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media-plugins/alsa-plugins: Removed old
Package-Manager: Portage-2.3.89, Repoman-2.3.20 Signed-off-by: Lars Wendler <polynomial-c@gentoo.org>
This commit is contained in:
Lars Wendler
@@ -1,4 +1,3 @@
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DIST alsa-plugins-1.1.1.tar.bz2 366193 BLAKE2B 02892b0c241cf34d3bfce6cf580dbb6bedbce9253c3e57f04d3093a6cbb1d5b5e9b92566945962b875de20ba3e556e9c1fc4bb4e970502e3c03fc821e5e264fe SHA512 9dd277cb5cce144ce4e5e0010cebcba0f8634c783e7bcd80ce6d04b198d6de5150a2764df3087a39a468b5fb7c4578de2e252095568e811dedaa5e7ffdbeb7c9
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DIST alsa-plugins-1.1.8.tar.bz2 375346 BLAKE2B 65c161ee375e7aed85dbe918a39bc2aa4818bbcfbaadf405f5f0170d9944ff55b5a9da3690e3cd78bd7ffff74ecf235f49cfc3a5dfe18b627c5a9cb0dc1eb207 SHA512 bb1a4411921ec332f47b43515f6d3b4c03a01568854950b9b5c3f0d8172d9a0c4a5b3652bc64c62eae5ea3aaa5d95fe3d32c0a213fe99adf490a9807bcb58ecd
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DIST alsa-plugins-1.1.9.tar.bz2 406127 BLAKE2B 10bf3e96289a88edabfa3018d33ab4666982fe409eb41c486e6b9eee0dde87e926f3241c31e240b3bab114c8ab2c6d6bab92dd6192bceb467a01fcb6adc44b8e SHA512 1969dfe8f2bb4283e81a18da73d2f7f86d8788d1a8764d7195b4830b2f4704c208eacb50a1e26e20c9fe57e7d5bb06a004112d84cc86b4682b4180d2fec4d5d9
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DIST alsa-plugins-1.2.1.tar.bz2 406309 BLAKE2B 2ac26ad87fcdcc4452ec32b2b437d3be4308f13ce2a8e24150f319805b6d684edc6be9e51654f27c1a5bbc01a7caa9eb70250be405dc6921ecd93ee417f24cdf SHA512 a49ef9290b66855112e301a5fd5a0932f13a50254e164f3340f2b342a36469532ee783beabfe5e95870dbf4cd81ca1e8d0415d94c8c7ee963133fb3c343721de
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@@ -1,100 +0,0 @@
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# Copyright 1999-2020 Gentoo Authors
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# Distributed under the terms of the GNU General Public License v2
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EAPI=6
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inherit autotools eutils flag-o-matic ltprune multilib multilib-minimal
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DESCRIPTION="ALSA extra plugins"
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HOMEPAGE="http://www.alsa-project.org/"
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SRC_URI="https://www.alsa-project.org/files/pub/plugins/${P}.tar.bz2"
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LICENSE="GPL-2 LGPL-2.1"
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SLOT="0"
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KEYWORDS="~alpha amd64 arm hppa ia64 ppc ppc64 ~sh sparc x86 ~amd64-linux"
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IUSE="debug ffmpeg jack libav libsamplerate pulseaudio speex"
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RDEPEND="
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>=media-libs/alsa-lib-${PV}:=[${MULTILIB_USEDEP}]
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ffmpeg? (
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libav? ( media-video/libav:= )
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!libav? ( media-video/ffmpeg:0= )
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)
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jack? ( virtual/jack[${MULTILIB_USEDEP}] )
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libsamplerate? ( >=media-libs/libsamplerate-0.1.8-r1:=[${MULTILIB_USEDEP}] )
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pulseaudio? ( >=media-sound/pulseaudio-2.1-r1[${MULTILIB_USEDEP}] )
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speex? (
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>=media-libs/speex-1.2.0:=[${MULTILIB_USEDEP}]
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media-libs/speexdsp[${MULTILIB_USEDEP}]
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)
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"
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DEPEND="${RDEPEND}
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virtual/pkgconfig"
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PATCHES=(
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"${FILESDIR}/${PN}-1.0.23-automagic.patch"
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"${FILESDIR}/${PN}-1.0.28-libav10.patch"
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)
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src_prepare() {
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default
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# For some reasons the polyp/pulse plugin does fail with alsaplayer with a
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# failed assert. As the code works just fine with asserts disabled, for now
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# disable them waiting for a better solution.
