diff --git a/daemon/configure.ac b/daemon/configure.ac
index c53f29b53338c831a910c682705dccb725d6029f..bcb9cd866365eada3b3b20f09bec4779fffa9dde 100644
--- a/daemon/configure.ac
+++ b/daemon/configure.ac
@@ -44,6 +44,8 @@ case "${host_os}" in
SYS=linux
# Necessary for linking .a to a .so
LDFLAGS+=" -Wl,-Bsymbolic"
+ # Needed for plugin system
+ LDFLAGS+=" -ldl"
;;
darwin*)
SYS=darwin
@@ -343,6 +345,18 @@ AS_IF([test "x$with_dbus" = "xyes"], [
AM_CONDITIONAL(RING_DBUS, false));
+dnl Check for libav
+PKG_CHECK_MODULES(LIBAVCODEC, libavcodec >= 53.5.0,, AC_MSG_ERROR([Missing libavcodec development files]))
+LIBAVCODEC_CFLAGS="${LIBAVCODEC_CFLAGS} -D__STDC_CONSTANT_MACROS"
+
+PKG_CHECK_MODULES(LIBAVFORMAT, libavformat >= 54.20.3,, AC_MSG_ERROR([Missing libavformat development files]))
+
+PKG_CHECK_MODULES(LIBSWSCALE, libswscale >= 1.1.0,, AC_MSG_ERROR([Missing libswscale development files]))
+
+PKG_CHECK_MODULES(LIBAVDEVICE, libavdevice >= 53.0.0,, AC_MSG_ERROR([Missing libavdevice development files]))
+
+PKG_CHECK_MODULES(LIBAVUTIL, libavutil >= 51.0.0,, AC_MSG_ERROR([Missing libavutil development files]))
+
dnl Video is default-enabled
AC_ARG_ENABLE([video], AS_HELP_STRING([--disable-video], [Disable video]))
@@ -350,30 +364,12 @@ AS_IF([test "x$enable_video" != "xno"],
[
AC_DEFINE(RING_VIDEO, [], [Video support enabled])
AM_CONDITIONAL(RING_VIDEO, true)
-
- PKG_CHECK_MODULES(LIBAVCODEC, libavcodec >= 53.5.0,, AC_MSG_ERROR([Missing libavcodec development files]))
- LIBAVCODEC_CFLAGS="${LIBAVCODEC_CFLAGS} -D__STDC_CONSTANT_MACROS"
-
- PKG_CHECK_MODULES(LIBAVFORMAT, libavformat >= 54.20.3,, AC_MSG_ERROR([Missing libavformat development files]))
-
- PKG_CHECK_MODULES(LIBSWSCALE, libswscale >= 1.1.0,, AC_MSG_ERROR([Missing libswscale development files]))
-
- PKG_CHECK_MODULES(LIBAVDEVICE, libavdevice >= 53.0.0,, AC_MSG_ERROR([Missing libavdevice development files]))
-
- PKG_CHECK_MODULES(LIBAVUTIL, libavutil >= 51.0.0,, AC_MSG_ERROR([Missing libavutil development files]))
-
PKG_CHECK_MODULES(UDEV, libudev,, AC_MSG_ERROR([Missing libudev development files]))
],
[
AM_CONDITIONAL(RING_VIDEO, false)
- AC_DEFINE_UNQUOTED([USE_CCRTP], 1, [Use ccrtp instead of libavformat])
]);
-LIBCCRTP_MIN_VERSION=1.3.0
-PKG_CHECK_MODULES([CCRTP], [libccrtp] >= ${LIBCCRTP_MIN_VERSION},, [
- PKG_CHECK_MODULES([CCRTP], [libccrtp1] >= ${LIBCCRTP_MIN_VERSION},, AC_MSG_ERROR([Missing ccrtp development files]))
- ])
-
dnl TLS support is enabled if it's installed and up to date
AC_ARG_WITH([tls],
[AS_HELP_STRING([--with-tls],
@@ -388,25 +384,6 @@ AS_CASE(["$with_tls"],
AC_DEFINE_UNQUOTED([HAVE_TLS], `if test $HAVE_GNUTLS -eq 1; then echo 1; else echo 0; fi`, [Define if you have tls support])
AM_CONDITIONAL(BUILD_TLS, test "$HAVE_GNUTLS" -eq 1)
-# ZRTP
-# required dependency(ies): libzrtp
-dnl Check for libzrtpcpp, a ccRTP extension providing zrtp key exchange
-AC_ARG_WITH([zrtp],
- [AS_HELP_STRING([--without-zrtp], [disable support for zrtp])],
- [],
- [with_zrtp=no])
-AS_IF([test "x$with_zrtp" = "xyes"], [
- PKG_CHECK_MODULES([ZRTPCPP], libzrtpcpp >= 2.3.0, [with_zrtp_configure=yes],
- dnl Fallback to older version
- [PKG_CHECK_MODULES([ZRTPCPP], libzrtpcpp >= 1.3.0, [with_zrtp_configure=no],
- AC_MSG_ERROR([Missing zrtp development files]))])
- ]);
-
-dnl This is a workaround until libzrtpcpp gets a version macro
-AC_DEFINE_UNQUOTED([HAVE_ZRTP], `if test "x$with_zrtp" = "xyes"; then echo 1; else echo 0; fi`, [Define if you have zrtp support])
-AC_DEFINE_UNQUOTED([HAVE_ZRTP_CONFIGURE], `if test "x$with_zrtp_configure" = "xyes"; then echo 1; else echo 0; fi`, [Define if you have zrtp >= 2.3.0])
-AM_CONDITIONAL(BUILD_ZRTP, test "x$with_zrtp" = "xyes" )
-
# Instant Messaging
# required dependency(ies): libxpat
AC_ARG_WITH([instant_messaging],
@@ -598,22 +575,23 @@ AC_CONFIG_FILES([Makefile \
src/im/Makefile \
src/iax/Makefile \
src/ringdht/Makefile \
- src/audio/Makefile \
- src/audio/audiortp/Makefile \
- src/audio/pulseaudio/Makefile \
- src/audio/alsa/Makefile \
- src/audio/opensl/Makefile \
- src/audio/jack/Makefile \
- src/audio/coreaudio/Makefile \
- src/audio/sound/Makefile \
- src/audio/codecs/Makefile \
+ src/media/Makefile \
+ src/media/audio/Makefile \
+ src/media/audio/audiortp/Makefile \
+ src/media/audio/pulseaudio/Makefile \
+ src/media/audio/alsa/Makefile \
+ src/media/audio/opensl/Makefile \
+ src/media/audio/jack/Makefile \
+ src/media/audio/coreaudio/Makefile \
+ src/media/audio/sound/Makefile \
+ src/media/audio/codecs/Makefile \
src/config/Makefile \
src/client/Makefile \
src/hooks/Makefile \
src/history/Makefile \
- src/video/Makefile \
- src/video/v4l2/Makefile \
- src/video/test/Makefile \
+ src/media/video/Makefile \
+ src/media/video/v4l2/Makefile \
+ src/media/video/test/Makefile \
test/Makefile \
ringtones/Makefile \
man/Makefile \
diff --git a/daemon/contrib/src/ccrtp/SHA512SUMS b/daemon/contrib/src/ccrtp/SHA512SUMS
deleted file mode 100644
index 9701adead230a13e77e2ed2bf2053c947cbd440e..0000000000000000000000000000000000000000
--- a/daemon/contrib/src/ccrtp/SHA512SUMS
+++ /dev/null
@@ -1 +0,0 @@
-8eb07e2119903bf52dfcc6a7932843bdf14ef686302f333844713c48ba723b5c0211c4b893f29024d4875c3f747fe94eaeadc8a9a4b9993b9766a1373f486062 ccrtp-6d4f1b9ebe9a9b93ec45927b7d8bcda497cc19f6.tar.gz
diff --git a/daemon/contrib/src/ccrtp/rules.mak b/daemon/contrib/src/ccrtp/rules.mak
deleted file mode 100644
index 73de7b448680561767d229d53bf76143b3f1d52e..0000000000000000000000000000000000000000
--- a/daemon/contrib/src/ccrtp/rules.mak
+++ /dev/null
@@ -1,27 +0,0 @@
-# CCRTP
-# FIXME: switch to next release when it's out
-CCRTP_VERSION := 6d4f1b9ebe9a9b93ec45927b7d8bcda497cc19f6
-CCRTP_URL := $(GNUTELEPHONY)/ccrtp/archive/$(CCRTP_VERSION).tar.gz
-
-PKGS += ccrtp
-ifeq ($(call need_pkg,'libccrtp >= 2.0.3'),)
-PKGS_FOUND += ccrtp
-endif
-
-DEPS_ccrtp = ucommon gcrypt gnutls
-
-$(TARBALLS)/ccrtp-$(CCRTP_VERSION).tar.gz:
- $(call download,$(CCRTP_URL))
-
-.sum-ccrtp: ccrtp-$(CCRTP_VERSION).tar.gz
-
-ccrtp: ccrtp-$(CCRTP_VERSION).tar.gz .sum-ccrtp
- $(UNPACK)
- $(UPDATE_AUTOCONFIG) && cd $(UNPACK_DIR)
- $(MOVE)
-
-.ccrtp: ccrtp
- mkdir -p $</m4 && $(RECONF)
- cd $< && $(HOSTVARS) ./configure $(HOSTCONF)
- cd $< && $(MAKE) install
- touch $@
diff --git a/daemon/contrib/src/libav/rules.mak b/daemon/contrib/src/libav/rules.mak
index db962b67486710ffd12440ae0279c4bfc387fb37..42962e38d10a97d14acf55a9df181358aa071311 100644
--- a/daemon/contrib/src/libav/rules.mak
+++ b/daemon/contrib/src/libav/rules.mak
@@ -1,6 +1,8 @@
HASH=HEAD
LIBAV_SNAPURL := http://git.libav.org/?p=libav.git;a=snapshot;h=$(HASH);sf=tgz
+PKGS += libav
+
LIBAVCONF = \
--cc="$(CC)" \
--pkg-config="$(PKG_CONFIG)" \
@@ -104,13 +106,6 @@ else # !Windows
LIBAVCONF += --enable-pthreads
endif
-# Only build on Linux for now, since no one else has implemented video
-ifdef HAVE_LINUX
-ifndef HAVE_ANDROID
-PKGS += libav
-endif
-endif
-
ifeq ($(call need_pkg,"libavcodec >= 53.5.0 libavformat >= 54.20.3 libswscale libavdevice >= 53.0.0 libavutil >= 51.0.0"),)
PKGS_FOUND += libav
endif
diff --git a/daemon/contrib/src/ucommon/rules.mak b/daemon/contrib/src/ucommon/rules.mak
deleted file mode 100644
index 8f86dccece8a56258fbe286c028f39c311f9e83c..0000000000000000000000000000000000000000
--- a/daemon/contrib/src/ucommon/rules.mak
+++ /dev/null
@@ -1,33 +0,0 @@
-# UCOMMON
-# FIXME: switch to next release when it's out
-UCOMMON_VERSION := 907e98d36cdffa1a0c6c0344993fe0a86f1f239f
-UCOMMON_URL := $(GNUTELEPHONY)/ucommon/archive/$(UCOMMON_VERSION).tar.gz
-
-UCOMMON_OPTIONS := --enable-stdcpp --with-pkg-config --disable-utils --disable-tests
-
-ifeq ($(call need_pkg,'ucommon >= ${UCOMMON_VERSION} commoncpp >= ${UCOMMON_VERSION}'),)
-PKGS_FOUND += ucommon
-endif
-
-DEPS_ucommon = gnutls
-
-$(TARBALLS)/ucommon-$(UCOMMON_VERSION).tar.gz:
- $(call download,$(UCOMMON_URL))
-
-.sum-ucommon: ucommon-$(UCOMMON_VERSION).tar.gz
- $(warning $@ not implemented)
- touch $@
-
-ucommon: ucommon-$(UCOMMON_VERSION).tar.gz .sum-ucommon
- $(UNPACK)
-ifdef HAVE_WIN32
- $(APPLY) $(SRC)/ucommon/windows_platform.patch
-endif
- $(UPDATE_AUTOCONFIG) && cd $(UNPACK_DIR)
- $(MOVE)
-
-.ucommon: ucommon
- mkdir -p $</m4 && $(RECONF)
- cd $< && $(HOSTVARS) ./configure $(HOSTCONF) $(UCOMMON_OPTIONS)
- cd $< && $(MAKE) install
- touch $@
diff --git a/daemon/contrib/src/ucommon/windows_platform.patch b/daemon/contrib/src/ucommon/windows_platform.patch
deleted file mode 100644
index f6163e406cca8f93f6d2cbcd98d2040af253c1a8..0000000000000000000000000000000000000000
--- a/daemon/contrib/src/ucommon/windows_platform.patch
+++ /dev/null
@@ -1,11 +0,0 @@
---- ucommon/inc/ucommon/platform.h.orig 2014-09-09 15:45:10.488481802 -0400
-+++ ucommon/inc/ucommon/platform.h 2014-09-09 15:45:39.268482233 -0400
-@@ -161,7 +161,7 @@
- #if defined(UCOMMON_RUNTIME) || defined(UCOMMON_STATIC)
- #define __SHARED
- #else
--#define __SHARED __EXPORT
-+#define __SHARED __declspec(dllimport) // Fix compile error.
