mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-01-11 09:55:36 +00:00
move wavpack plugin. See #352605.
Original commit message from CVS: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-autodetect.xml: * docs/plugins/inspect/plugin-gconfelements.xml: * docs/plugins/inspect/plugin-ladspa.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-wavpack.xml: * ext/Makefile.am: * tests/check/Makefile.am: move wavpack plugin. See #352605.
This commit is contained in:
Thomas Vander Stichele
parent
c0b48c47d0
commit
4e78ffa15c
15 changed files with 353 additions and 2257 deletions
18
ChangeLog
18
ChangeLog
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@ -1,3 +1,21 @@
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2007-06-08 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac:
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-good-plugins-docs.sgml:
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* docs/plugins/gst-plugins-good-plugins-sections.txt:
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* docs/plugins/gst-plugins-good-plugins.args:
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* docs/plugins/gst-plugins-good-plugins.hierarchy:
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* docs/plugins/gst-plugins-good-plugins.signals:
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* docs/plugins/inspect/plugin-autodetect.xml:
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* docs/plugins/inspect/plugin-gconfelements.xml:
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* docs/plugins/inspect/plugin-ladspa.xml:
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* docs/plugins/inspect/plugin-rtp.xml:
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* docs/plugins/inspect/plugin-wavpack.xml:
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* ext/Makefile.am:
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* tests/check/Makefile.am:
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move wavpack plugin. See #352605.
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2007-06-08 Jan Schmidt <thaytan@mad.scientist.com>
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* configure.ac:
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21
configure.ac
21
configure.ac
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@ -216,6 +216,9 @@ dnl *** checks for header files ***
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dnl check if we have ANSI C header files
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AC_HEADER_STDC
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dnl used by ext/wavpack
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AX_CREATE_STDINT_H
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dnl used in gst/rtp/gstasteriskh263.c
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AC_CHECK_HEADERS([netinet/in.h])
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AC_CHECK_HEADERS([winsock2.h])
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@ -849,6 +852,22 @@ AG_GST_CHECK_FEATURE(TAGLIB, [taglib tagging library], taglib, [
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fi
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])
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dnl *** wavpack ***
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translit(dnm, m, l) AM_CONDITIONAL(USE_WAVPACK, true)
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AG_GST_CHECK_FEATURE(WAVPACK, [wavpack plug-in], wavpack, [
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PKG_CHECK_MODULES(WAVPACK, wavpack >= 4.40.0, HAVE_WAVPACK=yes, [
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PKG_CHECK_MODULES(WAVPACK, wavpack >= 4.20, [
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HAVE_WAVPACK=yes
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AC_DEFINE(WAVPACK_OLD_API, 1, [old wavpack API])
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],[
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HAVE_WAVPACK=no
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AC_MSG_RESULT(no)
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])
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])
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AC_SUBST(WAVPACK_CFLAGS)
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AC_SUBST(WAVPACK_LIBS)
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])
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dnl *** id3demux prefers to have zlib ***
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translit(dnm, m, l) AM_CONDITIONAL(USE_ZLIB, true)
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AG_GST_CHECK_FEATURE(ZLIB, [zlib support for id3demux],, [
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@ -890,6 +909,7 @@ AM_CONDITIONAL(USE_DV1394, false)
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AM_CONDITIONAL(USE_SHOUT2, false)
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AM_CONDITIONAL(USE_SPEEX, false)
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AM_CONDITIONAL(USE_TAGLIB, false)
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AM_CONDITIONAL(USE_WAVPACK, false)
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AM_CONDITIONAL(USE_ZLIB, false)
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fi dnl of EXT plugins
