net-libs/pjproject: bump to 2.13.1-r1 to add IUSE=srtp

Bump to make linking in libsrtp optional so we can keyword asterisk for
sparc.

There are no known project in gentoo other than asterisk, which don't
care for pjproject having (or not having) srtp.  Thus in order to enable
pjproject to be used on sparc we need to be able to not depend on libsrtp.

Bug: https://bugs.gentoo.org/919424
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
Closes: https://github.com/gentoo/gentoo/pull/34719
Signed-off-by: Sam James <sam@gentoo.org>

3 files changed, 229 insertions, 0 deletions

diff --git a/net-libs/pjproject/files/pjproject-2.13.1-r1-config_site.h b/net-libs/pjproject/files/pjproject-2.13.1-r1-config_site.h
new file mode 100644
index 000000000000..c2dd59149e9f
--- /dev/null
+++ b/net-libs/pjproject/files/pjproject-2.13.1-r1-config_site.h
@@ -0,0 +1,86 @@
+/*
+ * Based off of the Asterisk config_site.h file.
+ *
+ * In general it's the same with some removals due to being ebuild-managed.
+ */
+
+#include <sys/select.h>
+
+#define GENTOO_INVALID	(Gentoo compile failure - please report a bug on bugs.gentoo.org)
+
+/* asterisk_malloc_debug.h is not required ... most of the operations are no-ops regardless
+ * and I can't see why asterisk is looking to compile this directly into pjproject */
+
+/* Ability to change this has ABI implications, force it on */
+/* Can be reconsidered in future:  https://bugs.gentoo.org/680496 */
+#define PJ_HAS_IPV6 1
+
+#define PJ_MAX_HOSTNAME (256)
+#define PJSIP_MAX_URL_SIZE (512)
+#ifdef PJ_HAS_LINUX_EPOLL
+#define PJ_IOQUEUE_MAX_HANDLES	(5000)
+#else
+#define PJ_IOQUEUE_MAX_HANDLES	(FD_SETSIZE)
+#endif
+#define PJ_IOQUEUE_HAS_SAFE_UNREG 1
+#define PJ_IOQUEUE_MAX_EVENTS_IN_SINGLE_POLL (16)
+
+#define PJ_SCANNER_USE_BITWISE	0
+#define PJ_OS_HAS_CHECK_STACK	0
+
+#ifndef PJ_LOG_MAX_LEVEL
+#define PJ_LOG_MAX_LEVEL		6
+#endif
+
+#define PJ_ENABLE_EXTRA_CHECK	1
+#define PJSIP_MAX_TSX_COUNT		((64*1024)-1)
+#define PJSIP_MAX_DIALOG_COUNT	((64*1024)-1)
+#define PJSIP_UDP_SO_SNDBUF_SIZE	(512*1024)
+#define PJSIP_UDP_SO_RCVBUF_SIZE	(512*1024)
+#define PJ_DEBUG			0
+#define PJSIP_SAFE_MODULE		0
+#define PJ_HAS_STRICMP_ALNUM		0
+
+/*
+ * Do not ever enable PJ_HASH_USE_OWN_TOLOWER because the algorithm is
+ * inconsistently used when calculating the hash value and doesn't
+ * convert the same characters as pj_tolower()/tolower().  Thus you
+ * can get different hash values if the string hashed has certain
+ * characters in it.  (ASCII '@', '[', '\\', ']', '^', and '_')
+ */
+#undef PJ_HASH_USE_OWN_TOLOWER
+
+/*
+  It is imperative that PJSIP_UNESCAPE_IN_PLACE remain 0 or undefined.
+  Enabling it will result in SEGFAULTS when URIs containing escape sequences are encountered.
+*/
+#undef PJSIP_UNESCAPE_IN_PLACE
+#define PJSIP_MAX_PKT_LEN			65535
+
+#undef PJ_TODO
+#define PJ_TODO(x)
+
+/* Defaults too low for WebRTC */
+#define PJ_ICE_MAX_CAND 64
