||
GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_src_debug);
|
|
||||||
#define GST_CAT_DEFAULT gst_jack_audio_src_debug
|
|
||||||
|
|
||||||
static gboolean
|
|
||||||
gst_jack_audio_src_allocate_channels (GstJackAudioSrc * src, gint channels)
|
|
||||||
{
|
|
||||||
jack_client_t *client;
|
|
||||||
|
|
||||||
client = gst_jack_audio_client_get_client (src->client);
|
|
||||||
|
|
||||||
/* remove ports we don't need */
|
|
||||||
while (src->port_count > channels)
|
|
||||||
jack_port_unregister (client, src->ports[--src->port_count]);
|
|
||||||
|
|
||||||
/* alloc enough input ports */
|
|
||||||
src->ports = g_realloc (src->ports, sizeof (jack_port_t *) * channels);
|
|
||||||
src->buffers = g_realloc (src->buffers, sizeof (sample_t *) * channels);
|
|
||||||
|
|
||||||
/* create an input port for each channel */
|
|
||||||
while (src->port_count < channels) {
|
|
||||||
gchar *name;
|
|
||||||
|
|
||||||
/* port names start from 1 and are local to the element */
|
|
||||||
name =
|
|
||||||
g_strdup_printf ("in_%s_%d", GST_ELEMENT_NAME (src),
|
|
||||||
src->port_count + 1);
|
|
||||||
src->ports[src->port_count] =
|
|
||||||
jack_port_register (client, name, JACK_DEFAULT_AUDIO_TYPE,
|
|
||||||
JackPortIsInput, 0);
|
|
||||||
if (src->ports[src->port_count] == NULL)
|
|
||||||
return FALSE;
|
|
||||||
|
|
||||||
src->port_count++;
|
|
||||||
|
|
||||||
g_free (name);
|
|
||||||
}
|
|
||||||
return TRUE;
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_jack_audio_src_free_channels (GstJackAudioSrc * src)
|
|
||||||
{
|
|
||||||
gint res, i = 0;
|
|
||||||
jack_client_t *client;
|
|
||||||
|
|
||||||
client = gst_jack_audio_client_get_client (src->client);
|
|
||||||
|
|
||||||
/* get rid of all ports */
|
|
||||||
while (src->port_count) {
|
|
||||||
GST_LOG_OBJECT (src, "unregister port %d", i);
|
|
||||||
if ((res = jack_port_unregister (client, src->ports[i++])))
|
|
||||||
GST_DEBUG_OBJECT (src, "unregister of port failed (%d)", res);
|
|
||||||
|
|
||||||
src->port_count--;
|
|
||||||
}
|
|
||||||
g_free (src->ports);
|
|
||||||
src->ports = NULL;
|
|
||||||
g_free (src->buffers);
|
|
||||||
src->buffers = NULL;
|
|
||||||
}
|
|
||||||
|
|
||||||
/* ringbuffer abstract base class */
|
|
||||||
static GType
|
|
||||||
gst_jack_ring_buffer_get_type (void)
|
|
||||||
{
|
|
||||||
static GType ringbuffer_type = 0;
|
|
||||||
|
|
||||||
if (!ringbuffer_type) {
|
|
||||||
static const GTypeInfo ringbuffer_info = { sizeof (GstJackRingBufferClass),
|
|
||||||
NULL,
|
|
||||||
NULL,
|
|
||||||
(GClassInitFunc) gst_jack_ring_buffer_class_init,
|
|
||||||
NULL,
|
|
||||||
NULL,
|
|
||||||
sizeof (GstJackRingBuffer),
|
|
||||||
0,
|
|
||||||
(GInstanceInitFunc) gst_jack_ring_buffer_init,
|
|
||||||
NULL
|
|
||||||
};
|
|
||||||
|
|
||||||
ringbuffer_type =
|
|
||||||
g_type_register_static (GST_TYPE_RING_BUFFER,
|
|
||||||
"GstJackAudioSrcRingBuffer", &ringbuffer_info, 0);
|
|
||||||
}
|
|
||||||
return ringbuffer_type;
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass)
|
|
||||||
{
|
|
||||||
GObjectClass *gobject_class;
|
|
||||||
GstObjectClass *gstobject_class;
|
|
||||||
GstRingBufferClass *gstringbuffer_class;
|
|
||||||
|
|
||||||
gobject_class = (GObjectClass *) klass;
|
|
||||||
gstobject_class = (GstObjectClass *) klass;
|
|
||||||
gstringbuffer_class = (GstRingBufferClass *) klass;
|
|
||||||
|
|
||||||
ring_parent_class = g_type_class_peek_parent (klass);
|
|
||||||
|
|
||||||
gstringbuffer_class->open_device =
|
|
||||||
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_open_device);
|
|
||||||
gstringbuffer_class->close_device =
|
|
||||||
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_close_device);
|
|
||||||
gstringbuffer_class->acquire =
|
|
||||||
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_acquire);
|
|
||||||
gstringbuffer_class->release =
|
|
||||||
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_release);
|
|
||||||
gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
|
|
||||||
gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_pause);
|
|
||||||
gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
|
|
||||||
gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_stop);
|
|
||||||
|
|
||||||
gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_delay);
|
|
||||||
}
|
|
||||||
|
|
||||||
/* this is the callback of jack. This should be RT-safe.
|
|
||||||
* Writes samples from the jack input port's buffer to the gst ring buffer.