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@ -990,6 +1010,7 @@ ext/raw1394/Makefile
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ext/shout2/Makefile
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ext/speex/Makefile
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ext/taglib/Makefile
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ext/wavpack/Makefile
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sys/Makefile
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sys/directdraw/Makefile
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sys/directsound/Makefile
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@ -118,6 +118,9 @@ EXTRA_HFILES = \
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$(top_srcdir)/gst/videofilter/gstvideobalance.h \
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$(top_srcdir)/gst/videofilter/gstvideoflip.h \
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$(top_srcdir)/gst/videomixer/videomixer.c \
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$(top_srcdir)/ext/wavpack/gstwavpackdec.h \
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$(top_srcdir)/ext/wavpack/gstwavpackenc.h \
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$(top_srcdir)/ext/wavpack/gstwavpackparse.h \
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$(top_srcdir)/gst/wavparse/gstwavparse.h \
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$(top_srcdir)/sys/directdraw/gstdirectdrawsink.h \
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$(top_srcdir)/sys/directsound/gstdirectsoundsink.h \
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@ -59,6 +59,9 @@
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<xi:include href="xml/element-videobalance.xml" />
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<xi:include href="xml/element-videoflip.xml" />
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<xi:include href="xml/element-videomixer.xml" />
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<xi:include href="xml/element-wavpackdec.xml" />
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<xi:include href="xml/element-wavpackenc.xml" />
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<xi:include href="xml/element-wavpackparse.xml" />
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<xi:include href="xml/element-wavparse.xml" />
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<!-- <xi:include href="xml/element-v4l2src.xml" /> -->
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<xi:include href="xml/element-ximagesrc.xml" />
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@ -116,6 +119,7 @@
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<xi:include href="xml/plugin-videoflip.xml" />
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<xi:include href="xml/plugin-videomixer.xml" />
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<xi:include href="xml/plugin-wavenc.xml" />
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<xi:include href="xml/plugin-wavpack.xml" />
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<xi:include href="xml/plugin-wavparse.xml" />
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<!-- <xi:include href="xml/plugin-video4linux2.xml" /> -->
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<xi:include href="xml/plugin-ximagesrc.xml" />
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@ -686,6 +686,30 @@ ROUND_UP_4
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ROUND_UP_8
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</SECTION>
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<SECTION>
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<FILE>element-wavpackdec</FILE>
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GstWavpackDec
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<TITLE>wavpackdec</TITLE>
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<SUBSECTION Standard>
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GstWavpackDecClass
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</SECTION>
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<SECTION>
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<FILE>element-wavpackenc</FILE>
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GstWavpackEnc
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<TITLE>wavpackenc</TITLE>
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<SUBSECTION Standard>
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GstWavpackEncClass
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</SECTION>
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<SECTION>
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<FILE>element-wavpackparse</FILE>
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GstWavpackParse
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<TITLE>wavpackparse</TITLE>
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<SUBSECTION Standard>
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GstWavpackParseClass
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</SECTION>
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<SECTION>
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<FILE>element-wavparse</FILE>
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GstWavParse