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sed -i \
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-e '/AM_CFLAGS/s:-Wall:-DNDEBUG -Wall:' \
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pulse/Makefile.am || die
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eautoreconf
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}
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multilib_src_configure() {
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use debug || append-cppflags -DNDEBUG
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local myspeex=no
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use speex && myspeex=lib
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ECONF_SOURCE=${S} \
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econf \
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$(use_enable ffmpeg avcodec) \
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$(use_enable jack) \
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$(use_enable libsamplerate samplerate) \
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$(use_enable pulseaudio) \
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--with-speex=${myspeex}
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}
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multilib_src_install_all() {
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einstalldocs
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cd doc || die
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dodoc upmix.txt vdownmix.txt README-pcm-oss
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use jack && dodoc README-jack
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use libsamplerate && dodoc samplerate.txt
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use ffmpeg && dodoc lavcrate.txt a52.txt
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if use pulseaudio; then
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dodoc README-pulse
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# install ALSA configuration files
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# making PA to be used by alsa clients
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insinto /usr/share/alsa
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doins "${FILESDIR}"/pulse-default.conf
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insinto /usr/share/alsa/alsa.conf.d
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doins "${FILESDIR}"/51-pulseaudio-probe.conf
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# bug #410261, comment 5+
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# seems to work fine without any path
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sed -i \
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-e "s:/usr/lib/alsa-lib/::" \
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"${ED}"/usr/share/alsa/alsa.conf.d/51-pulseaudio-probe.conf || die #410261
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fi
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prune_libtool_files --all
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}
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pkg_postinst() {
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if use pulseaudio; then
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einfo "The PulseAudio device is now set as the default device if the"
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einfo "PulseAudio server is found to be running. Any custom"
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einfo "configuration in /etc/asound.conf or ~/.asoundrc for this"
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einfo "purpose should now be unnecessary."
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fi
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}
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@@ -1,103 +0,0 @@
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# Copyright 1999-2019 Gentoo Authors
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# Distributed under the terms of the GNU General Public License v2
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EAPI=7
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inherit autotools flag-o-matic multilib multilib-minimal
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DESCRIPTION="ALSA extra plugins"
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HOMEPAGE="http://www.alsa-project.org/"
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SRC_URI="https://www.alsa-project.org/files/pub/plugins/${P}.tar.bz2"
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LICENSE="GPL-2 LGPL-2.1"
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SLOT="0"
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KEYWORDS="~alpha ~amd64 ~arm ~arm64 ~hppa ~ia64 ~ppc ~ppc64 ~sh ~sparc ~x86 ~amd64-linux"
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IUSE="arcam_av debug ffmpeg jack libav libsamplerate +mix oss pulseaudio speex +usb_stream"
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RDEPEND="
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>=media-libs/alsa-lib-${PV}:=[${MULTILIB_USEDEP}]
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ffmpeg? (
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libav? ( media-video/libav:= )
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!libav? ( media-video/ffmpeg:0= )
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)
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jack? ( virtual/jack[${MULTILIB_USEDEP}] )
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libsamplerate? ( >=media-libs/libsamplerate-0.1.8-r1:=[${MULTILIB_USEDEP}] )
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pulseaudio? ( >=media-sound/pulseaudio-2.1-r1[${MULTILIB_USEDEP}] )
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speex? (
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>=media-libs/speex-1.2.0:=[${MULTILIB_USEDEP}]
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media-libs/speexdsp[${MULTILIB_USEDEP}]
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)
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"
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DEPEND="${RDEPEND}"
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BDEPEND="virtual/pkgconfig"
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src_prepare() {
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default
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# For some reasons the polyp/pulse plugin does fail with alsaplayer with a
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# failed assert. As the code works just fine with asserts disabled, for now
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# disable them waiting for a better solution.