+#define PJ_ICE_MAX_CHECKS (PJ_ICE_MAX_CAND * PJ_ICE_MAX_CAND)
+
+/* Increase limits to allow more formats */
+#define	PJMEDIA_MAX_SDP_FMT   64
+#define	PJMEDIA_MAX_SDP_BANDW   4
+#define	PJMEDIA_MAX_SDP_ATTR   (PJMEDIA_MAX_SDP_FMT*3 + 4)
+#define	PJMEDIA_MAX_SDP_MEDIA   16
+
+/*
+ * Turn off the periodic sending of CRLNCRLN.  Default is on (90 seconds),
+ * which conflicts with the global section's keep_alive_interval option in
+ * pjsip.conf.
+ */
+#define PJSIP_TCP_KEEP_ALIVE_INTERVAL	0
+#define PJSIP_TLS_KEEP_ALIVE_INTERVAL	0
+
+#define PJSIP_TSX_UAS_CONTINUE_ON_TP_ERROR 0
+#define PJ_SSL_SOCK_OSSL_USE_THREAD_CB 0
+#define PJSIP_AUTH_ALLOW_MULTIPLE_AUTH_HEADER 1
+
+/* Required to enable things like USE=video. */
+#define PJMEDIA_HAS_VIDEO GENTOO_INVALID
diff --git a/net-libs/pjproject/metadata.xml b/net-libs/pjproject/metadata.xml
index 526020157e7d..b7d5dd9a9604 100644
--- a/net-libs/pjproject/metadata.xml
+++ b/net-libs/pjproject/metadata.xml
@@ -22,6 +22,7 @@
 		<flag name="openh264">Include Open H.264 support in the build</flag>
 		<flag name="resample">Include resampling implementations in the build</flag>
 		<flag name="silk">Include SILK support in the build</flag>
+		<flag name="srtp">Enable support for encrypted voice transmission (secure RTP)</flag>
 		<flag name="v4l2">Include Video4Linux v2 support in the build</flag>
 		<flag name="vpx">Include VP8 and VP9 codec support in the build</flag>
 		<flag name="webrtc">Enable WebRTC support</flag>
diff --git a/net-libs/pjproject/pjproject-2.13.1-r1.ebuild b/net-libs/pjproject/pjproject-2.13.1-r1.ebuild
new file mode 100644
index 000000000000..536c98527bac
--- /dev/null
+++ b/net-libs/pjproject/pjproject-2.13.1-r1.ebuild
@@ -0,0 +1,142 @@
+# Copyright 1999-2024 Gentoo Authors
+# Distributed under the terms of the GNU General Public License v2
+# TODO: Figure out a way to disable SRTP from pjproject entirely.
+EAPI=8
+
+inherit autotools flag-o-matic toolchain-funcs
+
+DESCRIPTION="Open source SIP, Media, and NAT Traversal Library"
+HOMEPAGE="https://github.com/pjsip/pjproject https://www.pjsip.org/"
+SRC_URI="https://github.com/pjsip/${PN}/archive/refs/tags/${PV}.tar.gz -> ${P}.tar.gz"
+KEYWORDS="~amd64 ~arm ~arm64 ~ppc ~ppc64 ~x86"
+
+LICENSE="GPL-2"
+SLOT="0/${PV}"
+
+# g729 not included due to special bcg729 handling.
+CODEC_FLAGS="g711 g722 g7221 gsm ilbc speex l16"
+VIDEO_FLAGS="sdl ffmpeg v4l2 openh264 libyuv vpx"
+SOUND_FLAGS="alsa portaudio"
+IUSE="amr debug epoll examples opus resample silk srtp ssl static-libs webrtc
+	${CODEC_FLAGS} g729
+	${VIDEO_FLAGS}
+	${SOUND_FLAGS}"
+
+RDEPEND="
+	alsa? ( media-libs/alsa-lib )
+	amr? ( media-libs/opencore-amr )
+	ffmpeg? ( media-video/ffmpeg:= )
+	g729? ( media-libs/bcg729 )
+	gsm? ( media-sound/gsm )
+	ilbc? ( media-libs/libilbc )
+	openh264? ( media-libs/openh264 )
+	opus? ( media-libs/opus )
+	portaudio? ( media-libs/portaudio )
+	resample? ( media-libs/libsamplerate )