|
|
||||||
*/
|
|
||||||
static int
|
|
||||||
jack_process_cb (jack_nframes_t nframes, void *arg)
|
|
||||||
{
|
|
||||||
GstJackAudioSrc *src;
|
|
||||||
GstRingBuffer *buf;
|
|
||||||
gint len;
|
|
||||||
guint8 *writeptr;
|
|
||||||
gint writeseg;
|
|
||||||
gint channels, i, j, flen;
|
|
||||||
sample_t *data;
|
|
||||||
|
|
||||||
buf = GST_RING_BUFFER_CAST (arg);
|
|
||||||
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
|
|
||||||
|
|
||||||
channels = buf->spec.channels;
|
|
||||||
|
|
||||||
/* get input buffers */
|
|
||||||
for (i = 0; i < channels; i++)
|
|
||||||
src->buffers[i] =
|
|
||||||
(sample_t *) jack_port_get_buffer (src->ports[i], nframes);
|
|
||||||
|
|
||||||
if (gst_ring_buffer_prepare_read (buf, &writeseg, &writeptr, &len)) {
|
|
||||||
flen = len / channels;
|
|
||||||
|
|
||||||
/* the number of samples must be exactly the segment size */
|
|
||||||
if (nframes * sizeof (sample_t) != flen)
|
|
||||||
goto wrong_size;
|
|
||||||
|
|
||||||
/* the samples in the jack input buffers have to be interleaved into the
|
|
||||||
* ringbuffer */
|
|
||||||
data = (sample_t *) writeptr;
|
|
||||||
for (i = 0; i < nframes; ++i)
|
|
||||||
for (j = 0; j < channels; ++j)
|
|
||||||
*data++ = src->buffers[j][i];
|
|
||||||
|
|
||||||
GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, writeptr,
|
|
||||||
len / channels, channels);
|
|
||||||
|
|
||||||
/* we wrote one segment */
|
|
||||||
gst_ring_buffer_advance (buf, 1);
|
|
||||||
}
|
|
||||||
return 0;
|
|
||||||
|
|
||||||
/* ERRORS */
|
|
||||||
wrong_size:
|
|
||||||
{
|
|
||||||
GST_ERROR_OBJECT (src, "nbytes (%d) != flen (%d)",
|
|
||||||
(gint) (nframes * sizeof (sample_t)), flen);
|
|
||||||
return 1;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
/* we error out */
|
|
||||||
static int
|
|
||||||
jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
|
|
||||||
{
|
|
||||||
GstJackAudioSrc *src;
|
|
||||||
GstJackRingBuffer *abuf;
|
|
||||||
|
|
||||||
abuf = GST_JACK_RING_BUFFER_CAST (arg);
|
|
||||||
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
|
|
||||||
|
|
||||||
if (abuf->sample_rate != -1 && abuf->sample_rate != nframes)
|
|
||||||
goto not_supported;
|
|
||||||
|
|
||||||
return 0;
|
|
||||||
|
|
||||||
/* ERRORS */
|
|
||||||
not_supported:
|
|
||||||
{
|
|
||||||
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
|
|
||||||
(NULL), ("Jack changed the sample rate, which is not supported"));
|
|
||||||
return 1;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
/* we error out */
|
|
||||||
static int
|
|
||||||
jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
|
|
||||||
{
|
|
||||||
GstJackAudioSrc *src;
|
|
||||||
GstJackRingBuffer *abuf;
|
|
||||||
|
|
||||||
abuf = GST_JACK_RING_BUFFER_CAST (arg);
|
|
||||||
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
|
|
||||||
|
|
||||||
if (abuf->buffer_size != -1 && abuf->buffer_size != nframes)
|
|
||||||
goto not_supported;
|
|
||||||
|
|
||||||
return 0;
|
|
||||||
|
|
||||||
/* ERRORS */
|
|
||||||
not_supported:
|
|
||||||
{
|
|
||||||
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
|
|
||||||
(NULL), ("Jack changed the buffer size, which is not supported"));
|
|
||||||
return 1;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
jack_shutdown_cb (void *arg)
|
|
||||||
{
|
|
||||||
GstJackAudioSrc *src;
|
|
||||||
|
|
||||||
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (src, "shutdown");
|
|
||||||
|
|
||||||
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
|
|
||||||
(NULL), ("Jack server shutdown"));
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_jack_ring_buffer_init (GstJackRingBuffer * buf,
|
|
||||||
GstJackRingBufferClass * g_class)
|
|
||||||
{
|
|
||||||
buf->channels = -1;
|
|
||||||
buf->buffer_size = -1;
|
|
||||||
buf->sample_rate = -1;
|
|
||||||
}
|
|
||||||
|
|
||||||
/* the _open_device method should make a connection with the server
|
|
||||||
*/
|
|
||||||
static gboolean
|
|
||||||
gst_jack_ring_buffer_open_device (GstRingBuffer * buf)
|
|
||||||
{
|
|
||||||
GstJackAudioSrc *src;
|
|
||||||
jack_status_t status = 0;
|
|
||||||
const gchar *name;
|
|
||||||
|
|
||||||
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (src, "open");
|
|
||||||
|
|
||||||
name = g_get_application_name ();
|
|
||||||
if (!name)
|
|
||||||
name = "GStreamer";
|
|
||||||
|
|
||||||
src->client = gst_jack_audio_client_new (name, src->server,
|
|
||||||
src->jclient,
|
|
||||||
GST_JACK_CLIENT_SOURCE,
|
|
||||||
jack_shutdown_cb,
|
|
||||||
jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status);
|
|
||||||
if (src->client == NULL)
|
|
||||||
goto could_not_open;
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (src, "opened");
|
|
||||||
|
|
||||||
return TRUE;
|
|
||||||
|
|
||||||
/* ERRORS */
|
|
||||||
could_not_open:
|
|
||||||
{
|
|
||||||
if (status & JackServerFailed) {
|
|
||||||
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
|
|
||||||
(_("Jack server not found")),
|
|
||||||
("Cannot connect to the Jack server (status %d)", status));
|
|
||||||
} else {
|
|
||||||
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_WRITE,
|
|
||||||
(NULL), ("Jack client open error (status %d)", status));
|
|
||||||
}
|
|
||||||
return FALSE;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
/* close the connection with the server
|
|
||||||
*/
|
|
||||||
static gboolean
|
|
||||||
gst_jack_ring_buffer_close_device (GstRingBuffer * buf)
|
|
||||||
{
|
|
||||||
GstJackAudioSrc *src;
|
|
||||||
|
|
||||||
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (src, "close");
|
|
||||||
|
|
||||||
gst_jack_audio_src_free_channels (src);
|
|
||||||
gst_jack_audio_client_free (src->client);
|
|
||||||
src->client = NULL;
|
|
||||||
|
|
||||||
return TRUE;
|
|
||||||
}
|
|
||||||
|
|
||||||
|
|
||||||
/* allocate a buffer and setup resources to process the audio samples of
|
|
||||||
* the format as specified in @spec.
|
|
||||||
*
|
|
||||||
* We allocate N jack ports, one for each channel. If we are asked to
|
|
||||||
* automatically make a connection with physical ports, we connect as many
|
|
||||||
* ports as there are physical ports, leaving leftover ports unconnected.
|
|
||||||
*
|
|
||||||
* It is assumed that samplerate and number of channels are acceptable since our
|
|
||||||
* getcaps method will always provide correct values. If unacceptable caps are
|
|
||||||
* received for some reason, we fail here.