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sed -i \
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-e '/AM_CFLAGS/s:-Wall:-DNDEBUG -Wall:' \
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pulse/Makefile.am || die
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eautoreconf
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}
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multilib_src_configure() {
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use debug || append-cppflags -DNDEBUG
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local myeconfargs=(
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# default does not contain $prefix: bug #673464
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--with-alsalconfdir="${EPREFIX}"/etc/alsa/conf.d
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--with-speex="$(usex speex lib no)"
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$(use_enable arcam_av arcamav)
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$(use_enable ffmpeg libav)
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$(use_enable jack)
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$(use_enable libsamplerate samplerate)
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$(use_enable mix)
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$(use_enable oss)
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$(use_enable pulseaudio)
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$(use_enable speex speexdsp)
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$(use_enable usb_stream usbstream)
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)
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ECONF_SOURCE="${S}" econf "${myeconfargs[@]}"
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}
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multilib_src_install_all() {
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einstalldocs
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cd doc || die
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dodoc upmix.txt vdownmix.txt README-pcm-oss
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use jack && dodoc README-jack
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use libsamplerate && dodoc samplerate.txt
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use ffmpeg && dodoc lavrate.txt a52.txt
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if use pulseaudio; then
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dodoc README-pulse
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# install ALSA configuration files
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# making PA to be used by alsa clients
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insinto /usr/share/alsa
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doins "${FILESDIR}"/pulse-default.conf
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insinto /usr/share/alsa/alsa.conf.d
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doins "${FILESDIR}"/51-pulseaudio-probe.conf
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# bug #410261, comment 5+
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# seems to work fine without any path
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sed -i \
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-e "s:/usr/lib/alsa-lib/::" \
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"${ED}"/usr/share/alsa/alsa.conf.d/51-pulseaudio-probe.conf || die #410261
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dosym ../../../usr/share/alsa/alsa.conf.d/51-pulseaudio-probe.conf \
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/etc/alsa/conf.d/51-pulseaudio-probe.conf #670960
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fi
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find "${ED}" \( -name '*.a' -o -name '*.la' \) -delete || die
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}
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pkg_postinst() {
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if use pulseaudio; then
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einfo "The PulseAudio device is now set as the default device if the"
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einfo "PulseAudio server is found to be running. Any custom"
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einfo "configuration in /etc/asound.conf or ~/.asoundrc for this"
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einfo "purpose should now be unnecessary."
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fi
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}
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@@ -1,12 +0,0 @@
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diff -uNr alsa-plugins-1.0.23.ORIg//Makefile.am alsa-plugins-1.0.23/Makefile.am
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--- alsa-plugins-1.0.23.ORIg//Makefile.am 2010-04-16 23:38:58.546243512 +0100
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+++ alsa-plugins-1.0.23/Makefile.am 2010-04-16 23:39:20.049278487 +0100
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@@ -17,7 +17,7 @@
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if HAVE_PPH
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SUBDIRS += pph
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endif
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-if HAVE_SPEEXDSP
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+if USE_LIBSPEEX
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SUBDIRS += speex
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endif
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@@ -1,338 +0,0 @@
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https://bugs.gentoo.org/539680
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From: Luca Barbato <lu_zero@gentoo.org>
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Description: lavr: Add a libavresample based rate plugin
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Date: Mon, 14 Apr 2014 10:01:07 +0200
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Provide lavcrate compatibility.