- #endif
-
- #elif UCOMMON_VISIBILITY > 0
diff --git a/daemon/contrib/src/zrtpcpp/SHA512SUMS b/daemon/contrib/src/zrtpcpp/SHA512SUMS
deleted file mode 100644
index efcd32a904e49daf075e511db4176a4822bce54d..0000000000000000000000000000000000000000
--- a/daemon/contrib/src/zrtpcpp/SHA512SUMS
+++ /dev/null
@@ -1 +0,0 @@
-8166fdf7cc0cf9f35ca3245e6152be23e30010bf2113bc4be03c3cafb7c300fd8bd065829c151d05fda748c62d79d4d5d696524b2298effcac2498efd84420d0 libzrtpcpp-2.3.3.tar.gz
diff --git a/daemon/contrib/src/zrtpcpp/forcegcrypt.patch b/daemon/contrib/src/zrtpcpp/forcegcrypt.patch
deleted file mode 100644
index 194d802430e79a895ad40365bb29cfc19db0eca5..0000000000000000000000000000000000000000
--- a/daemon/contrib/src/zrtpcpp/forcegcrypt.patch
+++ /dev/null
@@ -1,11 +0,0 @@
---- a/CMakeLists.txt.orig 2014-05-06 19:35:46.443707232 -0400
-+++ b/CMakeLists.txt 2014-05-06 19:13:04.235686028 -0400
-@@ -107,7 +107,7 @@
- endif()
-
- # now get info about crypto libraries
--pkg_check_modules(OPENSSL libcrypto>=0.9.8)
-+#pkg_check_modules(OPENSSL libcrypto>=0.9.8)
- if (OPENSSL_FOUND)
- set(CMAKE_REQUIRED_INCLUDES ${CMAKE_REQUIRED_INCLUDES} ${OPENSSL_INCLUDE_DIRS}) #update include files search directory
- check_include_files(openssl/bn.h HAVE_OPENSSL_BN_H)
diff --git a/daemon/contrib/src/zrtpcpp/nodemo.patch b/daemon/contrib/src/zrtpcpp/nodemo.patch
deleted file mode 100644
index 977af3bb9093de5c73bc92f78462b7b97486b55e..0000000000000000000000000000000000000000
--- a/daemon/contrib/src/zrtpcpp/nodemo.patch
+++ /dev/null
@@ -1,13 +0,0 @@
---- zrtpcpp/CMakeLists.txt.orig 2014-08-01 13:09:30.402259035 -0400
-+++ zrtpcpp/CMakeLists.txt 2014-08-01 13:09:35.089304129 -0400
-@@ -167,10 +167,6 @@
-
- add_subdirectory(src)
-
--if (enable_ccrtp)
-- add_subdirectory(demo)
--endif()
--
- if (NOT EXISTS ${CMAKE_CURRENT_BINARY_DIR}/package/)
- MESSAGE(STATUS "package dir not found")
- file(MAKE_DIRECTORY ${CMAKE_CURRENT_BINARY_DIR}/package/)
diff --git a/daemon/contrib/src/zrtpcpp/rules.mak b/daemon/contrib/src/zrtpcpp/rules.mak
deleted file mode 100644
index cd26c18656c059908c7993503db05b6f28470003..0000000000000000000000000000000000000000
--- a/daemon/contrib/src/zrtpcpp/rules.mak
+++ /dev/null
@@ -1,33 +0,0 @@
-# ZRTPCPP
-ZRTPCPP_VERSION := 2.3.3
-ZRTPCPP_URL := $(GNUTELEPHONY)/ZRTPCPP/archive/v$(ZRTPCPP_VERSION).tar.gz
-
-PKGS += zrtpcpp
-ifeq ($(call need_pkg,'libzrtpcpp >= 2.0.0'),)
-PKGS_FOUND += zrtpcpp
-endif
-
-DEPS_zrtpcpp = ccrtp ucommon gcrypt
-
-ZRTPCPP_CMAKECONF := -DBUILD_STATIC:BOOL=ON \
- -DBUILD_SHARED:BOOL=OFF \
- -DBUILD_SHARED_LIBS:BOOL=OFF
-
-$(TARBALLS)/libzrtpcpp-$(ZRTPCPP_VERSION).tar.gz:
- $(call download,$(ZRTPCPP_URL))
-
-.sum-zrtpcpp: libzrtpcpp-$(ZRTPCPP_VERSION).tar.gz
-
-zrtpcpp: libzrtpcpp-$(ZRTPCPP_VERSION).tar.gz .sum-zrtpcpp
- $(UNPACK)
- mv ZRTPCPP-$(ZRTPCPP_VERSION) libzrtpcpp-$(ZRTPCPP_VERSION)
- $(APPLY) $(SRC)/zrtpcpp/forcegcrypt.patch
- $(APPLY) $(SRC)/zrtpcpp/threadcbsupdate.patch
- $(APPLY) $(SRC)/zrtpcpp/standardheader.patch
- $(APPLY) $(SRC)/zrtpcpp/nodemo.patch
- $(MOVE)
-
-.zrtpcpp: zrtpcpp toolchain.cmake
- cd $< && $(HOSTVARS) $(CMAKE) . $(ZRTPCPP_CMAKECONF)
- cd $< && $(MAKE) install VERBOSE=1
- touch $@
diff --git a/daemon/contrib/src/zrtpcpp/standardheader.patch b/daemon/contrib/src/zrtpcpp/standardheader.patch
deleted file mode 100644
index 9f5faedcf252a6f59515dae6f8517ed43318e90d..0000000000000000000000000000000000000000
--- a/daemon/contrib/src/zrtpcpp/standardheader.patch
+++ /dev/null
@@ -1,10 +0,0 @@
---- a/src/libzrtpcpp/crypto/gcrypt/InitializeGcrypt.cpp.orig 2014-06-25 20:40:20.000000000 -0400
-+++ b/src/libzrtpcpp/crypto/gcrypt/InitializeGcrypt.cpp 2014-06-25 20:40:32.000000000 -0400
-@@ -17,7 +17,7 @@
-
- #include <stdio.h>
-
--#include <malloc.h>
-+#include <stdlib.h>
- #include <errno.h>
- #include <gcrypt.h>
diff --git a/daemon/contrib/src/zrtpcpp/threadcbsupdate.patch b/daemon/contrib/src/zrtpcpp/threadcbsupdate.patch
deleted file mode 100644
index 6a9ddb29dd34e317f2fd7cd577c6e41d12960c04..0000000000000000000000000000000000000000
--- a/daemon/contrib/src/zrtpcpp/threadcbsupdate.patch
+++ /dev/null
@@ -1,16 +0,0 @@
---- a/src/libzrtpcpp/crypto/gcrypt/InitializeGcrypt.cpp.orig 2014-06-12 17:36:14.367890764 -0400
-+++ b/src/libzrtpcpp/crypto/gcrypt/InitializeGcrypt.cpp 2014-06-12 17:36:52.899889626 -0400
-@@ -68,9 +68,12 @@
- }
-
- static struct gcry_thread_cbs gcry_threads = {
-- GCRY_THREAD_OPTION_PTHREAD, NULL,
-+ GCRY_THREAD_OPTION_PTHREAD,
-+#if 0
-+ NULL,
- gcry_thread_mutex_init, gcry_thread_mutex_destroy,
- gcry_thread_mutex_lock, gcry_thread_mutex_unlock
-+#endif
- };
-
- #else
diff --git a/daemon/globals.mak b/daemon/globals.mak
index 90f2f5162de1a6c46364c2bee1dd6a17e6e39fb3..c6e53e47bb0f85af1d0394d3808623fcc819df8f 100644
--- a/daemon/globals.mak
+++ b/daemon/globals.mak
@@ -24,6 +24,7 @@ endif
AM_CPPFLAGS = \
-I$(src)/src \
-I$(src)/src/config \
+ -I$(src)/src/media \
-I$(src)/test \
-I$(src)/src/public \
$(SIP_CFLAGS) \
diff --git a/daemon/src/Makefile.am b/daemon/src/Makefile.am
index 37535306928462a9844d78220cc7ebbb592382ca..efa64bd00906f589797d091403ae8323923001ac 100644
--- a/daemon/src/Makefile.am
+++ b/daemon/src/Makefile.am
@@ -1,9 +1,8 @@
include ../globals.mak
if RING_VIDEO
-RING_VIDEO_SUBDIR = video
-RING_VIDEO_LIBS=./video/libvideo.la \
- ./video/v4l2/libv4l2.la
+RING_VIDEO_LIBS=./media/video/libvideo.la \
+ ./media/video/v4l2/libv4l2.la
endif
if BUILD_INSTANT_MESSAGING
@@ -35,7 +34,7 @@ TLS_LIB = @GNUTLS_LIBS@
TLS_CFLAGS = @GNUTLS_CFLAGS@
endif
-SUBDIRS = client audio config hooks history sip $(IAX_SUBDIR) $(RINGACC_SUBDIR) $(INSTANT_MESSAGING_SUBDIR) $(RING_VIDEO_SUBDIR)
+SUBDIRS = client media config hooks history sip $(IAX_SUBDIR) $(RINGACC_SUBDIR) $(INSTANT_MESSAGING_SUBDIR) $(RING_VIDEO_SUBDIR)
# libring
@@ -43,7 +42,7 @@ lib_LTLIBRARIES = libring.la
libring_la_LIBADD = \
./sip/libsiplink.la \
- ./audio/libaudio.la \
+ ./media/libmedia.la \
./client/libclient.la \
./config/libconfig.la \
./hooks/libhooks.la \
@@ -54,9 +53,7 @@ libring_la_LIBADD = \
$(RING_VIDEO_LIBS)
libring_la_LDFLAGS = \
- @ZRTPCPP_LIBS@ \
@PJPROJECT_LIBS@ \
- @CCRTP_LIBS@ \
@ALSA_LIBS@ \
@PULSEAUDIO_LIBS@ \
@SAMPLERATE_LIBS@ \
@@ -72,10 +69,13 @@ if USE_DHT
libring_la_LDFLAGS += $(OPENDHT_LIBS)
endif
+if HAVE_OSX
+#FIXME necessary for -lintl
+libring_la_LDFLAGS += -L/usr/local/opt/gettext/lib
+endif
+
libring_la_CFLAGS = \
- @ZRTPCPP_CFLAGS@ \
@PJPROJECT_CFLAGS@ \
- @CCRTP_CFLAGS@ \
@ALSA_CFLAGS@ \
@PULSEAUDIO_CFLAGS@ \
@SAMPLERATE_CFLAGS@ \
@@ -131,5 +131,4 @@ libring_la_SOURCES = conference.cpp \
string_utils.h \
string_utils.cpp \
rw_mutex.h \
- ring.h \
ring_api.cpp
diff --git a/daemon/src/account.cpp b/daemon/src/account.cpp
index 8d6de36d9dadbbfc0d379ef898cf5be8f696c785..78c014b0c9615c247c1c6d856f2474535e00f3a7 100644
--- a/daemon/src/account.cpp
+++ b/daemon/src/account.cpp
@@ -39,7 +39,7 @@
#include <iterator>
#ifdef RING_VIDEO
-#include "video/libav_utils.h"
+#include "libav_utils.h"
#endif
#include "logger.h"
diff --git a/daemon/src/audio/audiortp/Makefile.am b/daemon/src/audio/audiortp/Makefile.am
deleted file mode 100644
index b1e36b05f9a09827cd19ce533bbbe0391d12964b..0000000000000000000000000000000000000000
--- a/daemon/src/audio/audiortp/Makefile.am
+++ /dev/null
@@ -1,37 +0,0 @@
-include $(top_srcdir)/globals.mak
-
-noinst_LTLIBRARIES = libaudiortp.la
-
-if BUILD_ZRTP
-RING_ZRTP_SRC=audio_zrtp_session.h audio_zrtp_session.cpp zrtp_session_callback.cpp zrtp_session_callback.h
-endif
-
-libaudiortp_la_SOURCES = \
- $(RING_ZRTP_SRC) \
- base64.c base64.h
-
-if RING_VIDEO
-libaudiortp_la_SOURCES += \
- avformat_rtp_session.cpp \
- avformat_rtp_session.h
-
-AM_CXXFLAGS = @LIBAVFORMAT_CFLAGS@
-
-else
-libaudiortp_la_SOURCES += \
- audio_rtp_session.cpp \
- audio_symmetric_rtp_session.cpp \
- audio_rtp_stream.cpp \
- audio_rtp_factory.cpp \
- audio_srtp_session.cpp \
- audio_rtp_session.h \
- dtmf_event.h \
- dtmf_event.cpp \
- audio_rtp_stream.h \
- audio_rtp_factory.h \
- audio_symmetric_rtp_session.h \
- audio_srtp_session.h
-endif
-
-# FIXME
-AM_CPPFLAGS += -I$(top_srcdir)/src
diff --git a/daemon/src/audio/audiortp/audio_rtp_factory.cpp b/daemon/src/audio/audiortp/audio_rtp_factory.cpp
deleted file mode 100644
index e10c4d78026fc6e31a9ceace80a3a74b1fca3b57..0000000000000000000000000000000000000000
--- a/daemon/src/audio/audiortp/audio_rtp_factory.cpp
+++ /dev/null
@@ -1,259 +0,0 @@
-/*
- * Copyright (C) 2004-2015 Savoir-Faire Linux Inc.
- * Author: Pierre-Luc Bacon <pierre-luc.bacon@savoirfairelinux.com>
- * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- * This program 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- *
- * Additional permission under GNU GPL version 3 section 7:
- *
- * If you modify this program, or any covered work, by linking or
- * combining it with the OpenSSL project's OpenSSL library (or a
- * modified version of that library), containing parts covered by the
- * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
- * grants you additional permission to convey the resulting work.
- * Corresponding Source for a non-source form of such a combination
- * shall include the source code for the parts of OpenSSL used as well
- * as that of the covered work.
- */
-
-#include "audio_rtp_factory.h"
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#if HAVE_ZRTP
-#include "audio_zrtp_session.h"
-#endif
-#include "audio_symmetric_rtp_session.h"
-#include "manager.h"
-#include "sip/sdp.h"
-#include "sip/sipcall.h"
-#include "sip/sipaccount.h"
-#include "logger.h"
-
-namespace ring {
-
-AudioRtpFactory::AudioRtpFactory(SIPCall *ca) : rtpSession_(),
- audioRtpThreadMutex_(), srtpEnabled_(false), helloHashEnabled_(false),
- remoteOfferIsSet_(false), call_(ca),
- keyExchangeProtocol_(NONE)
-{
- // FIXME: workaround for uncatchable ost::Socket "exceptions"
- ost::Thread::setException(ost::Thread::throwNothing);
-}
-
-AudioRtpFactory::~AudioRtpFactory()
-{}
-
-void AudioRtpFactory::initConfig()
-{
- stop();
-
- const std::string accountId(call_->getAccountId());
-
- const auto sipaccount = Manager::instance().getAccount<SIPAccount>(accountId);
-
- if (sipaccount) {
- srtpEnabled_ = sipaccount->getSrtpEnabled();
- std::string key(sipaccount->getSrtpKeyExchange());
-
- if (srtpEnabled_) {
-#if HAVE_ZRTP
-
- if (key == "sdes")
- keyExchangeProtocol_ = SDES;
- else if (key == "zrtp")
- keyExchangeProtocol_ = ZRTP;
-
-#else
- keyExchangeProtocol_ = SDES;
-#endif
- } else {
- keyExchangeProtocol_ = NONE;
- }
-
- helloHashEnabled_ = sipaccount->getZrtpHelloHash();
- } else {
- srtpEnabled_ = false;
- keyExchangeProtocol_ = NONE;
- helloHashEnabled_ = false;
- }
-}
-
-void AudioRtpFactory::initSession()
-{
- std::lock_guard<std::mutex> lock(audioRtpThreadMutex_);
-
- if (srtpEnabled_) {
-
- switch (keyExchangeProtocol_) {
-#if HAVE_ZRTP
- case ZRTP: {
- const std::string zidFilename(Manager::instance().voipPreferences.getZidFile());
- rtpSession_.reset(new AudioZrtpSession(*call_, zidFilename, call_->getLocalIp()));
-
- // TODO: be careful with that. The hello hash is computed asynchronously. Maybe it's
- // not even available at that point.