|
|
||||||
*/
|
|
||||||
static gboolean
|
|
||||||
gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
|
|
||||||
{
|
|
||||||
GstJackAudioSrc *src;
|
|
||||||
GstJackRingBuffer *abuf;
|
|
||||||
const char **ports;
|
|
||||||
gint sample_rate, buffer_size;
|
|
||||||
gint i, channels, res;
|
|
||||||
jack_client_t *client;
|
|
||||||
|
|
||||||
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
|
|
||||||
abuf = GST_JACK_RING_BUFFER_CAST (buf);
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (src, "acquire");
|
|
||||||
|
|
||||||
client = gst_jack_audio_client_get_client (src->client);
|
|
||||||
|
|
||||||
/* sample rate must be that of the server */
|
|
||||||
sample_rate = jack_get_sample_rate (client);
|
|
||||||
if (sample_rate != spec->rate)
|
|
||||||
goto wrong_samplerate;
|
|
||||||
|
|
||||||
channels = spec->channels;
|
|
||||||
|
|
||||||
if (!gst_jack_audio_src_allocate_channels (src, channels))
|
|
||||||
goto out_of_ports;
|
|
||||||
|
|
||||||
gst_jack_set_layout_on_caps (&spec->caps, channels);
|
|
||||||
|
|
||||||
buffer_size = jack_get_buffer_size (client);
|
|
||||||
|
|
||||||
/* the segment size in bytes, this is large enough to hold a buffer of 32bit floats
|
|
||||||
* for all channels */
|
|
||||||
spec->segsize = buffer_size * sizeof (gfloat) * channels;
|
|
||||||
spec->latency_time = gst_util_uint64_scale (spec->segsize,
|
|
||||||
(GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample);
|
|
||||||
/* segtotal based on buffer-time latency */
|
|
||||||
spec->segtotal = spec->buffer_time / spec->latency_time;
|
|
||||||
if (spec->segtotal < 2) {
|
|
||||||
spec->segtotal = 2;
|
|
||||||
spec->buffer_time = spec->latency_time * spec->segtotal;
|
|
||||||
}
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (src, "buffer time: %" G_GINT64_FORMAT " usec",
|
|
||||||
spec->buffer_time);
|
|
||||||
GST_DEBUG_OBJECT (src, "latency time: %" G_GINT64_FORMAT " usec",
|
|
||||||
spec->latency_time);
|
|
||||||
GST_DEBUG_OBJECT (src, "buffer_size %d, segsize %d, segtotal %d",
|
|
||||||
buffer_size, spec->segsize, spec->segtotal);
|
|
||||||
|
|
||||||
/* allocate the ringbuffer memory now */
|
|
||||||
buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
|
|
||||||
memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
|
|
||||||
|
|
||||||
if ((res = gst_jack_audio_client_set_active (src->client, TRUE)))
|
|
||||||
goto could_not_activate;
|
|
||||||
|
|
||||||
/* if we need to automatically connect the ports, do so now. We must do this
|
|
||||||
* after activating the client. */
|
|
||||||
if (src->connect == GST_JACK_CONNECT_AUTO
|
|
||||||
|| src->connect == GST_JACK_CONNECT_AUTO_FORCED) {
|
|
||||||
/* find all the physical output ports. A physical output port is a port
|
|
||||||
* associated with a hardware device. Someone needs connect to a physical
|
|
||||||
* port in order to capture something. */
|
|
||||||
ports =
|
|
||||||
jack_get_ports (client, NULL, NULL,
|
|
||||||
JackPortIsPhysical | JackPortIsOutput);
|
|
||||||
if (ports == NULL) {
|
|
||||||
/* no ports? fine then we don't do anything except for posting a warning
|
|
||||||
* message. */
|
|
||||||
GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL),
|
|
||||||
("No physical output ports found, leaving ports unconnected"));
|
|
||||||
goto done;
|
|
||||||
}
|
|
||||||
|
|
||||||
for (i = 0; i < channels; i++) {
|
|
||||||
/* stop when all output ports are exhausted */
|
|
||||||
if (ports[i] == NULL) {
|
|
||||||
/* post a warning that we could not connect all ports */
|
|
||||||
GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL),
|
|
||||||
("No more physical ports, leaving some ports unconnected"));
|
|
||||||
break;
|
|
||||||
}
|
|
||||||
GST_DEBUG_OBJECT (src, "try connecting to %s",
|
|
||||||
jack_port_name (src->ports[i]));
|
|
||||||
|
|
||||||
/* connect the physical port to a port */
|
|
||||||
res = jack_connect (client, ports[i], jack_port_name (src->ports[i]));
|
|
||||||
if (res != 0 && res != EEXIST)
|
|
||||||
goto cannot_connect;
|
|
||||||
}
|
|
||||||
free (ports);
|
|
||||||
}
|
|
||||||
done:
|
|
||||||
|
|
||||||
abuf->sample_rate = sample_rate;
|
|
||||||
abuf->buffer_size = buffer_size;
|
|
||||||
abuf->channels = spec->channels;
|
|
||||||
|
|
||||||
return TRUE;
|
|
||||||
|
|
||||||
/* ERRORS */
|
|
||||||
wrong_samplerate:
|
|
||||||
{
|
|
||||||
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
|
|
||||||
("Wrong samplerate, server is running at %d and we received %d",
|
|
||||||
sample_rate, spec->rate));
|
|
||||||
return FALSE;
|
|
||||||
}
|
|
||||||
out_of_ports:
|
|
||||||
{
|
|
||||||
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
|
|
||||||
("Cannot allocate more Jack ports"));
|
|
||||||
return FALSE;
|
|
||||||
}
|
|
||||||
could_not_activate:
|
|
||||||
{
|
|
||||||
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
|
|
||||||
("Could not activate client (%d:%s)", res, g_strerror (res)));
|
|
||||||
return FALSE;
|
|
||||||
}
|
|
||||||
cannot_connect:
|
|
||||||
{
|
|
||||||
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
|
|
||||||
("Could not connect input ports to physical ports (%d:%s)",
|
|
||||||
res, g_strerror (res)));
|
|
||||||
free (ports);
|
|
||||||
return FALSE;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
/* function is called with LOCK */
|
|
||||||
static gboolean
|
|
||||||
gst_jack_ring_buffer_release (GstRingBuffer * buf)
|
|
||||||
{
|
|
||||||
GstJackAudioSrc *src;
|
|
||||||
GstJackRingBuffer *abuf;
|
|
||||||
gint res;
|
|
||||||
|
|
||||||
abuf = GST_JACK_RING_BUFFER_CAST (buf);
|
|
||||||
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (src, "release");
|
|
||||||
|
|
||||||
if ((res = gst_jack_audio_client_set_active (src->client, FALSE))) {
|
|
||||||
/* we only warn, this means the server is probably shut down and the client
|
|
||||||
* is gone anyway. */
|
|
||||||
GST_ELEMENT_WARNING (src, RESOURCE, CLOSE, (NULL),
|
|
||||||