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Index: alsa-plugins-1.0.28/configure.ac
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===================================================================
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--- alsa-plugins-1.0.28.orig/configure.ac
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+++ alsa-plugins-1.0.28/configure.ac
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@@ -66,7 +66,7 @@ if test "$use_maemo_rm" = "yes"; then
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fi
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AC_ARG_ENABLE([avcodec],
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- AS_HELP_STRING([--disable-avcodec], [Don't build plugins depending on avcodec (a52)]))
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+ AS_HELP_STRING([--disable-avcodec], [Do not build plugins depending on avcodec (a52)]))
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if test "x$enable_avcodec" != "xno"; then
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PKG_CHECK_MODULES(AVCODEC, [libavcodec libavutil], [HAVE_AVCODEC=yes], [HAVE_AVCODEC=no])
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@@ -101,6 +101,10 @@ if test $HAVE_AVCODEC = yes; then
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if test -z "$AVCODEC_HEADER"; then
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HAVE_AVCODEC=no
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fi
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+ SAVE_LIBS=$LIBS
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+ LIBS="$LIBS $AVCODEC_LIBS"
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+ AC_CHECK_FUNCS([av_resample_init])
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+ LIBS=$SAVE_LIBS
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fi
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AM_CONDITIONAL(HAVE_AVCODEC, test x$HAVE_AVCODEC = xyes)
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@@ -108,6 +112,18 @@ AC_SUBST(AVCODEC_CFLAGS)
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AC_SUBST(AVCODEC_LIBS)
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AC_SUBST(AVCODEC_HEADER)
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+AC_ARG_ENABLE([avresample],
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+ AS_HELP_STRING([--disable-avresample], [Do not build plugins depending on avcodec (lavrate)]))
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+
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+if test "x$enable_avresample" != "xno"; then
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+ PKG_CHECK_MODULES(AVRESAMPLE, [libavresample libavutil], [HAVE_AVRESAMPLE=yes], [HAVE_AVRESAMPLE=no])
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+fi
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+
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+AM_CONDITIONAL(HAVE_AVRESAMPLE, test x$HAVE_AVCODEC = xyes)
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+AC_SUBST(AVRESAMPLE_CFLAGS)
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+AC_SUBST(AVRESAMPLE_LIBS)
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+AC_SUBST(AVRESAMPLE_HEADER)
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+
|
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PKG_CHECK_MODULES(speexdsp, [speexdsp >= 1.2], [HAVE_SPEEXDSP="yes"], [HAVE_SPEEXDSP=""])
|
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AM_CONDITIONAL(HAVE_SPEEXDSP, test "$HAVE_SPEEXDSP" = "yes")