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@ -188,6 +188,26 @@
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<DEFAULT>0</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstUDPSrc::closefd</NAME>
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<TYPE>gboolean</TYPE>
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<RANGE></RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Close sockfd</NICK>
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<BLURB>Close sockfd if passed as property on state change.</BLURB>
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<DEFAULT>TRUE</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstUDPSrc::skip-first-bytes</NAME>
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<TYPE>gint</TYPE>
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<RANGE>>= 0</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Skip first bytes</NICK>
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<BLURB>number of bytes to skip for each udp packet.</BLURB>
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<DEFAULT>0</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstSMPTE::border</NAME>
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<TYPE>gint</TYPE>
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@ -284,10 +304,30 @@
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<RANGE></RANGE>
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<FLAGS>rwx</FLAGS>
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<NICK>Timeout</NICK>
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<BLURB>Retry TCP transport after timeout microseconds (0 = disabled).</BLURB>
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<BLURB>Retry TCP transport after UDP timeout microseconds (0 = disabled).</BLURB>
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<DEFAULT>5000000</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstRTSPSrc::latency</NAME>
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<TYPE>guint</TYPE>
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<RANGE></RANGE>
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<FLAGS>rwx</FLAGS>
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<NICK>Buffer latency in ms</NICK>
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<BLURB>Amount of ms to buffer.</BLURB>
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<DEFAULT>3000</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstRTSPSrc::tcp-timeout</NAME>
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<TYPE>guint64</TYPE>
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<RANGE></RANGE>
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<FLAGS>rwx</FLAGS>
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<NICK>TCP Timeout</NICK>
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<BLURB>Fail after timeout microseconds on TCP connections (0 = disabled).</BLURB>
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<DEFAULT>20000000</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstRTPDec::skip</NAME>
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<TYPE>gint</TYPE>
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@ -298,6 +338,16 @@
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<DEFAULT>0</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstRTPDec::latency</NAME>
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<TYPE>guint</TYPE>
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<RANGE></RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Buffer latency in ms</NICK>
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<BLURB>Amount of ms to buffer.</BLURB>
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<DEFAULT>200</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstLevel::interval</NAME>
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<TYPE>guint64</TYPE>
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@ -588,6 +638,16 @@
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<DEFAULT>""</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstShout2send::username</NAME>
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<TYPE>gchararray</TYPE>
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<RANGE></RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>username</NICK>
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<BLURB>username.</BLURB>
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<DEFAULT>"source"</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstSpeexDec::enh</NAME>
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<TYPE>gboolean</TYPE>
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@ -1741,11 +1801,21 @@
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<ARG>
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<NAME>GstDynUDPSink::sockfd</NAME>
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<TYPE>gint</TYPE>
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<RANGE>[0,32767]</RANGE>
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<RANGE>[-1,32767]</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>socket handle</NICK>
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<BLURB>Socket to use for UDP reception.</BLURB>
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<DEFAULT>0</DEFAULT>