("Could not deactivate Jack client (%d)", res));
|
|
||||||
}
|
|
||||||
|
|
||||||
abuf->channels = -1;
|
|
||||||
abuf->buffer_size = -1;
|
|
||||||
abuf->sample_rate = -1;
|
|
||||||
|
|
||||||
/* free the buffer */
|
|
||||||
gst_buffer_unref (buf->data);
|
|
||||||
buf->data = NULL;
|
|
||||||
|
|
||||||
return TRUE;
|
|
||||||
}
|
|
||||||
|
|
||||||
static gboolean
|
|
||||||
gst_jack_ring_buffer_start (GstRingBuffer * buf)
|
|
||||||
{
|
|
||||||
GstJackAudioSrc *src;
|
|
||||||
|
|
||||||
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (src, "start");
|
|
||||||
|
|
||||||
return TRUE;
|
|
||||||
}
|
|
||||||
|
|
||||||
static gboolean
|
|
||||||
gst_jack_ring_buffer_pause (GstRingBuffer * buf)
|
|
||||||
{
|
|
||||||
GstJackAudioSrc *src;
|
|
||||||
|
|
||||||
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (src, "pause");
|
|
||||||
|
|
||||||
return TRUE;
|
|
||||||
}
|
|
||||||
|
|
||||||
static gboolean
|
|
||||||
gst_jack_ring_buffer_stop (GstRingBuffer * buf)
|
|
||||||
{
|
|
||||||
GstJackAudioSrc *src;
|
|
||||||
|
|
||||||
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (src, "stop");
|
|
||||||
|
|
||||||
return TRUE;
|
|
||||||
}
|
|
||||||
|
|
||||||
static guint
|
|
||||||
gst_jack_ring_buffer_delay (GstRingBuffer * buf)
|
|
||||||
{
|
|
||||||
GstJackAudioSrc *src;
|
|
||||||
guint i, res = 0, latency;
|
|
||||||
jack_client_t *client;
|
|
||||||
|
|
||||||
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
|
|
||||||
client = gst_jack_audio_client_get_client (src->client);
|
|
||||||
|
|
||||||
for (i = 0; i < src->port_count; i++) {
|
|
||||||
latency = jack_port_get_total_latency (client, src->ports[i]);
|
|
||||||
if (latency > res)
|
|
||||||
res = latency;
|
|
||||||
}
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (src, "delay %u", res);
|
|
||||||
|
|
||||||
return res;
|
|
||||||
}
|
|
||||||
|
|
||||||
/* Audiosrc signals and args */
|
|
||||||
enum
|
|
||||||
{
|
|
||||||
/* FILL ME */
|
|
||||||
LAST_SIGNAL
|
|
||||||
};
|
|
||||||
|
|
||||||
#define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO
|
|
||||||
#define DEFAULT_PROP_SERVER NULL
|
|
||||||
|
|
||||||
enum
|
|
||||||
{
|
|
||||||
PROP_0,
|
|
||||||
PROP_CONNECT,
|
|
||||||
PROP_SERVER,
|
|
||||||
PROP_CLIENT,
|
|
||||||
PROP_LAST
|
|
||||||
};
|
|
||||||
|
|
||||||
|
|
||||||
/* the capabilities of the inputs and outputs.
|
|
||||||
*
|
|
||||||
* describe the real formats here.
|
|
||||||
*/
|
|
||||||
|
|
||||||
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
|
|
||||||
GST_PAD_SRC,
|
|
||||||
GST_PAD_ALWAYS,
|
|
||||||
GST_STATIC_CAPS ("audio/x-raw-float, "
|
|
||||||
"endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
|
|
||||||
"width = (int) 32, "
|
|
||||||
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
|
|
||||||
);
|
|
||||||
|
|
||||||
#define _do_init(bla) \
|
|
||||||
GST_DEBUG_CATEGORY_INIT(gst_jack_audio_src_debug, "jacksrc", 0, "jacksrc element");
|
|
||||||
|
|
||||||
GST_BOILERPLATE_FULL (GstJackAudioSrc, gst_jack_audio_src, GstBaseAudioSrc,
|
|
||||||
GST_TYPE_BASE_AUDIO_SRC, _do_init);
|
|
||||||
|
|
||||||
static void gst_jack_audio_src_dispose (GObject * object);
|
|
||||||
static void gst_jack_audio_src_set_property (GObject * object, guint prop_id,
|
|
||||||
const GValue * value, GParamSpec * pspec);
|
|
||||||
static void gst_jack_audio_src_get_property (GObject * object, guint prop_id,
|
|
||||||
GValue * value, GParamSpec * pspec);
|
|
||||||
|
|
||||||
static GstCaps *gst_jack_audio_src_getcaps (GstBaseSrc * bsrc);
|
|
||||||
static GstRingBuffer *gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc *
|
|
||||||
src);
|
|
||||||
|
|
||||||
/* GObject vmethod implementations */
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_jack_audio_src_base_init (gpointer gclass)
|
|
||||||
{
|
|
||||||
GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
|
|
||||||
|
|
||||||
gst_element_class_add_pad_template (element_class,
|
|
||||||
gst_static_pad_template_get (&src_factory));
|
|
||||||
gst_element_class_set_details_simple (element_class, "Audio Source (Jack)",
|
|
||||||
"Source/Audio",
|
|
||||||
"Input from Jack", "Tristan Matthews <tristan@sat.qc.ca>");
|
|
||||||
}
|
|
||||||
|
|
||||||
/* initialize the jack_audio_src's class */
|
|
||||||
static void
|
|
||||||
gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
|
|
||||||
{
|
|
||||||
GObjectClass *gobject_class;
|
|
||||||
GstElementClass *gstelement_class;
|
|
||||||
GstBaseSrcClass *gstbasesrc_class;
|
|
||||||
GstBaseAudioSrcClass *gstbaseaudiosrc_class;
|
|
||||||
|
|
||||||
gobject_class = (GObjectClass *) klass;
|
|
||||||
gstelement_class = (GstElementClass *) klass;
|
|
||||||
|
|
||||||
gstbasesrc_class = (GstBaseSrcClass *) klass;
|
|
||||||
gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
|
|
||||||
|
|
||||||
gobject_class->dispose = gst_jack_audio_src_dispose;
|
|
||||||
gobject_class->set_property = gst_jack_audio_src_set_property;
|
|
||||||
gobject_class->get_property = gst_jack_audio_src_get_property;
|
|
||||||
|
|
||||||
g_object_class_install_property (gobject_class, PROP_CONNECT,
|
|
||||||
g_param_spec_enum ("connect", "Connect",
|
|
||||||
"Specify how the input ports will be connected",
|
|
||||||
GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT,
|
|
||||||
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
||||||
|
|
||||||
g_object_class_install_property (gobject_class, PROP_SERVER,
|
|
||||||
g_param_spec_string ("server", "Server",
|
|
||||||
"The Jack server to connect to (NULL = default)",
|
|
||||||
DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
||||||
|
|
||||||
g_object_class_install_property (gobject_class, PROP_CLIENT,
|
|
||||||
g_param_spec_boxed ("client", "JackClient", "Handle for jack client",
|
|
||||||
GST_TYPE_JACK_CLIENT,
|
|
||||||
GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
|
|
||||||
G_PARAM_STATIC_STRINGS));
|
|
||||||
|
|
||||||
gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_src_getcaps);