- if (helloHashEnabled_)
- call_->getSDP().setZrtpHash(static_cast<AudioZrtpSession *>(rtpSession_.get())->getHelloHash());
- break;
- }
-#endif
-
- case SDES: {
- rtpSession_.reset(new AudioSrtpSession(*call_));
- break;
- }
-
- default:
- throw UnsupportedRtpSessionType("Unsupported Rtp Session Exception Type!");
- }
- } else {
-#if HAVE_IPV6
- if (call_->getLocalIp().isIpv6()) {
- rtpSession_.reset(new AudioSymmetricRtpSessionIPv6(*call_));
- } else
-#endif
- {
- rtpSession_.reset(new AudioSymmetricRtpSession(*call_));
- }
- }
-}
-
-std::vector<long>
-AudioRtpFactory::getSocketDescriptors()
-{
- std::lock_guard<std::mutex> lock(audioRtpThreadMutex_);
- if (!rtpSession_)
- throw AudioRtpFactoryException("RTP session was null when trying to get socket descriptors");
- return rtpSession_->getSocketDescriptors();
-}
-
-void AudioRtpFactory::start(const std::vector<AudioCodec*> &audioCodecs)
-{
- std::lock_guard<std::mutex> lock(audioRtpThreadMutex_);
- if (!rtpSession_)
- throw AudioRtpFactoryException("RTP session was null when trying to start audio thread");
-
- rtpSession_->startRtpThreads(audioCodecs);
-}
-
-void AudioRtpFactory::stop()
-{
- std::lock_guard<std::mutex> lock(audioRtpThreadMutex_);
- rtpSession_.reset();
-}
-
-void AudioRtpFactory::updateSessionMedia(const std::vector<AudioCodec*> &audioCodecs)
-{
- std::lock_guard<std::mutex> lock(audioRtpThreadMutex_);
- if (!rtpSession_)
- throw AudioRtpFactoryException("rtpSession_ was NULL when trying to update IP address");
-
- rtpSession_->updateSessionMedia(audioCodecs);
-}
-
-void AudioRtpFactory::updateDestinationIpAddress()
-{
- std::lock_guard<std::mutex> lock(audioRtpThreadMutex_);
- if (!rtpSession_)
- throw AudioRtpFactoryException("RTP session was null when trying to update IP address");
-
- rtpSession_->updateDestinationIpAddress();
-}
-
-#if HAVE_ZRTP
-AudioZrtpSession * AudioRtpFactory::getAudioZrtpSession()
-{
- std::lock_guard<std::mutex> lock(audioRtpThreadMutex_);
- if (rtpSession_ and keyExchangeProtocol_ == ZRTP)
- return static_cast<AudioZrtpSession *>(rtpSession_.get());
- else
- throw AudioRtpFactoryException("rtpSession_ is NULL in getAudioZrtpSession");
-}
-#endif
-
-void AudioRtpFactory::initLocalCryptoInfo()
-{
- std::lock_guard<std::mutex> lock(audioRtpThreadMutex_);
-
- if (rtpSession_ && keyExchangeProtocol_ == SDES) {
- AudioSrtpSession *srtp = static_cast<AudioSrtpSession*>(rtpSession_.get());
- // the context is invalidated and deleted by the call to initLocalCryptoInfo
- srtp->initLocalCryptoInfo();
- call_->getSDP().setLocalSdpCrypto(srtp->getLocalCryptoInfo());
- }
-}
-
-void AudioRtpFactory::initLocalCryptoInfoOnOffHold()
-{
- std::lock_guard<std::mutex> lock(audioRtpThreadMutex_);
-
- if (rtpSession_ && keyExchangeProtocol_ == SDES) {
- AudioSrtpSession *srtp = static_cast<AudioSrtpSession*>(rtpSession_.get());
- // the context is invalidated and deleted by the call to initLocalCryptoInfo
- srtp->initLocalCryptoInfoOnOffhold();
- call_->getSDP().setLocalSdpCrypto(srtp->getLocalCryptoInfo());
- }
-}
-
-
-void AudioRtpFactory::setRemoteCryptoInfo(SdesNegotiator& nego)
-{
- std::lock_guard<std::mutex> lock(audioRtpThreadMutex_);
- if (!rtpSession_)
- throw AudioRtpFactoryException("rtpSession_ is NULL in setRemoteCryptoInfo");
-
- if (keyExchangeProtocol_ == SDES) {
- AudioSrtpSession *srtp = static_cast<AudioSrtpSession *>(rtpSession_.get());
- try {
- srtp->setRemoteCryptoInfo(nego);
- } catch (const AudioSrtpException &e) {
- throw AudioRtpFactoryException(e.what());
- }
- } else {
- RING_ERR("Should not store remote crypto info for non-SDES sessions");
- }
-}
-
-void AudioRtpFactory::setDtmfPayloadType(unsigned int payloadType)
-{
- std::lock_guard<std::mutex> lock(audioRtpThreadMutex_);
- if (rtpSession_)
- rtpSession_->setDtmfPayloadType(payloadType);
-}
-
-void AudioRtpFactory::sendDtmfDigit(int digit)
-{
- std::lock_guard<std::mutex> lock(audioRtpThreadMutex_);
- if (rtpSession_)
- rtpSession_->putDtmfEvent(digit);
-}
-
-void AudioRtpFactory::saveLocalContext()
-{
- std::lock_guard<std::mutex> lock(audioRtpThreadMutex_);
- if (rtpSession_ and keyExchangeProtocol_ == SDES)
- cachedAudioRtpState_.reset(rtpSession_->saveState());
-}
-
-void AudioRtpFactory::restoreLocalContext()
-{
- std::lock_guard<std::mutex> lock(audioRtpThreadMutex_);
- if (rtpSession_ and keyExchangeProtocol_ == SDES)
- rtpSession_->restoreState(*cachedAudioRtpState_);
-}
-}
diff --git a/daemon/src/audio/audiortp/audio_rtp_factory.h b/daemon/src/audio/audiortp/audio_rtp_factory.h
deleted file mode 100644
index 12a4deed75e4ba2245f8456cd929f21c0d7e55de..0000000000000000000000000000000000000000
--- a/daemon/src/audio/audiortp/audio_rtp_factory.h
+++ /dev/null
@@ -1,171 +0,0 @@
-/*
- * Copyright (C) 2004-2015 Savoir-Faire Linux Inc.
- * Author: Pierre-Luc Bacon <pierre-luc.bacon@savoirfairelinux.com>
- * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- * This program 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- *
- * Additional permission under GNU GPL version 3 section 7:
- *
- * If you modify this program, or any covered work, by linking or
- * combining it with the OpenSSL project's OpenSSL library (or a
- * modified version of that library), containing parts covered by the
- * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
- * grants you additional permission to convey the resulting work.
- * Corresponding Source for a non-source form of such a combination
- * shall include the source code for the parts of OpenSSL used as well
- * as that of the covered work.
- */
-
-#ifndef __AUDIO_RTP_FACTORY_H__
-#define __AUDIO_RTP_FACTORY_H__
-
-#include <stdexcept>
-#include <mutex>
-#include <memory>
-#include "audio_rtp_session.h"
-#include "audio_srtp_session.h"
-#include "noncopyable.h"
-
-class SdesNegotiator;
-class SIPCall;
-
-namespace ring {
-
-#if HAVE_ZRTP
-class AudioZrtpSession;
-#endif
-class AudioCodec;
-
-class UnsupportedRtpSessionType : public std::logic_error {
- public:
- UnsupportedRtpSessionType(const std::string& msg = "") : std::logic_error(msg) {}
-};
-
-class AudioRtpFactoryException : public std::logic_error {
- public:
- AudioRtpFactoryException(const std::string& msg = "") : std::logic_error(msg) {}
-};
-
-class AudioRtpFactory {
- public:
- AudioRtpFactory(SIPCall *ca);
- ~AudioRtpFactory();
-
- std::vector<long>
- getSocketDescriptors();
-
- void initConfig();
-
- /**
- * Lazy instantiation method. Create a new RTP session of a given
- * type according to the content of the configuration file.
- * @param ca A pointer on a SIP call
- * @return A new AudioRtpSession object
- */
- void initSession();
-
- /**
- * Start the audio rtp thread of the type specified in the configuration
- * file. initAudioSymmetricRtpSession must have been called prior to that.
- * @param None
- */
- void start(const std::vector<AudioCodec*> &audioCodecs);
-
- /**
- * Stop the audio rtp thread of the type specified in the configuration
- * file. initAudioSymmetricRtpSession must have been called prior to that.
- * @param None
- */
- void stop();
-
- /**
- * Dynamically update session media
- */
- void updateSessionMedia(const std::vector<AudioCodec*> &audioCodecs);
-
- /**
- * Update current RTP destination address with one stored in call
- * @param None
- */
- void updateDestinationIpAddress();
-
- bool isSdesEnabled() const {
- return srtpEnabled_ and keyExchangeProtocol_ == SDES;
- }
-
- /**
- * Manually set the srtpEnable option (usefull for RTP fallback)
- */
- void setSrtpEnabled(bool enable) {
- srtpEnabled_ = enable;
- }
-
-#if HAVE_ZRTP
- /**
- * Get the current AudioZrtpSession. Throws an AudioRtpFactoryException
- * if the current rtp thread is null, or if it's not of the correct type.
- * @return The current AudioZrtpSession thread.
- */
- ring::AudioZrtpSession* getAudioZrtpSession();
-#endif
-
- void initLocalCryptoInfo();
- void initLocalCryptoInfoOnOffHold();
-
- /**
- * Set remote cryptographic info. Should be called after negotiation in SDP
- * offer/answer session.
- */
- void setRemoteCryptoInfo(ring::SdesNegotiator& nego);
-
- void setDtmfPayloadType(unsigned int);
-
- /**
- * Send DTMF over RTP (RFC2833).
- * The timestamp and sequence number must be incremented
- * as if it was microphone audio. This function change the payload type
- * of the rtp session, send the appropriate DTMF digit using
- * this payload, discard corresponding data from RingBufferPool and get
- * back the codec payload for further audio processing.
- */
- void sendDtmfDigit(int digit);
-
- void saveLocalContext();
-
- void restoreLocalContext();
-
- private:
- NON_COPYABLE(AudioRtpFactory);
- enum KeyExchangeProtocol { NONE, SDES, ZRTP };
- std::unique_ptr<AudioRtpSession> rtpSession_;
- std::unique_ptr<CachedAudioRtpState> cachedAudioRtpState_;
- std::mutex audioRtpThreadMutex_;
-
- // Field used when initializing audio rtp session
- // May be set manually or from config using initAudioRtpConfig
- bool srtpEnabled_;
-
- // Field used when initializinga udio rtp session
- // May be set manually or from config using initAudioRtpConfig
- bool helloHashEnabled_;
-
- /** Used to make sure remote crypto context not initialized twice. */
- bool remoteOfferIsSet_;
-
- SIPCall *call_;
- KeyExchangeProtocol keyExchangeProtocol_;
-};
-}
-#endif // __AUDIO_RTP_FACTORY_H__
diff --git a/daemon/src/audio/audiortp/audio_rtp_session.cpp b/daemon/src/audio/audiortp/audio_rtp_session.cpp
deleted file mode 100644
index 51d51a7fafce4b58b08ea487381768b0025e6b21..0000000000000000000000000000000000000000
--- a/daemon/src/audio/audiortp/audio_rtp_session.cpp
+++ /dev/null
@@ -1,348 +0,0 @@
-/*
- * Copyright (C) 2004-2015 Savoir-Faire Linux Inc.
- * Author: Pierre-Luc Bacon <pierre-luc.bacon@savoirfairelinux.com>
- * Author: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
- * Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
- * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- * This program 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- *
- * Additional permission under GNU GPL version 3 section 7:
- *
- * If you modify this program, or any covered work, by linking or
- * combining it with the OpenSSL project's OpenSSL library (or a
- * modified version of that library), containing parts covered by the
- * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
- * grants you additional permission to convey the resulting work.
- * Corresponding Source for a non-source form of such a combination
- * shall include the source code for the parts of OpenSSL used as well
- * as that of the covered work.
- */
-
-#include "audio_rtp_session.h"
-#include "logger.h"
-#include "sip/sdp.h"
-#include "sip/sipcall.h"
-#include "intrin.h"
-
-#ifdef RTP_DEBUG
-#include <numeric>
-#include <algorithm>
-#endif
-
-namespace ring {
-AudioRtpSession::AudioRtpSession(SIPCall &call, ost::RTPDataQueue &queue) :
- isStarted_(false)
- , queue_(queue)
- , call_(call)
- , timestamp_(0)
- , timestampIncrement_(0)
- , rtpStream_(call.getCallId())
- , transportRate_(20)
- , remoteIp_()
- , rxLastSeqNum_(0)
-#ifdef RTP_DEBUG
- , rxLast_()
- , rxJitters_()
- , jitterReportInterval_(0)
-#endif
- , dtmfQueue_()
- , dtmfPayloadType_(101) // same as Asterisk
- , loop_([] { return true; },
- std::bind(&AudioRtpSession::process, this),
- [] {})
-{
- queue_.setTypeOfService(ost::RTPDataQueue::tosEnhanced);
-}
-
-AudioRtpSession::~AudioRtpSession()
-{
- loop_.join();
-}
-
-void AudioRtpSession::updateSessionMedia(const std::vector<AudioCodec*> &audioCodecs)
-{
- if (rtpStream_.codecsDiffer(audioCodecs))
- setSessionMedia(audioCodecs);
-}
-
-void AudioRtpSession::setSessionMedia(const std::vector<AudioCodec*> &audioCodecs)
-{
- rtpStream_.setRtpMedia(audioCodecs);
-
- // G722 requires timestamp to be incremented at 8kHz
- const ost::PayloadType payloadType = rtpStream_.getEncoderPayloadType();
-
- if (payloadType == ost::sptG722) {
- const int G722_RTP_TIME_INCREMENT = 160;
- timestampIncrement_ = G722_RTP_TIME_INCREMENT;
- } else
- timestampIncrement_ = rtpStream_.getEncoderFrameSize();
-
- const AudioFormat encoderFormat(rtpStream_.getEncoderFormat());
-
- if (payloadType == ost::sptG722) {
- const int G722_RTP_CLOCK_RATE = 8000;
- queue_.setPayloadFormat(ost::DynamicPayloadFormat(payloadType, G722_RTP_CLOCK_RATE));
- } else {
- if (rtpStream_.hasDynamicPayload())
- queue_.setPayloadFormat(ost::DynamicPayloadFormat(payloadType, encoderFormat.sample_rate));
- else
- queue_.setPayloadFormat(ost::StaticPayloadFormat(static_cast<ost::StaticPayloadType>(payloadType)));
- }
-
- transportRate_ = rtpStream_.getTransportRate();
- RING_DBG("Switching to a transport rate of %d ms", transportRate_);
-}
-
-void AudioRtpSession::sendDtmfEvent()
-{
- DTMFEvent &dtmf(dtmfQueue_.front());
- RING_DBG("Send RTP Dtmf (%d)", dtmf.payload.event);
-
- const int increment = getIncrementForDTMF();
- if (dtmf.newevent)
- timestamp_ += increment;
-
- // discard equivalent size of audio
- rtpStream_.processDataEncode();
-
- // change Payload type for DTMF payload
- queue_.setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) dtmfPayloadType_, 8000));
-
- // Set marker in case this is a new Event
- if (dtmf.newevent)
- queue_.setMark(true);
-
- // Send end packet three times (without changing it). Sequence number is
- // incremented automatically by ccrtp, which is the correct behaviour.
- const unsigned repetitions = dtmf.payload.ebit ? 3 : 1;
- for (unsigned i = 0; i < repetitions; ++i)
- queue_.sendImmediate(timestamp_, (const unsigned char *)(& (dtmf.payload)), sizeof(ost::RTPPacket::RFC2833Payload));
-
- // This is no longer a new event
- if (dtmf.newevent) {
- dtmf.newevent = false;
- queue_.setMark(false);
- }
-
- // restore the payload to audio
- const int pt = rtpStream_.getEncoderPayloadType();
- const AudioFormat encoderFormat = rtpStream_.getEncoderFormat();
- if (rtpStream_.hasDynamicPayload()) {
- const ost::DynamicPayloadFormat pf(pt, encoderFormat.sample_rate);
- queue_.setPayloadFormat(pf);
- } else {
- const ost::StaticPayloadFormat pf(static_cast<ost::StaticPayloadType>(pt));
- queue_.setPayloadFormat(pf);
- }
-
- // decrease length remaining to process for this event
- dtmf.length -= increment;
- dtmf.payload.duration++;
-
- // next packet is going to be the end packet (transmitted 3 times)
- if ((dtmf.length - increment) < increment)
- dtmf.payload.ebit = true;
-
- if (dtmf.length < increment)
- dtmfQueue_.pop_front();
-}
-
-
-void AudioRtpSession::receiveSpeakerData()
-{
- const ost::AppDataUnit* adu = queue_.getData(queue_.getFirstTimestamp());
-
- if (!adu)
- return;
-
-#ifdef RTP_DEBUG
- // Compute and print RX jitter
- auto rxTime = std::chrono::high_resolution_clock::now();
- rxJitters_.push_back(std::chrono::duration_cast<std::chrono::duration<double, std::milli>>(rxTime - rxLast_).count());
- rxLast_ = rxTime;
- if (++jitterReportInterval_ == 100) {
- if (rxJitters_.size() > 1000)
- rxJitters_.erase(rxJitters_.begin(), rxJitters_.begin() + (rxJitters_.size() - 1000));
- const double jit_mean = std::accumulate(rxJitters_.begin(), rxJitters_.end(), 0.0) / rxJitters_.size();
- const double jit_sq_sum = std::inner_product(rxJitters_.begin(), rxJitters_.end(), rxJitters_.begin(), 0.0);
- const double jit_stdev = std::sqrt(jit_sq_sum / rxJitters_.size() - jit_mean * jit_mean);
- RING_DBG("Jitter avg: %fms std dev %fms", jit_mean, jit_stdev);
- jitterReportInterval_ = 0;
- }
-#endif
-
- int seqNumDiff = adu->getSeqNum() - rxLastSeqNum_;
- if (abs(seqNumDiff) > 512) {
- // Don't perform PLC if the delay can't be concealed without major distortion.