|
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|
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@@ -181,7 +197,7 @@ AC_OUTPUT([
|
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mix/Makefile
|
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rate/Makefile
|
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a52/Makefile
|
||||
- rate-lavc/Makefile
|
||||
+ rate-lavr/Makefile
|
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maemo/Makefile
|
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doc/Makefile
|
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usb_stream/Makefile
|
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Index: alsa-plugins-1.0.28/Makefile.am
|
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===================================================================
|
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--- alsa-plugins-1.0.28.orig/Makefile.am
|
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+++ alsa-plugins-1.0.28/Makefile.am
|
||||
@@ -9,8 +9,14 @@ if HAVE_SAMPLERATE
|
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SUBDIRS += rate
|
||||
endif
|
||||
if HAVE_AVCODEC
|
||||
+SUBDIRS += a52
|
||||
+if !HAVE_AVRESAMPLE
|
||||
SUBDIRS += a52 rate-lavc
|
||||
endif
|
||||
+endif
|
||||
+if HAVE_AVRESAMPLE
|
||||
+SUBDIRS += rate-lavr
|
||||
+endif
|
||||
if HAVE_MAEMO_PLUGIN
|
||||
SUBDIRS += maemo
|
||||
endif
|
||||
Index: alsa-plugins-1.0.28/rate-lavr/Makefile.am
|
||||
===================================================================
|
||||
--- /dev/null
|
||||
+++ alsa-plugins-1.0.28/rate-lavr/Makefile.am
|
||||
@@ -0,0 +1,22 @@
|
||||
+asound_module_rate_lavr_LTLIBRARIES = libasound_module_rate_lavr.la
|
||||
+
|
||||
+asound_module_rate_lavrdir = @ALSA_PLUGIN_DIR@
|
||||
+
|
||||
+AM_CFLAGS = -Wall -g @ALSA_CFLAGS@ @AVRESAMPLE_CFLAGS@
|
||||
+AM_LDFLAGS = -module -avoid-version -export-dynamic -no-undefined $(LDFLAGS_NOUNDEFINED)
|
||||
+
|
||||
+libasound_module_rate_lavr_la_SOURCES = rate_lavr.c
|
||||
+libasound_module_rate_lavr_la_LIBADD = @ALSA_LIBS@ @AVRESAMPLE_LIBS@
|
||||
+
|
||||
+
|
||||
+install-exec-hook:
|
||||
+ rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate*.so
|
||||
+ $(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate.so
|
||||
+ $(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_higher.so
|
||||
+ $(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_high.so
|
||||
+ $(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_fast.so
|
||||
+ $(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_faster.so
|
||||
+
|
||||
+uninstall-hook:
|
||||
+ rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate*.so
|
||||
+ rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavr*.so
|
||||
Index: alsa-plugins-1.0.28/rate-lavr/rate_lavr.c
|
||||
===================================================================
|
||||
--- /dev/null
|
||||
+++ alsa-plugins-1.0.28/rate-lavr/rate_lavr.c
|
||||
@@ -0,0 +1,227 @@
|
||||
+/*
|
||||
+ * Rate converter plugin using libavresample
|
||||
+ * Copyright (c) 2014 by Anton Khirnov
|
||||
+ *
|
||||
+ * This library is free software; you can redistribute it and/or
|
||||
+ * modify it under the terms of the GNU Lesser General Public
|
||||
+ * License as published by the Free Software Foundation; either
|
||||
+ * version 2.1 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
|
||||
+ * Lesser General Public License for more details.
|
||||
+ */
|
||||
+
|