|
|
||||||
gstbaseaudiosrc_class->create_ringbuffer =
|
|
||||||
GST_DEBUG_FUNCPTR (gst_jack_audio_src_create_ringbuffer);
|
|
||||||
|
|
||||||
/* ref class from a thread-safe context to work around missing bit of
|
|
||||||
* thread-safety in GObject */
|
|
||||||
g_type_class_ref (GST_TYPE_JACK_RING_BUFFER);
|
|
||||||
|
|
||||||
gst_jack_audio_client_init ();
|
|
||||||
}
|
|
||||||
|
|
||||||
/* initialize the new element
|
|
||||||
* instantiate pads and add them to element
|
|
||||||
* set pad calback functions
|
|
||||||
* initialize instance structure
|
|
||||||
*/
|
|
||||||
static void
|
|
||||||
gst_jack_audio_src_init (GstJackAudioSrc * src, GstJackAudioSrcClass * gclass)
|
|
||||||
{
|
|
||||||
//gst_base_src_set_live(GST_BASE_SRC (src), TRUE);
|
|
||||||
src->connect = DEFAULT_PROP_CONNECT;
|
|
||||||
src->server = g_strdup (DEFAULT_PROP_SERVER);
|
|
||||||
src->jclient = NULL;
|
|
||||||
src->ports = NULL;
|
|
||||||
src->port_count = 0;
|
|
||||||
src->buffers = NULL;
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_jack_audio_src_dispose (GObject * object)
|
|
||||||
{
|
|
||||||
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
|
|
||||||
|
|
||||||
gst_caps_replace (&src->caps, NULL);
|
|
||||||
G_OBJECT_CLASS (parent_class)->dispose (object);
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_jack_audio_src_set_property (GObject * object, guint prop_id,
|
|
||||||
const GValue * value, GParamSpec * pspec)
|
|
||||||
{
|
|
||||||
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
|
|
||||||
|
|
||||||
switch (prop_id) {
|
|
||||||
case PROP_CONNECT:
|
|
||||||
src->connect = g_value_get_enum (value);
|
|
||||||
break;
|
|
||||||
case PROP_SERVER:
|
|
||||||
g_free (src->server);
|
|
||||||
src->server = g_value_dup_string (value);
|
|
||||||
break;
|
|
||||||
case PROP_CLIENT:
|
|
||||||
if (GST_STATE (src) == GST_STATE_NULL ||
|
|
||||||
GST_STATE (src) == GST_STATE_READY) {
|
|
||||||
src->jclient = g_value_get_boxed (value);
|
|
||||||
}
|
|
||||||
break;
|
|
||||||
default:
|
|
||||||
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
||||||
break;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
static void
|
|
||||||
gst_jack_audio_src_get_property (GObject * object, guint prop_id,
|
|
||||||
GValue * value, GParamSpec * pspec)
|
|
||||||
{
|
|
||||||
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
|
|
||||||
|
|
||||||
switch (prop_id) {
|
|
||||||
case PROP_CONNECT:
|
|
||||||
g_value_set_enum (value, src->connect);
|
|
||||||
break;
|
|
||||||
case PROP_SERVER:
|
|
||||||
g_value_set_string (value, src->server);
|
|
||||||
break;
|
|
||||||
case PROP_CLIENT:
|
|
||||||
g_value_set_boxed (value, src->jclient);
|
|
||||||
break;
|
|
||||||
default:
|
|
||||||
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
||||||
break;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
static GstCaps *
|
|
||||||
gst_jack_audio_src_getcaps (GstBaseSrc * bsrc)
|
|
||||||
{
|
|
||||||
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (bsrc);
|
|
||||||
const char **ports;
|
|
||||||
gint min, max;
|
|
||||||
gint rate;
|
|
||||||
jack_client_t *client;
|
|
||||||
|
|
||||||
if (src->client == NULL)
|
|
||||||
goto no_client;
|
|
||||||
|
|
||||||
client = gst_jack_audio_client_get_client (src->client);
|
|
||||||
|
|
||||||
if (src->connect == GST_JACK_CONNECT_AUTO) {
|
|
||||||
/* get a port count, this is the number of channels we can automatically
|
|
||||||
* connect. */
|
|
||||||
ports = jack_get_ports (client, NULL, NULL,
|
|
||||||
JackPortIsPhysical | JackPortIsOutput);
|
|
||||||
max = 0;
|
|
||||||
if (ports != NULL) {
|
|
||||||
for (; ports[max]; max++);
|
|
||||||
|
|
||||||
free (ports);
|
|
||||||
} else
|
|
||||||
max = 0;
|
|
||||||
} else {
|
|
||||||
/* we allow any number of pads, something else is going to connect the
|
|
||||||
* pads. */
|
|
||||||
max = G_MAXINT;
|
|
||||||
}
|
|
||||||
min = MIN (1, max);
|
|
||||||
|
|
||||||
rate = jack_get_sample_rate (client);
|
|
||||||
|
|
||||||
GST_DEBUG_OBJECT (src, "got %d-%d ports, samplerate: %d", min, max, rate);
|
|
||||||
|
|
||||||
if (!src->caps) {
|
|
||||||
src->caps = gst_caps_new_simple ("audio/x-raw-float",
|
|
||||||
"endianness", G_TYPE_INT, G_BYTE_ORDER,
|
|
||||||
"width", G_TYPE_INT, 32,
|
|
||||||
"rate", G_TYPE_INT, rate,
|
|
||||||
"channels", GST_TYPE_INT_RANGE, min, max, NULL);
|
|
||||||
}
|
|
||||||
GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, src->caps);
|
|
||||||
|
|
||||||
return gst_caps_ref (src->caps);
|
|
||||||
|
|
||||||
/* ERRORS */
|
|
||||||
no_client:
|
|
||||||
{
|
|
||||||
GST_DEBUG_OBJECT (src, "device not open, using template caps");
|
|
||||||
/* base class will get template caps for us when we return NULL */
|
|
||||||
return NULL;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
static GstRingBuffer *
|
|
||||||
gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc * src)
|
|
||||||
{
|
|
||||||
GstRingBuffer *buffer;
|
|
||||||
|
|
||||||
buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL);
|
|
||||||
GST_DEBUG_OBJECT (src, "created ringbuffer @%p", buffer);
|
|
||||||
|
|
||||||
return buffer;
|
|
||||||
}
|
|
||||||
|
|
@ -1,97 +0,0 @@
|
||||||
/* GStreamer
|
|
||||||
* Copyright (C) 2008 Tristan Matthews <tristan@sat.qc.ca>
|
|
||||||
*
|
|
||||||
* Permission is hereby granted, free of charge, to any person obtaining a
|
|
||||||
* copy of this software and associated documentation files (the "Software"),
|
|
||||||
* to deal in the Software without restriction, including without limitation
|
|
||||||
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
|
|
||||||
* and/or sell copies of the Software, and to permit persons to whom the
|
|
||||||
* Software is furnished to do so, subject to the following conditions:
|
|
||||||
*
|
|
||||||
* The above copyright notice and this permission notice shall be included in
|
|
||||||
* all copies or substantial portions of the Software.