- // Also skip PLC in case of timestamp reset.
- rxLastSeqNum_ += seqNumDiff;
- seqNumDiff = 1;
- } else if (seqNumDiff < 0) {
- RING_DBG("Dropping out-of-order packet %d (last %d)", rxLastSeqNum_ + seqNumDiff, rxLastSeqNum_);
- return;
- } else {
- rxLastSeqNum_ += seqNumDiff;
- }
- if (rxLastSeqNum_ && seqNumDiff > 1) {
- RING_DBG("%d packets lost", seqNumDiff-1);
- for (unsigned i = 0, n = seqNumDiff - 1; i < n; i++)
- rtpStream_.processDataDecode(nullptr, 0, adu->getType());
- }
-
- uint8_t* spkrDataIn = (uint8_t*) adu->getData(); // data in char
- size_t size = adu->getSize(); // size in char
-
- // DTMF over RTP, size must be over 4 in order to process it as voice data
- if (size > 4)
- rtpStream_.processDataDecode(spkrDataIn, size, adu->getType());
-
- delete adu;
-}
-
-
-size_t AudioRtpSession::sendMicData()
-{
- size_t compSize = rtpStream_.processDataEncode();
-
- // if no data return
- if (compSize == 0)
- return 0;
-
- // initialize once
- int ccrtpTimestamp = queue_.getCurrentTimestamp();
- if (std::abs(timestamp_ - ccrtpTimestamp) > timestampIncrement_) {
- timestamp_ = ccrtpTimestamp;
- }
-
- // Increment timestamp for outgoing packet
- timestamp_ += timestampIncrement_;
-
- queue_.putData(timestamp_, rtpStream_.getMicDataEncoded(), compSize);
- return compSize;
-}
-
-
-void AudioRtpSession::setSessionTimeouts()
-{
- const unsigned schedulingTimeout = 4000;
- const unsigned expireTimeout = 1000000;
- RING_DBG("Set session scheduling timeout (%d) and expireTimeout (%d)",
- schedulingTimeout, expireTimeout);
-
- queue_.setSchedulingTimeout(schedulingTimeout);
- queue_.setExpireTimeout(expireTimeout);
-}
-
-void AudioRtpSession::updateDestinationIpAddress()
-{
- RING_DBG("Update destination ip address");
-
- // Destination address are stored in a list in ccrtp
- // This method remove the current destination entry
- if (!(remoteIp_.isIpv4() && queue_.forgetDestination(static_cast<ost::IPV4Host>(remoteIp_), remoteIp_.getPort()))
-#if HAVE_IPV6
- && !(remoteIp_.isIpv6() == AF_INET6 && queue_.forgetDestination(static_cast<ost::IPV6Host>(remoteIp_), remoteIp_.getPort()))
-#endif
- ) RING_DBG("Did not remove previous destination");
-
- IpAddr remote = {call_.getSDP().getRemoteIP()};
- remote.setPort(call_.getSDP().getRemoteAudioPort());
- if (!remote) {
- RING_WARN("Target IP address (%s) is not correct!", call_.getSDP().getRemoteIP().c_str());
- return;
- }
- remoteIp_ = remote;
- RING_DBG("New remote address for session: %s", remote.toString(true).c_str());
-
- if (!(remoteIp_.isIpv4() && queue_.addDestination(static_cast<ost::IPV4Host>(remoteIp_), remoteIp_.getPort()))
-#if HAVE_IPV6
- && !(remoteIp_.isIpv6() && queue_.addDestination(static_cast<ost::IPV6Host>(remoteIp_), remoteIp_.getPort()))
-#endif
- ) RING_WARN("Can't add new destination to session!");
-}
-
-
-void AudioRtpSession::prepareRtpReceiveThread(const std::vector<AudioCodec*> &audioCodecs)
-{
- RING_DBG("Preparing receiving thread");
- isStarted_ = true;
-#ifdef RTP_DEBUG
- rxLast_ = std::chrono::high_resolution_clock::now();
-#endif
- setSessionTimeouts();
- setSessionMedia(audioCodecs);
- rtpStream_.initBuffers();
-#if HAVE_SPEEXDSP
- rtpStream_.resetDSP();
-#endif
-}
-
-
-bool AudioRtpSession::onRTPPacketRecv(ost::IncomingRTPPkt&)
-{
- receiveSpeakerData();
- return true;
-}
-
-int AudioRtpSession::getIncrementForDTMF() const
-{
- return timestampIncrement_;
-}
-
-void AudioRtpSession::startRtpThreads(const std::vector<AudioCodec*> &audioCodecs)
-{
- if (isStarted_)
- return;
-
- prepareRtpReceiveThread(audioCodecs);
- // implemented in subclasses
- startRTPLoop();
- // only in this class
- loop_.start();
-}
-
-void AudioRtpSession::process()
-{
- // Send session
- if (hasDTMFPending())
- sendDtmfEvent();
- else
- sendMicData();
-
- rtpStream_.waitForDataEncode(std::chrono::milliseconds(transportRate_));
-}
-
-void AudioRtpSession::putDtmfEvent(char digit)
-{
- DTMFEvent dtmf(digit);
- dtmfQueue_.push_back(dtmf);
-}
-
-CachedAudioRtpState *
-AudioRtpSession::saveState() const
-{
- RING_ERR("Not implemented");
- return nullptr;
-}
-
-void
-AudioRtpSession::restoreState(const CachedAudioRtpState &state UNUSED)
-{
- RING_ERR("Not implemented");
-}
-}
diff --git a/daemon/src/audio/audiortp/audio_rtp_session.h b/daemon/src/audio/audiortp/audio_rtp_session.h
deleted file mode 100644
index b999e68c4f9c26714d59142346383559bdd87019..0000000000000000000000000000000000000000
--- a/daemon/src/audio/audiortp/audio_rtp_session.h
+++ /dev/null
@@ -1,192 +0,0 @@
-/*
- * Copyright (C) 2004-2015 Savoir-Faire Linux Inc.
- * Author: Pierre-Luc Bacon <pierre-luc.bacon@savoirfairelinux.com>
- * Author: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
- * Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
- * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- * This program 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- *
- * Additional permission under GNU GPL version 3 section 7:
- *
- * If you modify this program, or any covered work, by linking or
- * combining it with the OpenSSL project's OpenSSL library (or a
- * modified version of that library), containing parts covered by the
- * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
- * grants you additional permission to convey the resulting work.
- * Corresponding Source for a non-source form of such a combination
- * shall include the source code for the parts of OpenSSL used as well
- * as that of the covered work.
- */
-#ifndef AUDIO_RTP_SESSION_H_
-#define AUDIO_RTP_SESSION_H_
-
-#include "audio_rtp_stream.h"
-#include "dtmf_event.h"
-#include "ip_utils.h"
-#include "noncopyable.h"
-#include "logger.h"
-#include "threadloop.h"
-
-#include <ccrtp/rtp.h>
-#include <ccrtp/formats.h>
-
-#include <chrono>
-
-class SIPCall;
-
-namespace ring {
-
-class AudioCodec;
-class AudioRtpSession;
-
-class CachedAudioRtpState;
-
-class AudioRtpSession {
- public:
- /**
- * Constructor
- * @param sipcall The pointer on the SIP call
- */
- AudioRtpSession(SIPCall &sipcall, ost::RTPDataQueue &queue);
- virtual ~AudioRtpSession();
-
- void updateSessionMedia(const std::vector<AudioCodec*> &audioCodecs);
-
- void startRtpThreads(const std::vector<AudioCodec*> &audioCodecs);
-
- void putDtmfEvent(char digit);
- bool hasDTMFPending() const {
- return not dtmfQueue_.empty();
- }
-
- int getDtmfPayloadType() const {
- return dtmfPayloadType_;
- }
-
- void setDtmfPayloadType(int pt) {
- dtmfPayloadType_ = pt;
- }
-
- int getTransportRate() const;
-
- /**
- * Used mostly when receiving a reinvite
- */
- void updateDestinationIpAddress();
-
- virtual int getIncrementForDTMF() const;
-
- virtual std::vector<long>
- getSocketDescriptors() const = 0;
-
- virtual CachedAudioRtpState * saveState() const;
- virtual void restoreState(const CachedAudioRtpState &state);
-
- private:
- bool isStarted_;
-
- void prepareRtpReceiveThread(const std::vector<AudioCodec*> &audioCodecs);
- /**
- * Set the audio codec for this RTP session
- */
- void setSessionMedia(const std::vector<AudioCodec*> &codec);
-
- protected:
- bool onRTPPacketRecv(ost::IncomingRTPPkt&);
-
- ost::RTPDataQueue &queue_;
- SIPCall &call_;
-
- /**
- * Timestamp for this session
- */
- int timestamp_;
-
- /**
- * Timestamp incrementation value based on codec period length (framesize)
- * except for G722 which require a 8 kHz incrementation.
- */
- int timestampIncrement_;
-
- AudioRtpStream rtpStream_;
-
- private:
-
- /**
- * Rate at which the transport layer handle packets, should be
- * synchronized with codec requirements.
- */
- unsigned int transportRate_;
-
- NON_COPYABLE(AudioRtpSession);
-
- /**
- * Start ccRTP thread loop
- * This thread will send AND receive rtp packets
- */
- virtual void startRTPLoop() = 0;
-
- /**
- * Send DTMF over RTP (RFC2833).
- * The timestamp and sequence number must be incremented
- * as if it was microphone audio. This function change the payload type
- * of the rtp session, send the appropriate DTMF digit using
- * this payload, discard corresponding data from RingBufferPool and get
- * back the codec payload for further audio processing.
- */
- void sendDtmfEvent();
-
- /**
- * Send encoded data to peer
- */
- virtual size_t sendMicData();
-
- /**
- * Set RTP Sockets send/receive timeouts
- */
- void setSessionTimeouts();
-
- /**
- * Receive data from peer
- */
- void receiveSpeakerData();
-
- /**
- * Used by loop_ */
- void process();
-
- // Main destination address for this rtp session.
- // Stored in case of reINVITE, which may require to forget
- // this destination and update a new one.
- IpAddr remoteIp_;
-
- unsigned rxLastSeqNum_;
-#ifdef RTP_DEBUG
- std::chrono::high_resolution_clock::time_point rxLast_;
- std::vector<double> rxJitters_;
- unsigned jitterReportInterval_;
-#endif
-
- std::list<DTMFEvent> dtmfQueue_;
- int dtmfPayloadType_;
-
- // this must be last to ensure that it's destroyed first
- ThreadLoop loop_;
-};
-}
-#endif // AUDIO_RTP_SESSION_H__
-
diff --git a/daemon/src/audio/audiortp/audio_rtp_stream.cpp b/daemon/src/audio/audiortp/audio_rtp_stream.cpp
deleted file mode 100644
index 15f2f8403a8848aea42c18deb188d88c8ba822db..0000000000000000000000000000000000000000
--- a/daemon/src/audio/audiortp/audio_rtp_stream.cpp
+++ /dev/null
@@ -1,483 +0,0 @@
-/*
- * Copyright (C) 2004-2015 Savoir-Faire Linux Inc.
- * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
- * Author: Adrien Beraud <adrien.beraud@wisdomvibes.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- * This program 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- *
- * Additional permission under GNU GPL version 3 section 7:
- *
- * If you modify this program, or any covered work, by linking or
- * combining it with the OpenSSL project's OpenSSL library (or a
- * modified version of that library), containing parts covered by the
- * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
- * grants you additional permission to convey the resulting work.
- * Corresponding Source for a non-source form of such a combination
- * shall include the source code for the parts of OpenSSL used as well
- * as that of the covered work.
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include "audio_rtp_stream.h"
-
-#include "audio/audiolayer.h"
-#include "audio/resampler.h"
-#include "audio/ringbufferpool.h"
-#include "audio/ringbuffer.h"
-
-#if HAVE_SPEEXDSP
-#include "audio/dsp.h"
-#endif
-
-#include "manager.h"
-#include "logger.h"
-
-#include <fstream>
-#include <algorithm>
-#include <cassert>
-
-namespace ring {
-
-AudioRtpStream::AudioRtpStream(const std::string &id) :
- id_(id)
- , encoder_(AudioFormat::MONO())
- , decoder_(AudioFormat::MONO())
- , audioCodecs_()
- , codecEncMutex_(), codecDecMutex_()
- , hasDynamicPayloadType_(false)
- , rawBuffer_(RAW_BUFFER_SIZE, AudioFormat::MONO())
- , micData_(RAW_BUFFER_SIZE, AudioFormat::MONO())
- , encodedData_()
- , dead_(false)
- , currentEncoderIndex_(0)
- , currentDecoderIndex_(0)
- , warningInterval_(0)
- , plcCachePool_()
- , plcPool_(nullptr)
- , plcDec_()
-{
- ringbuffer_ = Manager::instance().getRingBufferPool().createRingBuffer(id_);
- pj_caching_pool_init(&plcCachePool_, &pj_pool_factory_default_policy, 0);
- plcPool_ = pj_pool_create(&plcCachePool_.factory, "plc", 64, 1024, nullptr);
-}
-
-AudioRtpStream::~AudioRtpStream()
-{
- dead_ = true;
- std::lock(codecEncMutex_, codecDecMutex_);
- deleteCodecs();
- codecEncMutex_.unlock();
- codecDecMutex_.unlock();
- plcDec_.clear();
- pj_pool_release(plcPool_);
- pj_caching_pool_destroy(&plcCachePool_);
-}
-
-// Call from processData*
-bool AudioRtpStream::isDead()
-{
- return dead_;
-}
-
-ring::AudioCodec *
-AudioRtpStream::getCurrentEncoder() const
-{
- if (audioCodecs_.empty() or currentEncoderIndex_ >= audioCodecs_.size()) {
- RING_ERR("No codec found");
- return nullptr;
- }
-
- return audioCodecs_[currentEncoderIndex_];
-}
-
-ring::AudioCodec *
-AudioRtpStream::getCurrentDecoder() const
-{
- if (audioCodecs_.empty() or currentDecoderIndex_ >= audioCodecs_.size()) {
- RING_ERR("No codec found");
- return nullptr;
- }
-
- return audioCodecs_[currentDecoderIndex_];
-}
-
-void
-AudioRtpStream::deleteCodecs()
-{
- for (auto &i : audioCodecs_)
- delete i;
-
- audioCodecs_.clear();
-}
-
-// Tries to change decoder (only) if we receive an unexpected payload type that we previously said
-// we could decode.
-bool AudioRtpStream::tryToSwitchDecoder(int newPt)
-{
- std::lock_guard<std::mutex> lock(codecDecMutex_);
- for (unsigned i=0, n=audioCodecs_.size(); i<n; i++) {
- auto codec = audioCodecs_[i];
- if (codec == nullptr || codec->getPayloadType() != newPt) continue;
- AudioFormat f = Manager::instance().getRingBufferPool().getInternalAudioFormat();
- codec->setOptimalFormat(f.sample_rate, f.nb_channels);
- decoder_.payloadType_ = codec->getPayloadType();
- decoder_.frameSize_ = codec->getFrameSize();
- decoder_.format_.sample_rate = codec->getCurrentClockRate();
- decoder_.format_.nb_channels = codec->getCurrentChannels();
- hasDynamicPayloadType_ = codec->hasDynamicPayload();
- resetDecoderPLC(codec);
- currentDecoderIndex_ = i; // FIXME: this is not reliable
- RING_DBG("Switched payload type to %d", newPt);
- return true;
- }
- RING_ERR("Could not switch payload types");
- return false;
-}
-
-void AudioRtpStream::resetDecoderPLC(const ring::AudioCodec * codec)
-{
- if (!plcPool_) return;
- pj_pool_reset(plcPool_);
- plcDec_.clear();
- if (not codec->supportsPacketLossConcealment()) {
- plcDec_.insert(plcDec_.begin(), decoder_.format_.nb_channels, nullptr);
- for (unsigned i = 0; i < decoder_.format_.nb_channels; i++)
- pjmedia_plc_create(plcPool_, decoder_.format_.sample_rate, decoder_.frameSize_, 0, &plcDec_[i]);
- }
-}
-
-AudioRtpContext::AudioRtpContext(AudioFormat f) :
- fadeFactor_(0.)