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<BLURB>Socket to use for UDP sending. (-1 == allocate).</BLURB>
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<DEFAULT>-1</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstDynUDPSink::closefd</NAME>
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<TYPE>gboolean</TYPE>
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<RANGE></RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Close sockfd</NICK>
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<BLURB>Close sockfd if passed as property on state change.</BLURB>
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<DEFAULT>TRUE</DEFAULT>
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</ARG>
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<ARG>
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@ -16718,3 +16788,103 @@
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<DEFAULT>FALSE</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstWavpackEnc::bitrate</NAME>
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<TYPE>guint</TYPE>
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<RANGE><= 9600000</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Bitrate</NICK>
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<BLURB>Try to encode with this average bitrate (bits/sec). This enables lossy encoding, values smaller than 24000 disable it again.</BLURB>
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<DEFAULT>0</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstWavpackEnc::bits-per-sample</NAME>
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<TYPE>gdouble</TYPE>
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<RANGE>[0,24]</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Bits per sample</NICK>
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<BLURB>Try to encode with this amount of bits per sample. This enables lossy encoding, values smaller than 2.0 disable it again.</BLURB>
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<DEFAULT>0</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstWavpackEnc::correction-mode</NAME>
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<TYPE>GstWavpackEncCorrectionMode</TYPE>
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<RANGE></RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Correction stream mode</NICK>
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<BLURB>Use this mode for the correction stream. Only works in lossy mode!.</BLURB>
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<DEFAULT>Create no correction file</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstWavpackEnc::extra-processing</NAME>
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<TYPE>guint</TYPE>
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<RANGE><= 6</RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Extra processing</NICK>
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<BLURB>Use better but slower filters for better compression/quality.</BLURB>
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<DEFAULT>0</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstWavpackEnc::joint-stereo-mode</NAME>
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<TYPE>GstWavpackEncJSMode</TYPE>
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<RANGE></RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Joint-Stereo mode</NICK>
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<BLURB>Use this joint-stereo mode.</BLURB>
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<DEFAULT>auto</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstWavpackEnc::md5</NAME>
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<TYPE>gboolean</TYPE>
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<RANGE></RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>MD5</NICK>
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<BLURB>Store MD5 hash of raw samples within the file.</BLURB>
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<DEFAULT>FALSE</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstWavpackEnc::mode</NAME>
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<TYPE>GstWavpackEncMode</TYPE>
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<RANGE></RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Encoding mode</NICK>
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<BLURB>Speed versus compression tradeoff.</BLURB>
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<DEFAULT>Normal Compression</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstRtpMP2TDepay::skip-first-bytes</NAME>
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<TYPE>guint</TYPE>
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<RANGE></RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Skip first bytes</NICK>
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<BLURB>The amount of bytes that need to be skipped at the beginning of the payload.</BLURB>
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<DEFAULT>0</DEFAULT>
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</ARG>
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<ARG>
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<NAME>GstRtpH263PPay::fragmentation-mode</NAME>