|
|
||||||
*
|
|
||||||
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
|
|
||||||
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
|
|
||||||
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
|
|
||||||
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
|
|
||||||
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
|
|
||||||
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
|
|
||||||
* DEALINGS IN THE SOFTWARE.
|
|
||||||
*
|
|
||||||
* Alternatively, the contents of this file may be used under the
|
|
||||||
* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
|
|
||||||
* which case the following provisions apply instead of the ones
|
|
||||||
* mentioned above:
|
|
||||||
*
|
|
||||||
* This library is free software; you can redistribute it and/or
|
|
||||||
* modify it under the terms of the GNU Library General Public
|
|
||||||
* License as published by the Free Software Foundation; either
|
|
||||||
* version 2 of the License, or (at your option) any later version.
|
|
||||||
*
|
|
||||||
* This library is distributed in the hope that it will be useful,
|
|
||||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
||||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
||||||
* Library General Public License for more details.
|
|
||||||
*
|
|
||||||
* You should have received a copy of the GNU Library General Public
|
|
||||||
* License along with this library; if not, write to the
|
|
||||||
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
|
||||||
* Boston, MA 02111-1307, USA.
|
|
||||||
*/
|
|
||||||
|
|
||||||
#ifndef __GST_JACK_AUDIO_SRC_H__
|
|
||||||
#define __GST_JACK_AUDIO_SRC_H__
|
|
||||||
|
|
||||||
#include <jack/jack.h>
|
|
||||||
|
|
||||||
#include <gst/gst.h>
|
|
||||||
#include <gst/audio/gstaudiosrc.h>
|
|
||||||
|
|
||||||
#include "gstjackaudioclient.h"
|
|
||||||
#include "gstjack.h"
|
|
||||||
|
|
||||||
G_BEGIN_DECLS
|
|
||||||
|
|
||||||
#define GST_TYPE_JACK_AUDIO_SRC (gst_jack_audio_src_get_type())
|
|
||||||
#define GST_JACK_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrc))
|
|
||||||
#define GST_JACK_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrcClass))
|
|
||||||
#define GST_JACK_AUDIO_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrcClass))
|
|
||||||
#define GST_IS_JACK_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_AUDIO_SRC))
|
|
||||||
#define GST_IS_JACK_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_AUDIO_SRC))
|
|
||||||
|
|
||||||
typedef struct _GstJackAudioSrc GstJackAudioSrc;
|
|
||||||
typedef struct _GstJackAudioSrcClass GstJackAudioSrcClass;
|
|
||||||
|
|
||||||
struct _GstJackAudioSrc
|
|
||||||
{
|
|
||||||
GstBaseAudioSrc src;
|
|
||||||
|
|
||||||
/*< private >*/
|
|
||||||
/* cached caps */
|
|
||||||
GstCaps *caps;
|
|
||||||
|
|
||||||
/* properties */
|
|
||||||
GstJackConnect connect;
|
|
||||||
gchar *server;
|
|
||||||
jack_client_t *jclient;
|
|
||||||
|
|
||||||
/* our client */
|
|
||||||
GstJackAudioClient *client;
|
|
||||||
|
|
||||||
/* our ports */
|
|
||||||
jack_port_t **ports;
|
|
||||||
int port_count;
|
|
||||||
sample_t **buffers;
|
|
||||||
};
|
|
||||||
|
|
||||||
struct _GstJackAudioSrcClass
|
|
||||||
{
|
|
||||||
GstBaseAudioSrcClass parent_class;
|
|
||||||
};
|
|
||||||
|
|
||||||
GType gst_jack_audio_src_get_type (void);
|
|
||||||
|
|
||||||
G_END_DECLS
|
|
||||||
|
|
||||||
#endif /* __GST_JACK_AUDIO_SRC_H__ */
|
|
||||||
|
|
@ -1,88 +0,0 @@
|
||||||
/*
|
|
||||||
* GStreamer
|
|
||||||
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
|
|
||||||
* Copyright (C) 2008 Tristan Matthews <tristan@sat.qc.ca>
|
|
||||||
*
|
|
||||||
* Permission is hereby granted, free of charge, to any person obtaining a
|
|
||||||
* copy of this software and associated documentation files (the "Software"),
|
|
||||||
* to deal in the Software without restriction, including without limitation
|
|
||||||
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
|
|
||||||
* and/or sell copies of the Software, and to permit persons to whom the
|
|
||||||
* Software is furnished to do so, subject to the following conditions:
|
|
||||||
*
|
|
||||||
* The above copyright notice and this permission notice shall be included in
|
|
||||||
* all copies or substantial portions of the Software.
|
|
||||||
*
|
|
||||||
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
|
|
||||||
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
|
|
||||||
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
|
|
||||||
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
|
|
||||||
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
|
|
||||||
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
|
|
||||||
* DEALINGS IN THE SOFTWARE.
|
|
||||||
*
|
|
||||||
* Alternatively, the contents of this file may be used under the
|
|
||||||
* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
|
|
||||||
* which case the following provisions apply instead of the ones
|
|
||||||
* mentioned above:
|
|
||||||
*
|
|
||||||
* This library is free software; you can redistribute it and/or
|
|
||||||
* modify it under the terms of the GNU Library General Public
|
|
||||||
* License as published by the Free Software Foundation; either
|
|
||||||
* version 2 of the License, or (at your option) any later version.
|
|
||||||
*
|
|
||||||
* This library is distributed in the hope that it will be useful,
|
|
||||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
||||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
||||||
* Library General Public License for more details.
|
|
||||||
*
|
|
||||||
* You should have received a copy of the GNU Library General Public
|
|
||||||
* License along with this library; if not, write to the
|
|
||||||
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
|
||||||
* Boston, MA 02111-1307, USA.