- , payloadType_(0)
- , frameSize_(0)
- , format_(f)
- , resampledData_(0, AudioFormat::MONO())
- , resampler_(nullptr)
-#if HAVE_SPEEXDSP
- , dspMutex_()
- , dsp_()
-#endif
-{}
-
-AudioRtpContext::~AudioRtpContext()
-{
-#if HAVE_SPEEXDSP
- std::lock_guard<std::mutex> lock(dspMutex_);
- dsp_.reset(nullptr);
-#endif
-}
-
-void AudioRtpContext::resetResampler()
-{
- // initialize resampler using AudioLayer's sampling rate
- // (internal buffers initialized with maximal sampling rate and frame size)
- resampler_.reset(new Resampler(format_.sample_rate));
-}
-
-void AudioRtpContext::fadeIn(AudioBuffer& buf)
-{
- if (fadeFactor_ >= 1.0)
- return;
- // http://en.wikipedia.org/wiki/Smoothstep
- const double gain = fadeFactor_ * fadeFactor_ * (3. - 2. * fadeFactor_);
- buf.applyGain(gain);
- fadeFactor_ += buf.size() / (double) format_.sample_rate;
-}
-
-#if HAVE_SPEEXDSP
-void AudioRtpContext::resetDSP()
-{
- std::lock_guard<std::mutex> lock(dspMutex_);
- assert(frameSize_);
- dsp_.reset(new DSP(frameSize_, format_.nb_channels, format_.sample_rate));
-}
-
-void AudioRtpContext::applyDSP(AudioBuffer &buffer)
-{
- const bool denoise = Manager::instance().audioPreference.getNoiseReduce();
- const bool agc = Manager::instance().audioPreference.isAGCEnabled();
-
- if (denoise or agc) {
- std::lock_guard<std::mutex> lock(dspMutex_);
- if (!dsp_)
- return;
-
- if (denoise)
- dsp_->enableDenoise();
- else
- dsp_->disableDenoise();
-
- if (agc)
- dsp_->enableAGC();
- else
- dsp_->disableAGC();
-
- dsp_->process(buffer, frameSize_);
- }
-}
-#endif
-
-
-void AudioRtpStream::setRtpMedia(const std::vector<AudioCodec*> &audioCodecs)
-{
- std::lock(codecEncMutex_, codecDecMutex_);
-
- deleteCodecs();
- // Set various codec info to reduce indirection
- audioCodecs_ = audioCodecs;
-
- if (audioCodecs.empty()) {
- codecEncMutex_.unlock();
- codecDecMutex_.unlock();
- RING_ERR("Audio codecs empty");
- return;
- }
-
- // FIXME: assuming right encoder/decoder are first?
- currentEncoderIndex_ = currentDecoderIndex_ = 0;
- AudioCodec& codec = *audioCodecs[currentEncoderIndex_];
-
- AudioFormat f = Manager::instance().getRingBufferPool().getInternalAudioFormat();
- codec.setOptimalFormat(f.sample_rate, f.nb_channels);
-
- const int pt = codec.getPayloadType();
- encoder_.payloadType_ = decoder_.payloadType_ = pt;
- encoder_.frameSize_ = decoder_.frameSize_ = codec.getFrameSize();
-
- AudioFormat codecFormat(codec.getCurrentClockRate(), codec.getCurrentChannels());
- if (codecFormat != decoder_.format_ or codecFormat != encoder_.format_) {
- encoder_.format_ = decoder_.format_ = codecFormat;
-#if HAVE_SPEEXDSP
- resetDSP();
-#endif
- }
- Manager::instance().audioFormatUsed(codecFormat);
- hasDynamicPayloadType_ = codec.hasDynamicPayload();
- codecEncMutex_.unlock();
-
- resetDecoderPLC(audioCodecs[currentDecoderIndex_]);
- codecDecMutex_.unlock();
-}
-
-void AudioRtpStream::initBuffers()
-{
- encoder_.resetResampler();
- decoder_.resetResampler();
-}
-
-#if HAVE_SPEEXDSP
-void AudioRtpStream::resetDSP()
-{
- encoder_.resetDSP();
- decoder_.resetDSP();
-}
-#endif
-
-bool AudioRtpStream::waitForDataEncode(const std::chrono::milliseconds& max_wait) const
-{
- const auto &mainBuffer = Manager::instance().getRingBufferPool();
- const AudioFormat mainBuffFormat = mainBuffer.getInternalAudioFormat();
- const double resampleFactor = (double) mainBuffFormat.sample_rate / encoder_.format_.sample_rate;
- const size_t samplesToGet = resampleFactor * encoder_.frameSize_;
-
- return mainBuffer.waitForDataAvailable(id_, samplesToGet, max_wait);
-}
-
-size_t AudioRtpStream::processDataEncode()
-{
- if (isDead())
- return 0;
-
- AudioFormat mainBuffFormat = Manager::instance().getRingBufferPool().getInternalAudioFormat();
-
- double resampleFactor = (double) mainBuffFormat.sample_rate / encoder_.format_.sample_rate;
-
- // compute nb of byte to get corresponding to 1 audio frame
- const size_t samplesToGet = resampleFactor * encoder_.frameSize_;
-
- if (Manager::instance().getRingBufferPool().availableForGet(id_) < samplesToGet)
- return 0;
-
- micData_.setFormat(mainBuffFormat);
- micData_.resize(samplesToGet);
- const size_t samples = Manager::instance().getRingBufferPool().getData(micData_, id_);
-
- if (samples != samplesToGet) {
- RING_ERR("Asked for %d samples from bindings on call '%s', got %d",
- samplesToGet, id_.c_str(), samples);
- return 0;
- }
-
- AudioBuffer *out = &micData_;
- if (encoder_.format_.sample_rate != mainBuffFormat.sample_rate) {
- if (!encoder_.resampler_) {
- RING_ERR("Resampler already destroyed");
- return 0;
- }
- encoder_.resampledData_.setChannelNum(mainBuffFormat.nb_channels);
- encoder_.resampledData_.setSampleRate(encoder_.format_.sample_rate);
- encoder_.resampler_->resample(micData_, encoder_.resampledData_);
- out = &encoder_.resampledData_;
- }
- if (encoder_.format_.nb_channels != mainBuffFormat.nb_channels)
- out->setChannelNum(encoder_.format_.nb_channels, true);
-
- encoder_.fadeIn(*out);
-#if HAVE_SPEEXDSP
- encoder_.applyDSP(*out);
-#endif
- {
- std::lock_guard<std::mutex> lock(codecEncMutex_);
- auto codec = getCurrentEncoder();
- if (!codec) {
- RING_ERR("Audio codec already destroyed");
- return 0;
- }
-
- const auto codecFrameSize = codec->getFrameSize();
- if (codecFrameSize > out->frames()) {
- // PCM too small (underflow), add zero padding to avoid reading past
- // end of buffer when encoding, for every channel
- out->resize(codecFrameSize);
- }
-
- size_t encoded = codec->encode(out->getData(), encodedData_.data(), encodedData_.size());
- return encoded;
- }
-}
-
-
-void AudioRtpStream::processDataDecode(unsigned char *spkrData, size_t size, int payloadType)
-{
- if (isDead())
- return;
-
- const int decPt = decoder_.payloadType_;
- if (decPt != payloadType) {
- const bool switched = tryToSwitchDecoder(payloadType);
-
- if (not switched) {
- if (!warningInterval_) {
- warningInterval_ = 250;
- RING_WARN("Invalid payload type %d, expected %d", payloadType, decPt);
- }
-
- warningInterval_--;
- return;
- }
- }
-
- rawBuffer_.setFormat(decoder_.format_);
- rawBuffer_.resize(RAW_BUFFER_SIZE);
- {
- std::lock_guard<std::mutex> lock(codecDecMutex_);
- auto codec = getCurrentDecoder();
- if (!codec) {
- RING_ERR("Audio codec already destroyed");
- return;
- }
- if (spkrData) { // Packet is available
- int decoded = codec->decode(rawBuffer_.getData(), spkrData, size);
- rawBuffer_.resize(decoded);
- if (not plcDec_.empty()) {
- for (unsigned i = 0; i < decoder_.format_.nb_channels; ++i) {
- pjmedia_plc_save(plcDec_[i], rawBuffer_.getChannel(i)->data());
- }
- }
- } else if (plcDec_.empty()) { // Packet loss concealment using codec
- int decoded = codec->decode(rawBuffer_.getData());
- rawBuffer_.resize(decoded);
- } else { // Generic PJSIP Packet loss concealment using codec
- rawBuffer_.resize(decoder_.frameSize_);
- for (unsigned i = 0; i < decoder_.format_.nb_channels; ++i) {
- pjmedia_plc_generate(plcDec_[i], rawBuffer_.getChannel(i)->data());
- }
- }
- }
-
-#if HAVE_SPEEXDSP
- decoder_.applyDSP(rawBuffer_);
-#endif
-
- decoder_.fadeIn(rawBuffer_);
-
- // test if resampling or up/down-mixing is required
- AudioBuffer *out = &rawBuffer_;
- AudioFormat decFormat = out->getFormat();
- AudioFormat mainBuffFormat = Manager::instance().getRingBufferPool().getInternalAudioFormat();
- if (decFormat.sample_rate != mainBuffFormat.sample_rate) {
- if (!decoder_.resampler_) {
- RING_ERR("Resampler already destroyed");
- return;
- }
- decoder_.resampledData_.setChannelNum(decFormat.nb_channels);
- decoder_.resampledData_.setSampleRate(mainBuffFormat.sample_rate);
- decoder_.resampler_->resample(rawBuffer_, decoder_.resampledData_);
- out = &decoder_.resampledData_;
- }
- if (decFormat.nb_channels != mainBuffFormat.nb_channels)
- out->setChannelNum(mainBuffFormat.nb_channels, true);
-
- ringbuffer_->put(*out);
-}
-
-bool
-AudioRtpStream::codecsDiffer(const std::vector<AudioCodec*> &codecs) const
-{
- const std::vector<AudioCodec*> ¤t = audioCodecs_;
-
- if (codecs.size() != current.size())
- return true;
-
- for (const auto &i : codecs) {
- if (!i)
- continue;
-
- bool matched = false;
-
- for (const auto &j : current)
- if ((matched = i->getPayloadType() == j->getPayloadType()))
- break;
-
- if (not matched)
- return true;
- }
-
- return false;
-}
-
-int
-AudioRtpStream::getEncoderPayloadType() const
-{
- return encoder_.payloadType_;
-}
-
-int
-AudioRtpStream::getEncoderFrameSize() const
-{
- return encoder_.frameSize_;
-}
-
-
-int
-AudioRtpStream::getTransportRate() const
-{
- const int transportRate = encoder_.frameSize_ / encoder_.format_.sample_rate / 1000;
- return transportRate > 0 ? transportRate : 20;
-}
-
-}
diff --git a/daemon/src/audio/audiortp/audio_rtp_stream.h b/daemon/src/audio/audiortp/audio_rtp_stream.h
deleted file mode 100644
index 473f61e7251dc64f7c5d51aa7aca67ed611e9fe4..0000000000000000000000000000000000000000
--- a/daemon/src/audio/audiortp/audio_rtp_stream.h
+++ /dev/null
@@ -1,181 +0,0 @@
-/*
- * Copyright (C) 2004-2015 Savoir-Faire Linux Inc.
- * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
- * Author: Adrien Beraud <adrien.beraud@wisdomvibes.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- * This program 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- *
- * Additional permission under GNU GPL version 3 section 7:
- *
- * If you modify this program, or any covered work, by linking or
- * combining it with the OpenSSL project's OpenSSL library (or a
- * modified version of that library), containing parts covered by the
- * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
- * grants you additional permission to convey the resulting work.
- * Corresponding Source for a non-source form of such a combination
- * shall include the source code for the parts of OpenSSL used as well
- * as that of the covered work.
- */
-
-#ifndef AUDIO_RTP_STREAM_H__
-#define AUDIO_RTP_STREAM_H__
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include "audio/codecs/audiocodec.h"
-#include "audio/audiobuffer.h"
-#include "noncopyable.h"
-
-#include <pjmedia/plc.h>
-#include <pj/pool.h>
-
-#include <array>
-#include <mutex>
-#include <memory>
-#include <atomic>
-#include <cstddef>
-
-namespace ring {
-
-class Resampler;
-class DSP;
-class RingBuffer;
-
-class AudioRtpContext {
- public:
- AudioRtpContext(AudioFormat f);
- ~AudioRtpContext();
-
- /**
- * Ramp in audio data to avoid audio click from peer
- */
- void fadeIn(AudioBuffer& buf);
-
- private:
- NON_COPYABLE(AudioRtpContext);
- static constexpr double FADEIN_STEP_SIZE = 4.0;
-
- void resetResampler();
- double fadeFactor_;
- int payloadType_;
- int frameSize_;
- AudioFormat format_;
- AudioBuffer resampledData_;
- std::unique_ptr<Resampler> resampler_;
-#if HAVE_SPEEXDSP
- void resetDSP();
- void applyDSP(AudioBuffer &rawBuffer);
- std::mutex dspMutex_;
- std::unique_ptr<DSP> dsp_;
-#endif
-
- friend class AudioRtpStream;
-};
-
-/**
- * Class meant to store internal data in order to encode/decode,
- * resample, process, and packetize audio streams.
- */
-class AudioRtpStream {
- public:
- AudioRtpStream(const std::string &id);
- virtual ~AudioRtpStream();
- void initBuffers();
-
- /**
- * Set the list of codecs supported by this stream.
- * The codec used for encoding must be first.
- */
- void setRtpMedia(const std::vector<AudioCodec*> &codecs);
-
- /**
- * Decode audio data received from peer
- */
- void processDataDecode(uint8_t *spkrData, size_t size, int payloadType);
-
- /**
- * Wait for available data to be encoded from RingBufferPool.
- */
- bool waitForDataEncode(const std::chrono::milliseconds& max_wait) const;
-
- /**
- * Encode audio data from RingBufferPool
- * @return size of encoded data, in bytes.
- */
- size_t processDataEncode();
-
- bool hasDynamicPayload() const {
- return hasDynamicPayloadType_;
- }
-
- const uint8_t *getMicDataEncoded() const {
- return encodedData_.data();
- }
-
- int getEncoderPayloadType() const;
- int getEncoderFrameSize() const;
- AudioFormat getEncoderFormat() const { return encoder_.format_; }
-
-#if HAVE_SPEEXDSP
- void initDSP();
- void resetDSP();
-#endif
- bool codecsDiffer(const std::vector<AudioCodec*> &codecs) const;
- int getTransportRate() const;
-
- private:
- const std::string id_;
-
- AudioRtpContext encoder_;
- AudioRtpContext decoder_;
-
- std::shared_ptr<RingBuffer> ringbuffer_{};
-
- void deleteCodecs();
- ring::AudioCodec* getCurrentEncoder() const;
- ring::AudioCodec* getCurrentDecoder() const;
-
- // Decoder-specific methods
- bool tryToSwitchDecoder(int newPt);
- void resetDecoderPLC(const ring::AudioCodec *);
-
- std::vector<AudioCodec*> audioCodecs_;
- std::mutex codecEncMutex_;
- std::mutex codecDecMutex_;
- // these will have the same value unless we are sending
- // a different codec than we are receiving (asymmetric RTP)
- bool hasDynamicPayloadType_;
- // FIXME: probably need one for pre-encoder data, one for post-decoder data
- AudioBuffer rawBuffer_, micData_;
- std::array<uint8_t, RAW_BUFFER_SIZE> encodedData_;
-
- bool isDead();
-
- NON_COPYABLE(AudioRtpStream);
- std::atomic<bool> dead_;
- size_t currentEncoderIndex_;
- size_t currentDecoderIndex_;
- int warningInterval_;
-
- pj_caching_pool plcCachePool_;
- pj_pool_t * plcPool_;
- // PLC instances, one per channel
- std::vector<pjmedia_plc*> plcDec_;
-
-};
-}
-
-#endif // AUDIO_RTP_STREAM_H__
diff --git a/daemon/src/audio/audiortp/audio_srtp_session.cpp b/daemon/src/audio/audiortp/audio_srtp_session.cpp
deleted file mode 100644
index 247727ba96cc7ec3e6b9d052860156eca223c59d..0000000000000000000000000000000000000000
--- a/daemon/src/audio/audiortp/audio_srtp_session.cpp
+++ /dev/null
@@ -1,320 +0,0 @@
-/*
- * Copyright (C) 2004-2015 Savoir-Faire Linux Inc.
- * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
- * Author: Pierre-Luc Bacon <pierre-luc.bacon@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- * This program 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- *
- * Additional permission under GNU GPL version 3 section 7:
- *
- * If you modify this program, or any covered work, by linking or
- * combining it with the OpenSSL project's OpenSSL library (or a
- * modified version of that library), containing parts covered by the
- * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
- * grants you additional permission to convey the resulting work.
- * Corresponding Source for a non-source form of such a combination
- * shall include the source code for the parts of OpenSSL used as well
- * as that of the covered work.
- */
-#include "audio_srtp_session.h"
-#include "logger.h"
-#include "array_size.h"
-#include "base64.h"
-
-#include <algorithm>
-#include <random>
-
-#include <cstdio>
-#include <cstring>
-#include <cerrno>
-
-#define STRINGIFY(x) #x
-#define TOSTRING(x) STRINGIFY(x)
-#define THROW_ERROR(EXCEPTION_CLASS, M) throw EXCEPTION_CLASS(__FILE__ ":" TOSTRING(__LINE__) ":" M)
-
-namespace ring {
-
-static std::string
-encodeBase64(unsigned char *input, int length)
-{
- size_t output_length;
- uint8_t *encoded_str = sfl_base64_encode(input, length, &output_length);
- if (!encoded_str)
- THROW_ERROR(AudioSrtpException, "Out of memory for base64 operation");
- std::string output((const char *)encoded_str, output_length);
- free(encoded_str);
- return output;
-}
-
-static std::string
-decodeBase64(unsigned char *input, int length)
-{
- size_t output_length;
- uint8_t *decoded_str = sfl_base64_decode(input, length, &output_length);
- if (!decoded_str)
- THROW_ERROR(AudioSrtpException, "Out of memory for base64 operation");
- std::string output((const char *)decoded_str, output_length);
- free(decoded_str);
- return output;
-}
-
-// Fills the array dest with length random bytes
-static void
-bufferFillMasterKey(std::vector<uint8_t>& dest)
-{
- RING_DBG("Init local master key");
- std::uniform_int_distribution<uint8_t> rand_byte(0, 255);
-
- // Fill the key
- {
- std::random_device rdev;
- std::generate(dest.begin(), dest.end(), std::bind(rand_byte, std::ref(rdev)));
- }
-}
-
-// Fills the array dest with length random bytes
-static void
-bufferFillMasterSalt(std::vector<uint8_t>& dest)
-{
- RING_DBG("Init local master salt");
- std::uniform_int_distribution<uint8_t> rand_byte(0, 255);
-
- // Fill the key
- {
- std::random_device rdev;
- std::generate(dest.begin(), dest.end(), std::bind(rand_byte, std::ref(rdev)));
- }
-}
-
-static const unsigned MAX_MASTER_KEY_LENGTH = 16;
-static const unsigned MAX_MASTER_SALT_LENGTH = 14;
-
-AudioSrtpSession::AudioSrtpSession(SIPCall &call) :
- AudioSymmetricRtpSession(call),
- remoteCryptoCtx_(),
- localCryptoCtx_(),
- localCryptoSuite_(0),
- remoteCryptoSuite_(0),
- localMasterKey_(MAX_MASTER_KEY_LENGTH),
- localMasterSalt_(MAX_MASTER_SALT_LENGTH),
- remoteMasterKey_(MAX_MASTER_KEY_LENGTH),
- remoteMasterSalt_(MAX_MASTER_SALT_LENGTH),
- remoteOfferIsSet_(false)
-{}
-
-AudioSrtpSession::~AudioSrtpSession()
-{
- if (remoteCryptoCtx_)
- removeInQueueCryptoContext(remoteCryptoCtx_);
- if (localCryptoCtx_)
- removeOutQueueCryptoContext(localCryptoCtx_);
-}
-
-void AudioSrtpSession::initLocalCryptoInfo()
-{
- RING_DBG("AudioSrtpSession: Set cryptographic info for this rtp session");
-
- // Initialize local Crypto context
- initializeLocalMasterKey();
- initializeLocalMasterSalt();
- initializeLocalCryptoContext();
-
- // Set local crypto context in ccrtp
- localCryptoCtx_->deriveSrtpKeys(0);
-
- setOutQueueCryptoContext(localCryptoCtx_);
-}
-
-void AudioSrtpSession::initLocalCryptoInfoOnOffhold()
-{
- RING_DBG("AudioSrtpSession: Set cryptographic info for this rtp session");
-
- // Initialize local Crypto context
- initializeLocalCryptoContext();
-
- // Set local crypto context in ccrtp
- localCryptoCtx_->deriveSrtpKeys(0);
-
- setOutQueueCryptoContext(localCryptoCtx_);
-}
-
-std::vector<std::string> AudioSrtpSession::getLocalCryptoInfo()
-{
- RING_DBG("Get Cryptographic info from this rtp session");
-
- std::vector<std::string> crypto_vector;
-
- // @TODO we should return a vector containing supported
- // cryptographic context tagged 1, 2, 3...
- std::string tag = "1";
-
- std::string crypto_suite(ring::CryptoSuites[localCryptoSuite_].name);
-
- // srtp keys formated as the following as the following
- // inline:keyParameters|keylifetime|MasterKeyIdentifier
- // inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32
- std::string srtp_keys = "inline:";
- srtp_keys += getBase64ConcatenatedKeys();
- // srtp_keys.append("|2^20|1:32");
-
- // generate crypto attribute
- std::string crypto_attr = tag.append(" ");
- crypto_attr += crypto_suite.append(" ");
- crypto_attr += srtp_keys;
-
- RING_DBG("%s", crypto_attr.c_str());
-
- crypto_vector.push_back(crypto_attr);
-
- return crypto_vector;
-}
-
-void AudioSrtpSession::setRemoteCryptoInfo(const ring::SdesNegotiator& nego)
-{
- if (not remoteOfferIsSet_) {
- // Use second crypto suite if key length is 32 bit, default is 80;
- if (nego.getAuthTagLength() == "32") {
- localCryptoSuite_ = 1;
- remoteCryptoSuite_ = 1;
- }
-
- // decode keys
- unBase64ConcatenatedKeys(nego.getKeyInfo());
-
- // init crypto content in Srtp session
- initializeRemoteCryptoContext();
-
- if (remoteCryptoCtx_) {
- setInQueueCryptoContext(remoteCryptoCtx_);
- }
-
- remoteOfferIsSet_ = true;
- }
-}
-
-static const size_t BITS_PER_BYTE = 8;
-
-void AudioSrtpSession::initializeLocalMasterKey()
-{
- localMasterKey_.resize(ring::CryptoSuites[localCryptoSuite_].masterKeyLength / BITS_PER_BYTE);
- bufferFillMasterKey(localMasterKey_);
-}
-
-void AudioSrtpSession::initializeLocalMasterSalt()
-{
- localMasterSalt_.resize(ring::CryptoSuites[localCryptoSuite_].masterSaltLength / BITS_PER_BYTE);
- bufferFillMasterSalt(localMasterSalt_);
-}
-
-std::string AudioSrtpSession::getBase64ConcatenatedKeys()
-{
- RING_DBG("Get base64 concatenated keys");
-
- // compute concatenated master and salt length
- std::vector<uint8> concatKeys;
- concatKeys.reserve(localMasterKey_.size() + localMasterSalt_.size());
-
- // concatenate keys
- concatKeys.insert(concatKeys.end(), localMasterKey_.begin(), localMasterKey_.end());
- concatKeys.insert(concatKeys.end(), localMasterSalt_.begin(), localMasterSalt_.end());
-
- // encode concatenated keys in base64
- return encodeBase64(concatKeys.data(), concatKeys.size());
-}
-
-void AudioSrtpSession::unBase64ConcatenatedKeys(std::string base64keys)
-{
- remoteMasterKey_.resize(ring::CryptoSuites[remoteCryptoSuite_].masterKeyLength / BITS_PER_BYTE);
- remoteMasterSalt_.resize(ring::CryptoSuites[remoteCryptoSuite_].masterSaltLength / BITS_PER_BYTE);
-
- // decode concatenated binary keys
- std::string output(decodeBase64((uint8_t *)base64keys.data(), base64keys.size()));
-
- // copy master and slt respectively
- const std::string::iterator key_end = output.begin() + remoteMasterKey_.size();
-
- if (key_end > output.end() or
- (size_t) std::distance(key_end, output.end()) > remoteMasterSalt_.size())
- THROW_ERROR(AudioSrtpException, "Out of bounds copy");
-
- std::copy(output.begin(), key_end, remoteMasterKey_.begin());
- std::copy(key_end, output.end(), remoteMasterSalt_.begin());
-
-}
-
-void AudioSrtpSession::initializeRemoteCryptoContext()
-{
- RING_DBG("Initialize remote crypto context");
-
- const CryptoSuiteDefinition &crypto = ring::CryptoSuites[remoteCryptoSuite_];
-
- if (remoteCryptoCtx_)
- removeInQueueCryptoContext(remoteCryptoCtx_);
-
- remoteCryptoCtx_ = new ost::CryptoContext(0x0,
- 0, // roc,
- 0L, // keydr,
- SrtpEncryptionAESCM,
- SrtpAuthenticationSha1Hmac,
- remoteMasterKey_.data(),
- remoteMasterKey_.size(),
- remoteMasterSalt_.data(),
- remoteMasterSalt_.size(),
- crypto.encryptionKeyLength / BITS_PER_BYTE,
- crypto.srtpAuthKeyLength / BITS_PER_BYTE,
- crypto.masterSaltLength / BITS_PER_BYTE,
- crypto.srtpAuthTagLength / BITS_PER_BYTE);
-}
-
-void AudioSrtpSession::initializeLocalCryptoContext()
-{
- RING_DBG("Initialize local crypto context");
-
- const CryptoSuiteDefinition &crypto = ring::CryptoSuites[localCryptoSuite_];
-
- if (localCryptoCtx_)
- removeOutQueueCryptoContext(localCryptoCtx_);
-
- localCryptoCtx_ = new ost::CryptoContext(OutgoingDataQueue::getLocalSSRC(),
- 0, // roc,
- 0L, // keydr,
- SrtpEncryptionAESCM,
- SrtpAuthenticationSha1Hmac,
- localMasterKey_.data(),
- localMasterKey_.size(),
- localMasterSalt_.data(),
- localMasterSalt_.size(),
- crypto.encryptionKeyLength / BITS_PER_BYTE,
- crypto.srtpAuthKeyLength / BITS_PER_BYTE,
- crypto.masterSaltLength / BITS_PER_BYTE,
- crypto.srtpAuthTagLength / BITS_PER_BYTE);
-}
-
-CachedAudioRtpState *
-AudioSrtpSession::saveState() const
-{
- return new CachedAudioRtpState(localMasterKey_, localMasterSalt_);
-}
-
-CachedAudioRtpState::CachedAudioRtpState(const std::vector<uint8> &key, const std::vector<uint8> &salt) : key_(key), salt_(salt)
-{}
-
-void
-AudioSrtpSession::restoreState(const CachedAudioRtpState &state)
-{
- localMasterKey_ = state.key_;
- localMasterSalt_ = state.salt_;
-}
-
-}
diff --git a/daemon/src/audio/audiortp/audio_srtp_session.h b/daemon/src/audio/audiortp/audio_srtp_session.h
deleted file mode 100644
index 459b976ce18a63c33326006509aa6167e01763cb..0000000000000000000000000000000000000000
--- a/daemon/src/audio/audiortp/audio_srtp_session.h
+++ /dev/null
@@ -1,208 +0,0 @@
-/*
- * Copyright (C) 2004-2015 Savoir-Faire Linux Inc.
- * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
- * Author: Pierre-Luc Bacon <pierre-luc.bacon@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- * This program 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- *
- * Additional permission under GNU GPL version 3 section 7:
- *
- * If you modify this program, or any covered work, by linking or
- * combining it with the OpenSSL project's OpenSSL library (or a
- * modified version of that library), containing parts covered by the
- * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
- * grants you additional permission to convey the resulting work.
- * Corresponding Source for a non-source form of such a combination
- * shall include the source code for the parts of OpenSSL used as well
- * as that of the covered work.
- */
-#ifndef AUDIO_SRTP_SESSION_H_
-#define AUDIO_SRTP_SESSION_H_
-
-#include "audio_symmetric_rtp_session.h"
-#include "sip/sdes_negotiator.h"
-#include "noncopyable.h"
-
-#include <ccrtp/CryptoContext.h>
-#include <vector>
-
-class SdesNegotiator;
-class SIPCall;
-
-/*
- Table from RFC 4568 6.2. Crypto-Suites, which define key parameters for supported
- cipher suite
-
- +---------------------+-------------+--------------+---------------+
- | |AES_CM_128_ | AES_CM_128_ | F8_128_ |
- | |HMAC_SHA1_80 | HMAC_SHA1_32 | HMAC_SHA1_80 |
- +---------------------+-------------+--------------+---------------+
- | Master key length | 128 bits | 128 bits | 128 bits |
- | Master salt length | 112 bits | 112 bits | 112 bits |
- | SRTP lifetime | 2^48 packets| 2^48 packets | 2^48 packets |
- | SRTCP lifetime | 2^31 packets| 2^31 packets | 2^31 packets |
- | Cipher | AES Counter | AES Counter | AES F8 Mode |
- | | Mode | Mode | |
- | Encryption key | 128 bits | 128 bits | 128 bits |
- | MAC | HMAC-SHA1 | HMAC-SHA1 | HMAC-SHA1 |
- | SRTP auth. tag | 80 bits | 32 bits | 80 bits |
- | SRTCP auth. tag | 80 bits | 80 bits | 80 bits |
- | SRTP auth. key len. | 160 bits | 160 bits | 160 bits |
- | SRTCP auth. key len.| 160 bits | 160 bits | 160 bits |
- +---------------------+-------------+--------------+---------------+
-*/
-
-namespace ring {
-
-struct AudioSrtpException : public std::runtime_error {
- AudioSrtpException(const char *msg) : std::runtime_error(msg) {}
-};
-
-class AudioSrtpSession : public AudioSymmetricRtpSession {
- public:
-
- /**
- * Constructor for this rtp session. The local and remote keys must be properly
- * initialized using initLocalCryptoInfo and setRemoteCryptoInfo respectively.
- */
- AudioSrtpSession(SIPCall &call);
-
- ~AudioSrtpSession();
-
- /**
- * Used to get sdp crypto header to be included in sdp session. This
- * method must be called befor setRemoteCryptoInfo in case of an
- * outgoing call or after in case of an outgoing call.
- */
- std::vector<std::string> getLocalCryptoInfo();
-
- /**
- * Set remote crypto header from incoming sdp offer. It is expected that the
- * local cryptographic context is initialized with mehod
- */
- void setRemoteCryptoInfo(const ring::SdesNegotiator &nego);
-
- /**
- * Init local crypto context for outgoing data
- * this method must be called before sending or receiving an SDP offer.