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<TYPE>GstFragmentationMode</TYPE>
|
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<RANGE></RANGE>
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<FLAGS>rw</FLAGS>
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<NICK>Fragmentation Mode</NICK>
|
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<BLURB>Packet Fragmentation Mode.</BLURB>
|
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<DEFAULT>Normal</DEFAULT>
|
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</ARG>
|
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|
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<ARG>
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<NAME>GstGamma::gamma</NAME>
|
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<TYPE>gdouble</TYPE>
|
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<RANGE>[0.01,10]</RANGE>
|
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<FLAGS>rw</FLAGS>
|
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<NICK>Gamma</NICK>
|
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<BLURB>gamma.</BLURB>
|
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<DEFAULT>1</DEFAULT>
|
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</ARG>
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|
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|
|
|
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|
|
@ -35,7 +35,7 @@ GObject
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GstBaseAudioSrc
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GstAudioSrc
|
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GstOssSrc
|
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GstOsxAudioSrc
|
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GstOsxAudioSrc
|
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GstUDPSrc
|
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GstPngDec
|
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GstPngEnc
|
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|
|
@ -72,6 +72,9 @@ GObject
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GstOssMixerElement
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GstWavParse
|
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GstWavEnc
|
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GstWavpackParse
|
||||
GstWavpackDec
|
||||
GstWavpackEnc
|
||||
GstVideoMixer
|
||||
GstID3Demux
|
||||
GstSMPTE
|
||||
|
|
|
|||
|
|
@ -262,3 +262,19 @@ GstElement *arg1
|
|||
GstDV1394Src *gstdv1394src
|
||||
</SIGNAL>
|
||||
|
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<SIGNAL>
|
||||
<NAME>GstRTPDec::clear-pt-map</NAME>
|
||||
<RETURNS>void</RETURNS>
|
||||
<FLAGS>l</FLAGS>
|
||||
GstRTPDec *gstrtpdec
|
||||
</SIGNAL>
|
||||
|
||||
<SIGNAL>
|
||||
<NAME>GstRTPDec::request-pt-map</NAME>
|
||||
<RETURNS>GstCaps*</RETURNS>
|
||||
<FLAGS>l</FLAGS>
|
||||
GstRTPDec *gstrtpdec
|
||||
guint arg1
|
||||
guint arg2
|
||||
</SIGNAL>
|
||||
|
||||
|
|
|
|||
|
|
@ -14,14 +14,16 @@
|
|||
<longname>Auto audio sink</longname>
|
||||
<class>Sink/Audio</class>
|
||||
<description>Wrapper audio sink for automatically detected audio sink</description>
|
||||
<author>Ronald Bultje <rbultje@ronald.bitfreak.net></author>
|
||||
<author>Ronald Bultje <rbultje@ronald.bitfreak.net>
|
||||
Jan Schmidt <thaytan@noraisin.net</author>
|
||||
</element>
|
||||
<element>
|
||||
<name>autovideosink</name>
|
||||
<longname>Auto video sink</longname>
|
||||
<class>Sink/Video</class>
|
||||
<description>Wrapper video sink for automatically detected video sink</description>
|
||||
<author>Ronald Bultje <rbultje@ronald.bitfreak.net></author>
|
||||
<author>Ronald Bultje <rbultje@ronald.bitfreak.net>
|
||||
Jan Schmidt <thaytan@noraisin.net</author>
|
||||
</element>
|
||||
</elements>
|
||||
</plugin>
|
||||
|
|
@ -14,7 +14,8 @@
|
|||
<longname>GConf audio sink</longname>
|
||||
<class>Sink/Audio</class>
|
||||
<description>Audio sink embedding the GConf-settings for audio output</description>
|
||||
<author>Ronald Bultje <rbultje@ronald.bitfreak.net></author>
|
||||
<author>Ronald Bultje <rbultje@ronald.bitfreak.net>
|
||||
Jan Schmidt <thaytan@mad.scientist.com></author>
|
||||
</element>
|
||||
<element>
|
||||
<name>gconfaudiosrc</name>
|
||||
|
|
|
|||
File diff suppressed because it is too large
Load diff
|
|
@ -41,14 +41,14 @@
|
|||
<name>rtpamrdepay</name>
|
||||
<longname>RTP packet depayloader</longname>
|
||||
<class>Codec/Depayloader/Network</class>
|
||||
<description>Extracts AMR audio from RTP packets (RFC 3267)</description>
|
||||
<description>Extracts AMR or AMR-WB audio from RTP packets (RFC 3267)</description>
|
||||
<author>Wim Taymans <wim@fluendo.com></author>
|
||||
</element>
|
||||
<element>
|
||||
<name>rtpamrpay</name>
|
||||
<longname>RTP packet payloader</longname>
|
||||
<class>Codec/Payloader/Network</class>
|
||||
<description>Payload-encode AMR audio into RTP packets (RFC 3267)</description>
|
||||
<description>Payload-encode AMR or AMR-WB audio into RTP packets (RFC 3267)</description>
|
||||
<author>Wim Taymans <wim@fluendo.com></author>
|
||||
</element>
|
||||
<element>
|
||||
|
|
@ -100,6 +100,13 @@
|
|||
<description>Extracts H264 video from RTP packets (RFC 3984)</description>
|
||||
<author>Wim Taymans <wim@fluendo.com></author>