|
|
||||||
*/
|
|
||||||
|
|
||||||
#ifndef __GST_JACK_RING_BUFFER_H__
|
|
||||||
#define __GST_JACK_RING_BUFFER_H__
|
|
||||||
|
|
||||||
#define GST_TYPE_JACK_RING_BUFFER (gst_jack_ring_buffer_get_type())
|
|
||||||
#define GST_JACK_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_RING_BUFFER,GstJackRingBuffer))
|
|
||||||
#define GST_JACK_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_RING_BUFFER,GstJackRingBufferClass))
|
|
||||||
#define GST_JACK_RING_BUFFER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_JACK_RING_BUFFER,GstJackRingBufferClass))
|
|
||||||
#define GST_JACK_RING_BUFFER_CAST(obj) ((GstJackRingBuffer *)obj)
|
|
||||||
#define GST_IS_JACK_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_RING_BUFFER))
|
|
||||||
#define GST_IS_JACK_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_RING_BUFFER))
|
|
||||||
|
|
||||||
typedef struct _GstJackRingBuffer GstJackRingBuffer;
|
|
||||||
typedef struct _GstJackRingBufferClass GstJackRingBufferClass;
|
|
||||||
|
|
||||||
struct _GstJackRingBuffer
|
|
||||||
{
|
|
||||||
GstRingBuffer object;
|
|
||||||
|
|
||||||
gint sample_rate;
|
|
||||||
gint buffer_size;
|
|
||||||
gint channels;
|
|
||||||
};
|
|
||||||
|
|
||||||
struct _GstJackRingBufferClass
|
|
||||||
{
|
|
||||||
GstRingBufferClass parent_class;
|
|
||||||
};
|
|
||||||
|
|
||||||
static void gst_jack_ring_buffer_class_init(GstJackRingBufferClass * klass);
|
|
||||||
static void gst_jack_ring_buffer_init(GstJackRingBuffer * ringbuffer,
|
|
||||||
GstJackRingBufferClass * klass);
|
|
||||||
|
|
||||||
static GstRingBufferClass *ring_parent_class = NULL;
|
|
||||||
|
|
||||||
static gboolean gst_jack_ring_buffer_open_device(GstRingBuffer * buf);
|
|
||||||
static gboolean gst_jack_ring_buffer_close_device(GstRingBuffer * buf);
|
|
||||||
static gboolean gst_jack_ring_buffer_acquire(GstRingBuffer * buf,GstRingBufferSpec * spec);
|
|
||||||
static gboolean gst_jack_ring_buffer_release(GstRingBuffer * buf);
|
|
||||||
static gboolean gst_jack_ring_buffer_start(GstRingBuffer * buf);
|
|
||||||
static gboolean gst_jack_ring_buffer_pause(GstRingBuffer * buf);
|
|
||||||
static gboolean gst_jack_ring_buffer_stop(GstRingBuffer * buf);
|
|
||||||
static guint gst_jack_ring_buffer_delay(GstRingBuffer * buf);
|
|
||||||
|
|
||||||
#endif
|
|
||||||
|
|
@ -1,114 +0,0 @@
|
||||||
/* GStreamer Jack utility functions
|
|
||||||
* Copyright (C) 2010 Tristan Matthews <tristan@sat.qc.ca>
|
|
||||||
*
|
|
||||||
* This library is free software; you can redistribute it and/or
|
|
||||||
* modify it under the terms of the GNU Library General Public
|
|
||||||
* License as published by the Free Software Foundation; either
|
|
||||||
* version 2 of the License, or (at your option) any later version.
|
|
||||||
*
|
|
||||||
* This library is distributed in the hope that it will be useful,
|
|
||||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
||||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
||||||
* Library General Public License for more details.
|
|
||||||
*
|
|
||||||
* You should have received a copy of the GNU Library General Public
|
|
||||||
* License along with this library; if not, write to the
|
|
||||||
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
|
||||||
* Boston, MA 02111-1307, USA.
|
|
||||||
*/
|
|
||||||
|
|
||||||
#include "gstjackutil.h"
|
|
||||||
#include <gst/audio/multichannel.h>
|
|
||||||
|
|
||||||
static const GstAudioChannelPosition default_positions[8][8] = {
|
|
||||||
/* 1 channel */
|
|
||||||
{
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_FRONT_MONO,
|
|
||||||
},
|
|
||||||
/* 2 channels */
|
|
||||||
{
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
|
|
||||||
},
|
|
||||||
/* 3 channels (2.1) */
|
|
||||||
{
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_LFE, /* or FRONT_CENTER for 3.0? */
|
|
||||||
},
|
|
||||||
/* 4 channels (4.0 or 3.1?) */
|
|
||||||
{
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
|
|
||||||
},
|
|
||||||
/* 5 channels */
|
|
||||||
{
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
|
|
||||||
},
|
|
||||||
/* 6 channels */
|
|
||||||
{
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_LFE,
|
|
||||||
},
|
|
||||||
/* 7 channels */
|
|
||||||
{
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_LFE,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
|
|
||||||
},
|
|
||||||
/* 8 channels */
|
|
||||||
{
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_LFE,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
|
|
||||||
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
|
|
||||||
}
|
|
||||||
};
|
|
||||||
|
|
||||||
|
|
||||||
/* if channels are less than or equal to 8, we set a default layout,
|
|
||||||
* otherwise set layout to an array of GST_AUDIO_CHANNEL_POSITION_NONE */
|
|
||||||
void
|
|
||||||
gst_jack_set_layout_on_caps (GstCaps ** caps, gint channels)
|
|
||||||
{
|
|
||||||
int c;
|
|
||||||
GValue pos = { 0 };
|
|
||||||
GValue chanpos = { 0 };
|
|
||||||
gst_caps_unref (*caps);
|
|
||||||
|
|
||||||
if (channels <= 8) {
|
|
||||||
g_assert (channels >= 1);
|
|
||||||
gst_audio_set_channel_positions (gst_caps_get_structure (*caps, 0),
|
|
||||||
default_positions[channels - 1]);
|
|
||||||
} else {
|
|
||||||
g_value_init (&chanpos, GST_TYPE_ARRAY);
|
|
||||||
g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
|
|
||||||
for (c = 0; c < channels; c++) {
|
|
||||||
g_value_set_enum (&pos, GST_AUDIO_CHANNEL_POSITION_NONE);
|
|
||||||
gst_value_array_append_value (&chanpos, &pos);
|
|
||||||
}
|
|
||||||
g_value_unset (&pos);
|
|
||||||
gst_structure_set_value (gst_caps_get_structure (*caps, 0),
|
|
||||||
"channel-positions", &chanpos);
|
|
||||||
g_value_unset (&chanpos);
|
|
||||||
}
|
|
||||||
gst_caps_ref (*caps);
|
|
||||||
}
|
|
||||||
|
|
@ -1,30 +0,0 @@
|
||||||
/* GStreamer
|
|
||||||
* Copyright (C) 2010 Tristan Matthews <tristan@sat.qc.ca>
|
|
||||||
*
|
|
||||||
* gstjackutil.h:
|
|
||||||
*
|
|
||||||
* This library is free software; you can redistribute it and/or
|
|
||||||
* modify it under the terms of the GNU Library General Public
|
|
||||||
* License as published by the Free Software Foundation; either
|
|
||||||
* version 2 of the License, or (at your option) any later version.
|
|
||||||
*
|
|
||||||
* This library is distributed in the hope that it will be useful,
|
|
||||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
||||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
||||||
* Library General Public License for more details.
|
|
||||||
*
|
|
||||||
* You should have received a copy of the GNU Library General Public
|
|
||||||
* License along with this library; if not, write to the
|
|
||||||
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
|
||||||
* Boston, MA 02111-1307, USA.