- * It is required for media negotiation that the local cryptographic
- * context be properly initialized.
- *
- * @return The new local crypto context, to be cached by the caller
- */
- void initLocalCryptoInfo();
-
- /**
- * Initialize crypto context
- */
- void initLocalCryptoInfoOnOffhold();
-
- private:
- NON_COPYABLE(AudioSrtpSession);
-
- CachedAudioRtpState *
- saveState() const;
-
- void restoreState(const CachedAudioRtpState &state);
-
- /**
- * Remote srtp crypto context to be set into incoming data queue.
- * XXX: don't use smart pointers, ccrtp deletes this
- */
- ost::CryptoContext *remoteCryptoCtx_;
-
- /**
- * Local srtp crypto context to be set into outgoing data queue.
- * XXX: don't use smart pointers, ccrtp deletes this
- */
- ost::CryptoContext *localCryptoCtx_;
-
- /**
- * Init local master key according to current crypto context
- * as defined in SdesNegotiator.h
- */
- void initializeLocalMasterKey();
-
- /**
- * Init local master salt according to current crypto context
- * as defined in SdesNegotiator.h
- */
- void initializeLocalMasterSalt();
-
- /**
- * Init remote crypto context in audio srtp session. This method
- * must be called after unBase64ConcatenatedKeys.
- */
- void initializeRemoteCryptoContext();
-
- /**
- * Init local crypto context in audio srtp session. Make sure remote
- * crypto context is set before calling this method for incoming calls.
- */
- void initializeLocalCryptoContext();
-
- /**
- * Used to generate local keys to be included in SDP offer/answer.
- */
- std::string getBase64ConcatenatedKeys();
-
- /**
- * Used to retreive keys from base64 serialization
- */
- void unBase64ConcatenatedKeys(std::string base64keys);
-
- /**
- * Default local crypto suite is AES_CM_128_HMAC_SHA1_80
- */
- int localCryptoSuite_;
-
- /**
- * Remote crypto suite is initialized at AES_CM_128_HMAC_SHA1_80
- */
- int remoteCryptoSuite_;
-
- /**
- * Array to store the local master key
- */
- std::vector<uint8> localMasterKey_;
-
- /**
- * Array to store local master salt
- */
- std::vector<uint8> localMasterSalt_;
-
- std::vector<uint8> remoteMasterKey_;
-
- /**
- * Array to store the remote master salt
- */
- std::vector<uint8> remoteMasterSalt_;
-
- /**
- * Used to make sure remote crypto context not initialized twice.
- */
- bool remoteOfferIsSet_;
-};
-
-class CachedAudioRtpState {
- public:
- CachedAudioRtpState(const std::vector<uint8> &key, const std::vector<uint8> &salt);
- private:
- friend class AudioSrtpSession;
- std::vector<uint8> key_, salt_;
-};
-
-}
-
-#endif // __AUDIO_SRTP_SESSION_H__
diff --git a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp b/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp
deleted file mode 100644
index 222a20d1b3437183ec2755749cc88790539171de..0000000000000000000000000000000000000000
--- a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp
+++ /dev/null
@@ -1,245 +0,0 @@
-/*
- * Copyright (C) 2004-2015 Savoir-Faire Linux Inc.
- * Author: Pierre-Luc Bacon <pierre-luc.bacon@savoirfairelinux.com>
- * Author: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
- * Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
- * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- * This program 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- *
- * Additional permission under GNU GPL version 3 section 7:
- *
- * If you modify this program, or any covered work, by linking or
- * combining it with the OpenSSL project's OpenSSL library (or a
- * modified version of that library), containing parts covered by the
- * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
- * grants you additional permission to convey the resulting work.
- * Corresponding Source for a non-source form of such a combination
- * shall include the source code for the parts of OpenSSL used as well
- * as that of the covered work.
- */
-
-#include "audio_symmetric_rtp_session.h"
-#include "logger.h"
-#include "sip/sipcall.h"
-#include "manager.h"
-#include "client/callmanager.h"
-
-namespace ring {
-
-AudioSymmetricRtpSession::AudioSymmetricRtpSession(SIPCall &call) :
- ost::SymmetricRTPSession(static_cast<ost::IPV4Host>(call.getLocalIp()), call.getLocalAudioPort())
- , AudioRtpSession(call, *this)
-{
- RING_DBG("Setting new RTP session with destination %s:%d",
- call_.getLocalIp().toString().c_str(), call_.getLocalAudioPort());
-}
-
-AudioSymmetricRtpSession::~AudioSymmetricRtpSession()
-{
- // This would normally be invoked in ost::~SymmetricRTPSession() BUT
- // we must stop it here because onRTPPacketRecv() runs in the other thread
- // and uses member variables that are destroyed here
- if (ost::Thread::isRunning()) {
- disableStack();
- ost::Thread::join();
- }
-}
-
-std::vector<long>
-AudioSymmetricRtpSession::getSocketDescriptors() const
-{
- std::vector<long> result;
- result.push_back(dso->getRecvSocket());
- result.push_back(cso->getRecvSocket());
- return result;
-}
-
-void AudioSymmetricRtpSession::startRTPLoop()
-{
- ost::SymmetricRTPSession::startRunning();
-}
-
-// redefined from QueueRTCPManager
-void AudioSymmetricRtpSession::onGotRR(ost::SyncSource& source, ost::RTCPCompoundHandler::RecvReport& RR, uint8 blocks)
-{
- ost::SymmetricRTPSession::onGotRR(source, RR, blocks);
-#ifdef RTP_DEBUG
- RING_DBG("onGotRR");
- RING_DBG("Unpacking %d blocks",blocks);
- for (int i = 0; i < blocks; ++i)
- {
- RING_DBG("fractionLost : %hhu", RR.blocks[i].rinfo.fractionLost);
- RING_DBG("lostMSB : %hhu", RR.blocks[i].rinfo.lostMSB);
- RING_DBG("lostLSW : %hu", RR.blocks[i].rinfo.lostLSW);
- RING_DBG("highestSeqNum : %u", RR.blocks[i].rinfo.highestSeqNum);
- RING_DBG("jitter : %u", RR.blocks[i].rinfo.jitter);
- RING_DBG("lsr : %u", RR.blocks[i].rinfo.lsr);
- RING_DBG("dlsr : %u", RR.blocks[i].rinfo.dlsr);
- }
-#endif
-}
-
-// redefined from QueueRTCPManager
-void AudioSymmetricRtpSession::onGotSR(ost::SyncSource& source, ost::RTCPCompoundHandler::SendReport& SR, uint8 blocks)
-{
-#ifdef RTP_DEBUG
- RING_DBG("onGotSR");
- std::cout << "I got an SR RTCP report from "
- << std::hex << (int)source.getID() << "@"
- << std::dec
- << source.getNetworkAddress() << ":"
- << source.getControlTransportPort() << std::endl;
-#endif
-
- std::map<std::string, int> stats;
- ost::SymmetricRTPSession::onGotSR(source, SR, blocks);
- ost::RTCPSenderInfo report(SR.sinfo);
-
- for (int i = 0; i < blocks; ++i)
- {
- // If this is the first report drop it, stats are not complete
- if(SR.blocks[i].rinfo.lsr == 0 || SR.blocks[i].rinfo.dlsr == 0)
- continue;
-
- ost::RTCPReceiverInfo receiver_report(SR.blocks[i].rinfo);
- /*
- How to calculate RTT (Round Trip delay)
- A : NTP timestamp
- lsr : The middle 32 bits out of 64 in the NTP timestamp (as explained in Section 4) received as part of the most
- recent RTCP sender report (SR) packet from source SSRC_n. If no SR has been received yet, the field is set to zero.
- dlsr : The delay, expressed in units of 1/65536 seconds, between receiving the last SR packet from source SSRC_n and
- sending this reception report block. If no SR packet has been received yet from SSRC_n, the DLSR field is set to zero.
- RTT = A - lsr - dlsr;
- */
-
- // integer portion of timestamp is in least significant 16-bits
- const uint16 timestamp = (uint16) report.getNTPTimestampInt() & 0x0000FFFF;
- // fractional portion of timestamp is in most significant 16-bits
- const uint16 timestampFrac = (uint16) report.getNTPTimestampFrac() >> 16;
-
-
- const uint16 rttMSW = timestamp - receiver_report.getLastSRNTPTimestampInt();
- const uint16 rttLSW = timestampFrac - receiver_report.getLastSRNTPTimestampFrac();
-
- uint32 rtt = rttMSW;
- rtt = rtt << 16 | rttLSW;
- rtt -= receiver_report.getDelayLastSR();
- stats["PACKET_COUNT"] = report.getPacketCount();
- stats["PACKET_LOSS"] = receiver_report.getFractionLost();
- stats["CUMUL_PACKET_LOSS"] = receiver_report.getCumulativePacketLost();
- stats["RTT"] = rtt;
- stats["LATENCY"] = 0; //TODO
- stats["HIGH_SEC_NUM"] = receiver_report.getExtendedSeqNum();
- stats["JITTER"] = receiver_report.getJitter();
- stats["DLSR"] = receiver_report.getDelayLastSR();
-
-#ifdef RTP_DEBUG
- RING_DBG("lastSR NTPTimestamp : %lu", receiver_report.getLastSRNTPTimestampFrac() << 16);
- RING_DBG("NTPTimestampFrac : %lu", timestampFrac);
- RING_DBG("rttMSW : %u", rttMSW);
- RING_DBG("rttLSW : %u", rttLSW);
- RING_DBG("RTT recomposed: %lu", rtt);
- RING_DBG("LDSR: %lu", receiver_report.getDelayLastSR());
- RING_DBG("Packet count : %u", stats["PACKET_COUNT"]);
- RING_DBG("Fraction packet loss : %.2f", (double) stats["PACKET_LOSS"] * 100 / 256);
- RING_DBG("Cumulative packet loss : %d", stats["CUMUL_PACKET_LOSS"]);
- RING_DBG("HighestSeqNum : %u", stats["HIGH_SEC_NUM"]);
- RING_DBG("Jitter : %u", stats["JITTER"]);
- RING_DBG("RTT : %.2f", (double) stats["RTT"] / 65536);
- RING_DBG("Delay since last report %.2f seconds", (double) stats["DLSR"] / 65536.0);
-#endif
- Manager::instance().getClient()->getCallManager()->onRtcpReportReceived(call_.getCallId(), stats);
- }
-}
-
-#if HAVE_IPV6
-
-AudioSymmetricRtpSessionIPv6::AudioSymmetricRtpSessionIPv6(SIPCall &call) :
- ost::SymmetricRTPSessionIPV6(static_cast<ost::IPV6Host>(call.getLocalIp()), call.getLocalAudioPort())
- , AudioRtpSession(call, *this)
-{
- RING_DBG("Setting new RTP/IPv6 session with destination %s:%d",
- call_.getLocalIp().toString().c_str(), call_.getLocalAudioPort());
-}
-
-AudioSymmetricRtpSessionIPv6::~AudioSymmetricRtpSessionIPv6()
-{
- // see explanation in AudioSymmetricRtpSessionIPv6()
- if (ost::Thread::isRunning()) {
- disableStack();
- ost::Thread::join();
- }
-}
-
-std::vector<long>
-AudioSymmetricRtpSessionIPv6::getSocketDescriptors() const
-{
- std::vector<long> result;
- result.push_back(dso->getRecvSocket());
- result.push_back(cso->getRecvSocket());
- return result;
-}
-
-void AudioSymmetricRtpSessionIPv6::startRTPLoop()
-{
- ost::SymmetricRTPSessionIPV6::startRunning();
-}
-
-// redefined from QueueRTCPManager
-void AudioSymmetricRtpSessionIPv6::onGotRR(ost::SyncSource& source, ost::RTCPCompoundHandler::RecvReport& RR, uint8 blocks)
-{
- ost::SymmetricRTPSessionIPV6::onGotRR(source, RR, blocks);
- // TODO: do something with this data
-#if 0
- std::cout << "I got an RR RTCP report from "
- << std::hex << (int)source.getID() << "@"
- << std::dec
- << source.getNetworkAddress() << ":"
- << source.getControlTransportPort() << std::endl;
-#endif
-}
-
-// redefined from QueueRTCPManager
-void AudioSymmetricRtpSessionIPv6::onGotSR(ost::SyncSource& source, ost::RTCPCompoundHandler::SendReport& SR, uint8 blocks)
-{
- ost::SymmetricRTPSessionIPV6::onGotSR(source, SR, blocks);
- // TODO: do something with this data
-}
-
-size_t
-AudioSymmetricRtpSessionIPv6::recvData(unsigned char* buffer, size_t len, ost::IPV4Host&, ost::tpport_t& port)
-{
- ost::IPV6Host hostv6 = call_.getLocalIp();
- RING_ERR("recvData %d ", hostv6.getAddressCount());
- size_t r = ost::SymmetricRTPSessionIPV6::recvData(buffer, len, hostv6, port);
- RING_ERR("recvData from %s %d called in ipv6 stack, size %d", IpAddr(hostv6.getAddress()).toString().c_str(), port, len);
- return r;
-}
-
-size_t
-AudioSymmetricRtpSessionIPv6::recvControl(unsigned char* buffer, size_t len, ost::IPV4Host&, ost::tpport_t& port)
-{
- ost::IPV6Host hostv6 = call_.getLocalIp();
- size_t r = ost::SymmetricRTPSessionIPV6::recvControl(buffer, len, hostv6, port);
- RING_ERR("recvControl from %s %d called in ipv6 stack, size %d", IpAddr(hostv6.getAddress()).toString().c_str(), port, len);
- return r;
-}
-
-
-#endif // HAVE_IPV6
-
-}
diff --git a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.h b/daemon/src/audio/audiortp/audio_symmetric_rtp_session.h
deleted file mode 100644
index b00ee519fe1229aaebe9fee687d0395d6d90db5b..0000000000000000000000000000000000000000
--- a/daemon/src/audio/audiortp/audio_symmetric_rtp_session.h
+++ /dev/null
@@ -1,112 +0,0 @@
-/*
- * Copyright (C) 2004-2015 Savoir-Faire Linux Inc.
- * Author: Pierre-Luc Bacon <pierre-luc.bacon@savoirfairelinux.com>
- * Author: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
- * Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
- * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- * This program 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- *
- * Additional permission under GNU GPL version 3 section 7:
- *
- * If you modify this program, or any covered work, by linking or
- * combining it with the OpenSSL project's OpenSSL library (or a
- * modified version of that library), containing parts covered by the
- * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
- * grants you additional permission to convey the resulting work.
- * Corresponding Source for a non-source form of such a combination
- * shall include the source code for the parts of OpenSSL used as well
- * as that of the covered work.