+	sdl? ( media-libs/libsdl2 )
+	speex? (
+		media-libs/speex
+		media-libs/speexdsp
+	)
+	srtp? ( >=net-libs/libsrtp-2.3.0:= )
+	ssl? ( dev-libs/openssl:0= )
+"
+DEPEND="${RDEPEND}"
+BDEPEND="virtual/pkgconfig"
+
+PATCHES=(
+	"${FILESDIR}/pjproject-2.13-r1-Make-sure-that-NOTIFY-tdata-is-set-before-sending-it_new.patch"
+	"${FILESDIR}/pjproject-2.13.1-fix-ptimesized-wav-input.patch"
+)
+
+src_prepare() {
+	default
+	rm configure || die "Unable to remove unwanted wrapper"
+	mv aconfigure.ac configure.ac || die "Unable to rename configure script source"
+	eautoreconf
+
+	cp "${FILESDIR}/pjproject-2.13.1-r1-config_site.h" "${S}/pjlib/include/pj/config_site.h" \
+		|| die "Unable to create config_site.h"
+}
+
+_pj_enable() {
+	usex "$1" '' "--disable-${2:-$1}"
+}
+
+_pj_get_define() {
+	local r="$(sed -nre "s/^#define[[:space:]]+$1[[:space:]]+//p" "${S}/pjlib/include/pj/config_site.h")"
+	[[ -z "${r}" ]] && die "Unable to fine #define $1 in config_site.h"
+	echo "$r"
+}
+
+_pj_set_define() {
+	local c=$(_pj_get_define "$1")
+	[[ "$c" = "$2" ]] && return 0
+	sed -re "s/^#define[[:space:]]+$1[[:space:]].*/#define $1 $2/" -i "${S}/pjlib/include/pj/config_site.h" \
+		|| die "sed failed updating $1 to $2."
+	[[ "$(_pj_get_define "$1")" != "$2" ]] && die "sed failed to perform update for $1 to $2."
+}
+
+_pj_use_set_define() {
+	_pj_set_define "$2" $(usex "$1" 1 0)
+}
+
+src_configure() {
+	local myconf=()
+	local videnable="--disable-video"
+	local t
+
+	use debug || append-cflags -DNDEBUG=1
+
+	for t in ${CODEC_FLAGS}; do
+		myconf+=( $(_pj_enable ${t} ${t}-codec) )
+	done
+	myconf+=( $(_pj_enable g729 bcg729) )
+
+	for t in ${VIDEO_FLAGS}; do
+		myconf+=( $(_pj_enable ${t}) )
+		use "${t}" && videnable="--enable-video"
+	done
+
+	[ "${videnable}" = "--enable-video" ] && _pj_set_define PJMEDIA_HAS_VIDEO 1 || _pj_set_define PJMEDIA_HAS_VIDEO 0
+
+	LD="$(tc-getCC)" econf \
+		--enable-shared \
+		${videnable} \
+		$(_pj_enable alsa sound) \
+		$(_pj_enable amr opencore-amr) \
+		$(_pj_enable epoll) \
+		$(_pj_enable opus) \
+		$(_pj_enable portaudio ext-sound) \
+		$(_pj_enable resample libsamplerate) \
+		$(_pj_enable resample resample-dll) \
+		$(_pj_enable resample) \
+		$(_pj_enable silk) \
+		$(_pj_enable speex speex-aec) \
+		$(_pj_enable ssl) \
+		$(_pj_enable webrtc libwebrtc) \
+		$(use_with gsm external-gsm) \
+		$(use_with portaudio external-pa) \
+		$(use_with speex external-speex) \
+		$(usex srtp --with-external-srtp --disable-libsrtp) \
+		"${myconf[@]}"
+}
+
+src_compile() {
+	emake dep LD="$(tc-getCC)"
+	emake LD="$(tc-getCC)"
+}
+
+src_install() {
+	default
+
+	newbin pjsip-apps/bin/pjsua-${CHOST} pjsua
+	newbin pjsip-apps/bin/pjsystest-${CHOST} pjsystest
+
+	if use examples; then
+		insinto "/usr/share/doc/${PF}/examples"
+		doins -r pjsip-apps/src/samples
+	fi
+
+	use static-libs || rm "${ED}/usr/$(get_libdir)"/*.a || die "Error removing static archives"
+}