||||
+#include <stdio.h>
|
||||
+#include <alsa/asoundlib.h>
|
||||
+#include <alsa/pcm_rate.h>
|
||||
+
|
||||
+#include <libavresample/avresample.h>
|
||||
+#include <libavutil/channel_layout.h>
|
||||
+#include <libavutil/opt.h>
|
||||
+#include <libavutil/mathematics.h>
|
||||
+#include <libavutil/samplefmt.h>
|
||||
+
|
||||
+
|
||||
+static int filter_size = 16;
|
||||
+static int phase_shift = 10; /* auto-adjusts */
|
||||
+static double cutoff = 0; /* auto-adjusts */
|
||||
+
|
||||
+struct rate_src {
|
||||
+ AVAudioResampleContext *avr;
|
||||
+
|
||||
+ int in_rate;
|
||||
+ int out_rate;
|
||||
+ unsigned int channels;
|
||||
+};
|
||||
+
|
||||
+static snd_pcm_uframes_t input_frames(void *obj, snd_pcm_uframes_t frames)
|
||||
+{
|
||||
+ return frames;
|
||||
+}
|
||||
+
|
||||
+static snd_pcm_uframes_t output_frames(void *obj, snd_pcm_uframes_t frames)
|
||||
+{
|
||||
+ return frames;
|
||||
+}
|
||||
+
|
||||
+static void pcm_src_free(void *obj)
|
||||
+{
|
||||
+ struct rate_src *rate = obj;
|
||||
+ avresample_free(&rate->avr);
|
||||
+}
|
||||
+
|
||||
+static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info)
|
||||
+{
|
||||
+ struct rate_src *rate = obj;
|
||||
+ int i, ir, or;
|
||||
+
|
||||
+ if (!rate->avr || rate->channels != info->channels) {
|
||||
+ int ret;
|
||||
+
|
||||
+ pcm_src_free(rate);
|
||||
+ rate->channels = info->channels;
|
||||
+ ir = rate->in_rate = info->in.rate;
|
||||
+ or = rate->out_rate = info->out.rate;
|
||||
+ i = av_gcd(or, ir);
|
||||
+ if (or > ir) {
|
||||
+ phase_shift = or/i;
|
||||
+ } else {
|
||||
+ phase_shift = ir/i;
|
||||
+ }
|
||||
+ if (cutoff <= 0.0) {
|
||||
+ cutoff = 1.0 - 1.0/filter_size;
|
||||
+ if (cutoff < 0.80)
|
||||
+ cutoff = 0.80;
|
||||
+ }
|
||||
+
|
||||
+ rate->avr = avresample_alloc_context();
|
||||
+ if (!rate->avr)
|
||||
+ return -ENOMEM;
|
||||
+
|
||||
+ av_opt_set_int(rate->avr, "in_sample_rate", info->in.rate, 0);
|
||||
+ av_opt_set_int(rate->avr, "out_sample_rate", info->out.rate, 0);
|
||||
+ av_opt_set_int(rate->avr, "in_sample_format", AV_SAMPLE_FMT_S16, 0);
|
||||
+ av_opt_set_int(rate->avr, "out_sample_format", AV_SAMPLE_FMT_S16, 0);
|
||||
+ av_opt_set_int(rate->avr, "in_channel_layout", av_get_default_channel_layout(rate->channels), 0);
|
||||
+ av_opt_set_int(rate->avr, "out_channel_layout", av_get_default_channel_layout(rate->channels), 0);
|
||||
+
|
||||
+ av_opt_set_int(rate->avr, "filter_size", filter_size, 0);
|
||||
+ av_opt_set_int(rate->avr, "phase_shift", phase_shift, 0);
|
||||
+ av_opt_set_double(rate->avr, "cutoff", cutoff, 0);
|
||||
+
|
||||
+ ret = avresample_open(rate->avr);
|
||||
+ if (ret < 0) {
|
||||
+ avresample_free(&rate->avr);
|
||||
+ return -EINVAL;
|
||||
+ }
|
||||
+ }
|
||||
+
|
||||
+ return 0;
|
||||
+}
|
||||
+
|
||||
+static int pcm_src_adjust_pitch(void *obj, snd_pcm_rate_info_t *info)
|
||||
+{
|
||||
+ struct rate_src *rate = obj;
|
||||
+
|
||||
+ if (info->out.rate != rate->out_rate || info->in.rate != rate->in_rate)
|
||||
+ pcm_src_init(obj, info);
|
||||
+ return 0;
|
||||
+}
|
||||
+
|
||||
+static void pcm_src_reset(void *obj)
|
||||
+{
|
||||
+ struct rate_src *rate = obj;
|
||||
+
|
||||
+ if (rate->avr) {
|
||||
+ avresample_close(rate->avr);
|
||||
+ avresample_open(rate->avr);
|
||||
+ }
|
||||
+}
|
||||
+
|
||||
+static void pcm_src_convert_s16(void *obj, int16_t *dst, unsigned int
|
||||
+ dst_frames, const int16_t *src, unsigned int src_frames)
|
||||
+{
|
||||
+ struct rate_src *rate = obj;