|
||||
</element>
|
||||
<element>
|
||||
<name>rtph264pay</name>
|
||||
<longname>RTP packet payloader</longname>
|
||||
<class>Codec/Payloader/Network</class>
|
||||
<description>Payload-encode H264 video into RTP packets (RFC 3984)</description>
|
||||
<author>Laurent Glayal <spglegle@yahoo.fr></author>
|
||||
</element>
|
||||
<element>
|
||||
<name>rtpilbcdepay</name>
|
||||
<longname>RTP iLBC packet depayloader</longname>
|
||||
|
|
@ -119,8 +126,23 @@
|
|||
<longname>RTP packet depayloader</longname>
|
||||
<class>Codec/Depayloader/Network</class>
|
||||
<description>Extracts MPEG2 TS from RTP packets (RFC 2250)</description>
|
||||
<author>Wim Taymans <wim@fluendo.com>
|
||||
Thijs Vermeir <thijs.vermeir@barco.com></author>
|
||||
</element>
|
||||
<element>
|
||||
<name>rtpmp2tpay</name>
|
||||
<longname>RTP MP2T audio payloader</longname>
|
||||
<class>Codec/Payloader/Network</class>
|
||||
<description>Payload-encodes MPEG2 TS into RTP packets (RFC 2250)</description>
|
||||
<author>Wim Taymans <wim@fluendo.com></author>
|
||||
</element>
|
||||
<element>
|
||||
<name>rtpmp4adepay</name>
|
||||
<longname>RTP packet parser</longname>
|
||||
<class>Codec/Depayloader/Network</class>
|
||||
<description>Extracts MPEG4 audio from RTP packets (RFC 3016)</description>
|
||||
<author>Nokia Corporation (contact <stefan.kost@nokia.com>), Wim Taymans <wim@fluendo.com></author>
|
||||
</element>
|
||||
<element>
|
||||
<name>rtpmp4gdepay</name>
|
||||
<longname>RTP packet depayloader</longname>
|
||||
|
|
@ -144,9 +166,9 @@
|
|||
</element>
|
||||
<element>
|
||||
<name>rtpmp4vpay</name>
|
||||
<longname>RTP packet payloader</longname>
|
||||
<longname>RTP MPEG-4 Video packet payloader</longname>
|
||||
<class>Codec/Payloader/Network</class>
|
||||
<description>Payload MPEG4 video as RTP packets (RFC 3016)</description>
|
||||
<description>Payload MPEG-4 video as RTP packets (RFC 3016)</description>
|
||||
<author>Wim Taymans <wim@fluendo.com></author>
|
||||
</element>
|
||||
<element>
|
||||
|
|
@ -237,7 +259,7 @@
|
|||
<name>rtpvorbisdepay</name>
|
||||
<longname>RTP packet depayloader</longname>
|
||||
<class>Codec/Depayloader/Network</class>
|
||||
<description>Extracts Vorbis Audio from RTP packets (draft-01 of RFC XXXX)</description>
|
||||
<description>Extracts Vorbis Audio from RTP packets (draft-04 of RFC XXXX)</description>
|
||||
<author>Wim Taymans <wim@fluendo.com></author>
|
||||
</element>
|
||||
<element>
|
||||
|
|
|
|||
34
docs/plugins/inspect/plugin-wavpack.xml
Normal file
34
docs/plugins/inspect/plugin-wavpack.xml
Normal file
|
|
@ -0,0 +1,34 @@
|
|||
<plugin>
|
||||
<name>wavpack</name>
|
||||
<description>Wavpack lossless/lossy audio format handling</description>
|
||||
<filename>../../ext/wavpack/.libs/libgstwavpack.so</filename>
|
||||
<basename>libgstwavpack.so</basename>
|
||||
<version>0.10.5.1</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-good</source>
|
||||
<package>GStreamer Good Plug-ins CVS/prerelease</package>
|
||||
<origin>Unknown package origin</origin>
|
||||
<elements>
|
||||
<element>
|
||||
<name>wavpackdec</name>
|
||||
<longname>Wavpack audio decoder</longname>
|
||||
<class>Codec/Decoder/Audio</class>
|
||||
<description>Decodes Wavpack audio data</description>
|
||||
<author>Arwed v. Merkatz <v.merkatz@gmx.net>, Sebastian Dröge <slomo@circular-chaos.org></author>
|
||||
</element>
|
||||
<element>
|
||||
<name>wavpackenc</name>
|
||||
<longname>Wavpack audio encoder</longname>
|
||||
<class>Codec/Encoder/Audio</class>
|
||||
<description>Encodes audio with the Wavpack lossless/lossy audio codec</description>
|
||||
<author>Sebastian Dröge <slomo@circular-chaos.org></author>
|
||||
</element>
|
||||
<element>
|
||||
<name>wavpackparse</name>
|
||||
<longname>Wavpack parser</longname>
|
||||
<class>Codec/Demuxer/Audio</class>
|
||||
<description>Parses Wavpack files</description>
|
||||
<author>Arwed v. Merkatz <v.merkatz@gmx.net>, Sebastian Dröge <slomo@circular-chaos.org></author>
|
||||
</element>
|
||||
</elements>
|
||||
</plugin>
|
||||
|
|
@ -118,6 +118,12 @@ else
|
|||
TAGLIB_DIR =
|
||||
endif
|
||||
|
||||
if USE_WAVPACK
|
||||
WAVPACK_DIR=wavpack
|
||||
else
|
||||
WAVPACK_DIR=
|
||||
endif
|
||||
|
||||
SUBDIRS = \
|
||||
$(AALIB_DIR) \
|
||||
$(ANNODEX_DIR) \
|
||||
|
|
@ -138,7 +144,8 @@ SUBDIRS = \
|
|||
$(MIKMOD_DIR) \
|
||||
$(SHOUT2_DIR) \
|
||||
$(SPEEX_DIR) \
|
||||
$(TAGLIB_DIR)
|
||||
$(TAGLIB_DIR) \
|
||||
$(WAVPACK_DIR)
|
||||
|
||||
DIST_SUBDIRS = \
|
||||
aalib \
|
||||
|
|
@ -158,4 +165,5 @@ DIST_SUBDIRS = \
|
|||
raw1394 \
|
||||
shout2 \
|
||||
speex \
|
||||
taglib
|
||||
taglib \
|
||||
wavpack
|
||||
|
|
|
|||
|
|
@ -40,6 +40,15 @@ else
|
|||
check_taglib =
|
||||
endif
|
||||
|
||||
if USE_WAVPACK
|
||||
check_wavpack = \
|
||||
elements/wavpackparse \
|
||||
elements/wavpackdec \
|
||||
elements/wavpackenc
|
||||
else
|
||||
check_wavpack =
|
||||
endif
|
||||
|
||||
check_PROGRAMS = \
|
||||
generic/states \
|
||||
$(check_annodex) \
|
||||
|
|
@ -55,7 +64,8 @@ check_PROGRAMS = \
|
|||
elements/icydemux \
|
||||
elements/videofilter \
|
||||
$(check_sunaudio) \
|
||||
$(check_taglib)
|
||||
$(check_taglib) \
|
||||
$(check_wavpack)
|
||||
|
||||
VALGRIND_TO_FIX =
|
||||
|
||||
|
|
|
|||
Loading…
Reference in a new issue