|
|
||||||
*/
|
|
||||||
|
|
||||||
#ifndef _GST_JACK_UTIL_H_
|
|
||||||
#define _GST_JACK_UTIL_H_
|
|
||||||
|
|
||||||
#include <gst/gst.h>
|
|
||||||
|
|
||||||
void
|
|
||||||
gst_jack_set_layout_on_caps (GstCaps **caps, gint channels);
|
|
||||||
|
|
||||||
#endif // _GST_JACK_UTIL_H_
|
|
||||||
|
|
@ -165,7 +165,6 @@ rm -rf $RPM_BUILD_ROOT
|
||||||
@USE_DC1394_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstdc1394.so
|
@USE_DC1394_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstdc1394.so
|
||||||
@USE_TIMIDITY_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgsttimidity.so
|
@USE_TIMIDITY_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgsttimidity.so
|
||||||
@USE_WILDMIDI_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstwildmidi.so
|
@USE_WILDMIDI_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstwildmidi.so
|
||||||
@USE_JACK_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstjack.so
|
|
||||||
@USE_SNDFILE_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstsndfile.so
|
@USE_SNDFILE_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstsndfile.so
|
||||||
@USE_CELT_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstcelt.so
|
@USE_CELT_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstcelt.so
|
||||||
@USE_MPEG2ENC_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstmpeg2enc.so
|
@USE_MPEG2ENC_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstmpeg2enc.so
|
||||||
|
|
|
||||||
|
|
@ -1,9 +1,4 @@
|
||||||
if HAVE_GTK
|
if HAVE_GTK
|
||||||
if USE_JACK
|
|
||||||
JACK_EXAMPLES=jack
|
|
||||||
else
|
|
||||||
JACK_EXAMPLES=
|
|
||||||
endif
|
|
||||||
GTK_EXAMPLES=camerabin mxf scaletempo camerabin2
|
GTK_EXAMPLES=camerabin mxf scaletempo camerabin2
|
||||||
else
|
else
|
||||||
GTK_EXAMPLES=
|
GTK_EXAMPLES=
|
||||||
|
|
@ -21,7 +16,7 @@ else
|
||||||
CAMERABIN2=
|
CAMERABIN2=
|
||||||
endif
|
endif
|
||||||
|
|
||||||
SUBDIRS= $(DIRECTFB_DIR) $(GTK_EXAMPLES) $(JACK_EXAMPLES)
|
SUBDIRS= $(DIRECTFB_DIR) $(GTK_EXAMPLES)
|
||||||
DIST_SUBDIRS= camerabin $(CAMERABIN2) directfb jack mxf scaletempo
|
DIST_SUBDIRS= camerabin $(CAMERABIN2) directfb mxf scaletempo
|
||||||
|
|
||||||
include $(top_srcdir)/common/parallel-subdirs.mak
|
include $(top_srcdir)/common/parallel-subdirs.mak
|
||||||
|
|
|
||||||
|
|
@ -1,6 +0,0 @@
|
||||||
noinst_PROGRAMS = jack_client
|
|
||||||
|
|
||||||
jack_client_SOURCES = jack_client.c
|
|
||||||
jack_client_CFLAGS = $(GST_CFLAGS) $(GTK_CFLAGS) $(JACK_CFLAGS)
|
|
||||||
jack_client_LDFLAGS = $(GST_LIBS) $(GTK_LIBS) $(JACK_LIBS)
|
|
||||||
|
|
||||||
|
|
@ -1,79 +0,0 @@
|
||||||
/* This app demonstrates the creation and use of a jack client in conjunction
|
|
||||||
* with the jack plugins. This way, an application can control the jack client
|
|
||||||
* directly.
|
|
||||||
*/
|
|
||||||
|
|
||||||
#include <gst/gst.h>
|
|
||||||
#include <gtk/gtk.h>
|
|
||||||
#include <jack/jack.h>
|
|
||||||
|
|
||||||
static gboolean
|
|
||||||
quit_cb (gpointer data)
|
|
||||||
{
|
|
||||||
gtk_main_quit ();
|
|
||||||
return FALSE;
|
|
||||||
}
|
|
||||||
|
|
||||||
int
|
|
||||||
main (int argc, char **argv)
|
|
||||||
{
|
|
||||||
jack_client_t *src_client, *sink_client;
|
|
||||||
jack_status_t status;
|
|
||||||
GstElement *pipeline, *src, *sink;
|
|
||||||
GstStateChangeReturn ret;
|
|
||||||
|
|
||||||
gst_init (&argc, &argv);
|
|
||||||
|
|
||||||
/* create jack clients */
|
|
||||||
src_client = jack_client_open ("src_client", JackNoStartServer, &status);
|
|
||||||
if (src_client == NULL) {
|
|
||||||
if (status & JackServerFailed)
|
|
||||||
g_print ("JACK server not running\n");
|
|
||||||
else
|
|
||||||
g_print ("jack_client_open() failed, status = 0x%2.0x\n", status);
|
|
||||||
return 1;
|
|
||||||
}
|
|
||||||
|
|
||||||
sink_client = jack_client_open ("sink_client", JackNoStartServer, &status);
|
|
||||||
if (sink_client == NULL) {
|
|
||||||
if (status & JackServerFailed)
|
|
||||||
g_print ("JACK server not running\n");
|
|
||||||
else
|
|
||||||
g_print ("jack_client_open() failed, status = 0x%2.0x\n", status);
|
|
||||||
return 1;
|
|
||||||
}
|
|
||||||
|
|
||||||
/* create gst elements */
|
|
||||||
pipeline = gst_pipeline_new ("my_pipeline");
|
|
||||||
|
|
||||||
src = gst_element_factory_make ("jackaudiosrc", NULL);
|
|
||||||
sink = gst_element_factory_make ("jackaudiosink", NULL);
|
|
||||||
|
|
||||||
g_object_set (src, "client", src_client, NULL);
|
|
||||||
g_object_set (sink, "client", sink_client, NULL);
|
|
||||||
|
|
||||||
gst_bin_add_many (GST_BIN (pipeline), src, sink, NULL);
|
|
||||||
|
|
||||||
/* link everything together */
|
|
||||||
if (!gst_element_link (src, sink)) {
|
|
||||||
g_print ("Failed to link elements!\n");
|
|
||||||
return 1;
|
|
||||||
}
|
|
||||||
|
|
||||||
/* run */
|
|
||||||
ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
|
|
||||||
if (ret == GST_STATE_CHANGE_FAILURE) {
|
|
||||||
g_print ("Failed to start up pipeline!\n");
|
|
||||||
return 1;
|
|
||||||
}
|
|
||||||
|
|
||||||
/* quit after 5 seconds */
|
|
||||||
g_timeout_add (5000, (GSourceFunc) quit_cb, NULL);
|
|
||||||
gtk_main ();
|
|
||||||
|
|
||||||
/* clean up */
|
|
||||||
gst_element_set_state (pipeline, GST_STATE_NULL);
|
|
||||||
gst_object_unref (pipeline);
|
|
||||||
|
|
||||||
return 0;
|
|
||||||
}
|
|
||||||
Loading…
Reference in a new issue