|
||||
+ int consumed = 0, chans=rate->channels, ret=0, i;
|
||||
+ int total_in = avresample_get_delay(rate->avr) + src_frames;
|
||||
+
|
||||
+ ret = avresample_convert(rate->avr, &dst, dst_frames * chans * 2, dst_frames,
|
||||
+ &src, src_frames * chans * 2, src_frames);
|
||||
+
|
||||
+ avresample_set_compensation(rate->avr,
|
||||
+ total_in - src_frames > filter_size ? 0 : 1, src_frames);
|
||||
+}
|
||||
+
|
||||
+static void pcm_src_close(void *obj)
|
||||
+{
|
||||
+ pcm_src_free(obj);
|
||||
+}
|
||||
+
|
||||
+#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002
|
||||
+static int get_supported_rates(void *obj, unsigned int *rate_min,
|
||||
+ unsigned int *rate_max)
|
||||
+{
|
||||
+ *rate_min = *rate_max = 0; /* both unlimited */
|
||||
+ return 0;
|
||||
+}
|
||||
+
|
||||
+static void dump(void *obj, snd_output_t *out)
|
||||
+{
|
||||
+ snd_output_printf(out, "Converter: libavr\n");
|
||||
+}
|
||||
+#endif
|
||||
+
|
||||
+static snd_pcm_rate_ops_t pcm_src_ops = {
|
||||
+ .close = pcm_src_close,
|
||||
+ .init = pcm_src_init,
|
||||
+ .free = pcm_src_free,
|
||||
+ .adjust_pitch = pcm_src_adjust_pitch,
|
||||
+ .convert_s16 = pcm_src_convert_s16,
|
||||
+ .input_frames = input_frames,
|
||||
+ .output_frames = output_frames,
|
||||
+#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002
|
||||
+ .version = SND_PCM_RATE_PLUGIN_VERSION,
|
||||
+ .get_supported_rates = get_supported_rates,
|
||||
+ .dump = dump,
|
||||
+#endif
|
||||
+};
|
||||
+
|
||||
+int pcm_src_open(unsigned int version, void **objp, snd_pcm_rate_ops_t *ops)
|
||||
+
|
||||
+{
|
||||
+ struct rate_src *rate;
|
||||
+
|
||||
+#if SND_PCM_RATE_PLUGIN_VERSION < 0x010002
|
||||
+ if (version != SND_PCM_RATE_PLUGIN_VERSION) {
|
||||
+ fprintf(stderr, "Invalid rate plugin version %x\n", version);
|
||||
+ return -EINVAL;
|
||||
+ }
|
||||
+#endif
|
||||
+ rate = calloc(1, sizeof(*rate));
|
||||
+ if (!rate)
|
||||
+ return -ENOMEM;
|
||||
+
|
||||
+ *objp = rate;
|
||||
+ rate->avr = NULL;
|
||||
+#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002
|
||||
+ if (version == 0x010001)
|
||||
+ memcpy(ops, &pcm_src_ops, sizeof(snd_pcm_rate_old_ops_t));
|
||||
+ else
|
||||
+#endif
|
||||
+ *ops = pcm_src_ops;
|
||||
+ return 0;
|
||||
+}
|
||||
+
|
||||
+int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate)(unsigned int version, void **objp,
|
||||
+ snd_pcm_rate_ops_t *ops)
|
||||
+{
|
||||
+ return pcm_src_open(version, objp, ops);
|
||||
+}
|
||||
+int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate_higher)(unsigned int version,
|
||||
+ void **objp, snd_pcm_rate_ops_t *ops)
|
||||
+{
|
||||
+ filter_size = 64;
|
||||
+ return pcm_src_open(version, objp, ops);
|
||||
+}
|
||||
+int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate_high)(unsigned int version,
|
||||
+ void **objp, snd_pcm_rate_ops_t *ops)
|
||||
+{
|
||||
+ filter_size = 32;
|
||||
+ return pcm_src_open(version, objp, ops);
|
||||
+}
|
||||
+int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate_fast)(unsigned int version,
|
||||
+ void **objp, snd_pcm_rate_ops_t *ops)
|
||||
+{
|
||||
+ filter_size = 8;
|
||||
+ return pcm_src_open(version, objp, ops);
|
||||
+}
|
||||
+int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate_faster)(unsigned int version,
|
||||
+ void **objp, snd_pcm_rate_ops_t *ops)
|
||||
+{
|
||||
+ filter_size = 4;
|
||||
+ return pcm_src_open(version, objp, ops);
|
||||
+}
|
||||
+
|
||||
+
|
||||
Reference in New Issue
Block a user