- */
-#ifndef AUDIO_SYMMETRIC_RTP_SESSION_H_
-#define AUDIO_SYMMETRIC_RTP_SESSION_H_
-
-#include "audio_rtp_session.h"
-#include "ip_utils.h"
-#include "noncopyable.h"
-
-#include <ccrtp/rtp.h>
-#include <ccrtp/iqueue.h>
-
-#include <exception>
-#include <cassert>
-#include <cstddef>
-
-class SIPCall;
-
-namespace ring {
-
-class AudioSymmetricRtpSession : public ost::SymmetricRTPSession, public AudioRtpSession {
- public:
- /**
- * Constructor
- * @param call The SIP call
- */
- AudioSymmetricRtpSession(SIPCall &call);
- ~AudioSymmetricRtpSession();
-
- std::vector<long>
- getSocketDescriptors() const;
-
- virtual bool onRTPPacketRecv(ost::IncomingRTPPkt& pkt) {
- return AudioRtpSession::onRTPPacketRecv(pkt);
- }
-
- private:
- void onGotRR(ost::SyncSource& source, ost::RTCPCompoundHandler::RecvReport& RR, uint8 blocks);
- void onGotSR(ost::SyncSource& source, ost::RTCPCompoundHandler::SendReport& SR, uint8 blocks);
-
- NON_COPYABLE(AudioSymmetricRtpSession);
-
- void startRTPLoop();
-};
-
-#if HAVE_IPV6
-
-class AudioSymmetricRtpSessionIPv6 : public ost::SymmetricRTPSessionIPV6, public AudioRtpSession {
- public:
- /**
- * Constructor
- * @param call The SIP call
- */
- AudioSymmetricRtpSessionIPv6(SIPCall &call);
- ~AudioSymmetricRtpSessionIPv6();
-
- std::vector<long>
- getSocketDescriptors() const;
-
- virtual bool onRTPPacketRecv(ost::IncomingRTPPkt& pkt) {
- return AudioRtpSession::onRTPPacketRecv(pkt);
- }
-
- protected:
- virtual size_t recvData(unsigned char* buffer, size_t len, ost::IPV4Host&, ost::tpport_t& port);
- virtual size_t recvControl(unsigned char* buffer, size_t len, ost::IPV4Host&, ost::tpport_t& port);
-
- private:
- void onGotRR(ost::SyncSource& source, ost::RTCPCompoundHandler::RecvReport& RR, uint8 blocks);
- void onGotSR(ost::SyncSource& source, ost::RTCPCompoundHandler::SendReport& SR, uint8 blocks);
-
- NON_COPYABLE(AudioSymmetricRtpSessionIPv6);
-
- void startRTPLoop();
-};
-
-#endif // HAVE_IPV6
-
-}
-#endif // AUDIO_SYMMETRIC_RTP_SESSION_H__
-
diff --git a/daemon/src/audio/audiortp/audio_zrtp_session.cpp b/daemon/src/audio/audiortp/audio_zrtp_session.cpp
deleted file mode 100644
index 6b36ede425954b21e7785f706c77c44da0332150..0000000000000000000000000000000000000000
--- a/daemon/src/audio/audiortp/audio_zrtp_session.cpp
+++ /dev/null
@@ -1,120 +0,0 @@
-/*
- * Copyright (C) 2004-2015 Savoir-Faire Linux Inc.
- * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
- * Author: Pierre-Luc Bacon <pierre-luc.bacon@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- * This program 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- *
- * Additional permission under GNU GPL version 3 section 7:
- *
- * If you modify this program, or any covered work, by linking or
- * combining it with the OpenSSL project's OpenSSL library (or a
- * modified version of that library), containing parts covered by the
- * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
- * grants you additional permission to convey the resulting work.
- * Corresponding Source for a non-source form of such a combination
- * shall include the source code for the parts of OpenSSL used as well
- * as that of the covered work.
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include "audio_zrtp_session.h"
-#include "zrtp_session_callback.h"
-#include "sip/sipcall.h"
-#include "logger.h"
-#include "manager.h"
-#include "fileutils.h"
-
-namespace ring {
-
-AudioZrtpSession::AudioZrtpSession(SIPCall &call, const std::string &zidFilename,
- const std::string &localIP) :
- ost::SymmetricZRTPSession(ost::InetHostAddress(localIP.c_str()), call.getLocalAudioPort())
- , AudioRtpSession(call, *this)
- , zidFilename_(zidFilename)
-#if HAVE_ZRTP_CONFIGURE
- , zrtpConfigure_()
-{
- setEnrollmentMode(false);
- setSignSas(false);
-#else
-{
-#endif
- initializeZid();
- RING_DBG("Setting new RTP session with destination %s:%d",
- localIP.c_str(), call_.getLocalAudioPort());
-}
-
-std::vector<long>
-AudioZrtpSession::getSocketDescriptors() const
-{
- std::vector<long> result;
- result.push_back(dso->getRecvSocket());
- result.push_back(cso->getRecvSocket());
- return result;
-}
-
-void AudioZrtpSession::initializeZid()
-{
- if (zidFilename_.empty())
- throw ZrtpZidException("zid filename empty");
-
- const std::string zidDirName(fileutils::get_cache_dir());
- const std::string zidCompleteFilename(zidDirName + DIR_SEPARATOR_STR + zidFilename_);
-
- fileutils::check_dir(zidDirName.c_str());
-
-#if HAVE_ZRTP_CONFIGURE
- // workaround buggy libzrtpcpp that can't parse EC25, see:
- // https://projects.savoirfairelinux.com/issues/40216
- zrtpConfigure_.reset(new ZrtpConfigure);
- // FIXME: perhaps this should be only done for libzrtpcpp < 3?
- zrtpConfigure_->setMandatoryOnly();
-
- if (initialize(zidCompleteFilename.c_str(), true, zrtpConfigure_.get()) >= 0) {
- setEnableZrtp(true);
- setUserCallback(new ZrtpSessionCallback(call_));
- return;
- }
-#else
- if (initialize(zidCompleteFilename.c_str()) >= 0) {
- setEnableZrtp(true);
- setUserCallback(new ZrtpSessionCallback(call_));
- return;
- }
-#endif
-
- RING_DBG("Initialization from ZID file failed. Trying to remove...");
-
- if (remove(zidCompleteFilename.c_str()) != 0)
- throw ZrtpZidException("zid file deletion failed");
-
- if (initialize(zidCompleteFilename.c_str()) < 0)
- throw ZrtpZidException("zid initialization failed");
-}
-
-int AudioZrtpSession::getIncrementForDTMF() const
-{
- return 160;
-}
-
-void AudioZrtpSession::startRTPLoop()
-{
- ost::SymmetricZRTPSession::startRunning();
-}
-
-}
diff --git a/daemon/src/audio/audiortp/audio_zrtp_session.h b/daemon/src/audio/audiortp/audio_zrtp_session.h
deleted file mode 100644
index 1ea8eaa30159f2c48c0b3eeb70240a01964fba65..0000000000000000000000000000000000000000
--- a/daemon/src/audio/audiortp/audio_zrtp_session.h
+++ /dev/null
@@ -1,84 +0,0 @@
-/*
- * Copyright (C) 2004-2015 Savoir-Faire Linux Inc.
- * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
- * Author: Pierre-Luc Bacon <pierre-luc.bacon@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- * This program 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- *
- * Additional permission under GNU GPL version 3 section 7:
- *
- * If you modify this program, or any covered work, by linking or
- * combining it with the OpenSSL project's OpenSSL library (or a
- * modified version of that library), containing parts covered by the
- * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
- * grants you additional permission to convey the resulting work.
- * Corresponding Source for a non-source form of such a combination
- * shall include the source code for the parts of OpenSSL used as well
- * as that of the covered work.
- */
-#ifndef __AUDIO_ZRTP_SESSION_H__
-#define __AUDIO_ZRTP_SESSION_H__
-
-#include <cstddef>
-#include <stdexcept>
-
-using std::ptrdiff_t;
-#include <ccrtp/rtp.h>
-#include <libzrtpcpp/zrtpccrtp.h>
-
-#if HAVE_ZRTP_CONFIGURE
-#include <memory>
-#endif
-
-#include "audio_rtp_session.h"
-
-class SIPCall;
-class AudioCodec;
-
-namespace ring {
-
-class ZrtpZidException : public std::runtime_error {
- public:
- ZrtpZidException(const char *str):
- std::runtime_error(str) {}
-};
-
-class AudioZrtpSession :
- public ::ost::SymmetricZRTPSession,
- public AudioRtpSession {
- public:
- AudioZrtpSession(SIPCall &call, const std::string& zidFilename, const std::string &localIP);
-
- std::vector<long>
- getSocketDescriptors() const;
-
- virtual bool onRTPPacketRecv(::ost::IncomingRTPPkt &pkt) {
- return AudioRtpSession::onRTPPacketRecv(pkt);
- }
-
- private:
- NON_COPYABLE(AudioZrtpSession);
-
- void initializeZid();
- std::string zidFilename_;
-#if HAVE_ZRTP_CONFIGURE
- std::unique_ptr<ZrtpConfigure> zrtpConfigure_;
-#endif
- void startRTPLoop();
- virtual int getIncrementForDTMF() const;
-};
-
-}
-
-#endif // __AUDIO_ZRTP_SESSION_H__
diff --git a/daemon/src/audio/audiortp/dtmf_event.cpp b/daemon/src/audio/audiortp/dtmf_event.cpp
deleted file mode 100644
index 77635b96b942f9a26f7266369c33dbd79e31e98f..0000000000000000000000000000000000000000
--- a/daemon/src/audio/audiortp/dtmf_event.cpp
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- * Copyright (C) 2004-2015 Savoir-Faire Linux Inc.
- * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- * This program 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- *
- * Additional permission under GNU GPL version 3 section 7:
- *
- * If you modify this program, or any covered work, by linking or
- * combining it with the OpenSSL project's OpenSSL library (or a
- * modified version of that library), containing parts covered by the
- * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
- * grants you additional permission to convey the resulting work.
- * Corresponding Source for a non-source form of such a combination
- * shall include the source code for the parts of OpenSSL used as well
- * as that of the covered work.
- */
-
-#include "dtmf_event.h"
-#include "logger.h"
-
-namespace ring {
-
-DTMFEvent::DTMFEvent(char digit) : payload(), newevent(true), length(1000)
-{
- /*
- From RFC2833:
-
- Event encoding (decimal)
- _________________________
- 0--9 0--9
- * 10
- # 11
- A--D 12--15
- Flash 16
- */
-
- switch (digit) {
- case '!':
- digit = 16;
- break;
-
- case '*':
- digit = 10;
- break;
-
- case '#':
- digit = 11;
- break;
-
- case 'A' ... 'D':
- digit = digit - 'A' + 12;
- break;
-
- case '0' ... '9':
- digit = digit - '0';
- break;
-
- default:
- RING_ERR("Unexpected DTMF %c", digit);
- }
-
- payload.event = digit;
- payload.ebit = false; // end of event bit
- payload.rbit = false; // reserved bit
- payload.duration = 1; // duration for this event
- payload.vol = 10;
-}
-
-}
diff --git a/daemon/src/audio/audiortp/dtmf_event.h b/daemon/src/audio/audiortp/dtmf_event.h
deleted file mode 100644
index 560ae6ab7c61c23b19bff62d762d8b04703084a0..0000000000000000000000000000000000000000
--- a/daemon/src/audio/audiortp/dtmf_event.h
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- * Copyright (C) 2004-2015 Savoir-Faire Linux Inc.
- * Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- * This program 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- *
- * Additional permission under GNU GPL version 3 section 7:
- *
- * If you modify this program, or any covered work, by linking or
- * combining it with the OpenSSL project's OpenSSL library (or a
- * modified version of that library), containing parts covered by the
- * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
- * grants you additional permission to convey the resulting work.
- * Corresponding Source for a non-source form of such a combination
- * shall include the source code for the parts of OpenSSL used as well
- * as that of the covered work.
- */
-
-#ifndef DTMF_EVENT_H_
-#define DTMF_EVENT_H_
-
-#include <ccrtp/rtp.h>
-
-namespace ring {
-
-struct DTMFEvent {
- DTMFEvent(char digit);
- ost::RTPPacket::RFC2833Payload payload;
- bool newevent;
- int length;
-};
-
-}
-
-#endif // DTMF_EVENT_H_
diff --git a/daemon/src/audio/audiortp/zrtp_session_callback.cpp b/daemon/src/audio/audiortp/zrtp_session_callback.cpp
deleted file mode 100644
index 8e7a22ab37714fe48400500f61d63df1fcdd7560..0000000000000000000000000000000000000000
--- a/daemon/src/audio/audiortp/zrtp_session_callback.cpp
+++ /dev/null
@@ -1,180 +0,0 @@
-/*
- * Copyright (C) 2004-2015 Savoir-Faire Linux Inc.
- * Author: Pierre-Luc Bacon <pierre-luc.bacon@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- *
- * Additional permission under GNU GPL version 3 section 7:
- *
- * If you modify this program, or any covered work, by linking or
- * combining it with the OpenSSL project's OpenSSL library (or a
- * modified version of that library), containing parts covered by the
- * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
- * grants you additional permission to convey the resulting work.
- * Corresponding Source for a non-source form of such a combination
- * shall include the source code for the parts of OpenSSL used as well
- * as that of the covered work.
- */
-#include "zrtp_session_callback.h"
-#include "logger.h"
-#include "sip/sipcall.h"
-#include "client/client.h"
-#include "client/callmanager.h"
-#include "manager.h"
-
-#include <cstdlib>
-#include <map>
-
-using namespace GnuZrtpCodes;
-
-using namespace ost;
-
-namespace ring {
-
-ZrtpSessionCallback::ZrtpSessionCallback(SIPCall &call) : call_(call)
-{
- // we've already initialized the maps, we only need to check one
- if (not infoMap_.empty())
- return;
-
- // Information Map
- infoMap_[InfoHelloReceived] = "Hello received, preparing a Commit";
- infoMap_[InfoCommitDHGenerated] = "Commit: Generated a public DH key";
- infoMap_[InfoRespCommitReceived] = "Responder: Commit received, preparing DHPart1";
- infoMap_[InfoDH1DHGenerated] = "DH1Part: Generated a public DH key";
- infoMap_[InfoInitDH1Received] = "Initiator: DHPart1 received, preparing DHPart2";
- infoMap_[InfoRespDH2Received] = "Responder: DHPart2 received, preparing Confirm1";
- infoMap_[InfoInitConf1Received] = "Initiator: Confirm1 received, preparing Confirm2";
- infoMap_[InfoRespConf2Received] = "Responder: Confirm2 received, preparing Conf2Ack";
- infoMap_[InfoRSMatchFound] = "At least one retained secrets matches - security OK";
- infoMap_[InfoSecureStateOn] = "Entered secure state";
- infoMap_[InfoSecureStateOff] = "No more security for this session";
-
- // Warning Map
- warningMap_[WarningDHAESmismatch] = "Commit contains an AES256 cipher but does not offer a Diffie-Helman 4096";
- warningMap_[WarningGoClearReceived] = "Received a GoClear message";
- warningMap_[WarningDHShort] = "Hello offers an AES256 cipher but does not offer a Diffie-Helman 4096";
- warningMap_[WarningNoRSMatch] = "No retained secret matches - verify SAS";
- warningMap_[WarningCRCmismatch] = "Internal ZRTP packet checksum mismatch - packet dropped";
- warningMap_[WarningSRTPauthError] = "Dropping packet because SRTP authentication failed!";
- warningMap_[WarningSRTPreplayError] = "Dropping packet because SRTP replay check failed!";
-
- severeMap_[SevereHelloHMACFailed] = "Hash HMAC check of Hello failed!";
- severeMap_[SevereCommitHMACFailed] = "Hash HMAC check of Commit failed!";
- severeMap_[SevereDH1HMACFailed] = "Hash HMAC check of DHPart1 failed!";
- severeMap_[SevereDH2HMACFailed] = "Hash HMAC check of DHPart2 failed!";
- severeMap_[SevereCannotSend] = "Cannot send data - connection or peer down?";
- severeMap_[SevereProtocolError] = "Internal protocol error occured!";
- severeMap_[SevereNoTimer] = "Cannot start a timer - internal resources exhausted?";
- severeMap_[SevereTooMuchRetries] = "Too much retries during ZRTP negotiation - connection or peer down?";
-
- // Zrtp protocol related messages map
- zrtpMap_[MalformedPacket] = "Malformed packet (CRC OK, but wrong structure)";
- zrtpMap_[CriticalSWError] = "Critical software error";
- zrtpMap_[UnsuppZRTPVersion] = "Unsupported ZRTP version";
- zrtpMap_[HelloCompMismatch] = "Hello components mismatch";
- zrtpMap_[UnsuppHashType] = "Hash type not supported";
- zrtpMap_[UnsuppCiphertype] = "Cipher type not supported";
- zrtpMap_[UnsuppPKExchange] = "Public key exchange not supported";
- zrtpMap_[UnsuppSRTPAuthTag] = "SRTP auth. tag not supported";
- zrtpMap_[UnsuppSASScheme] = "SAS scheme not supported";
- zrtpMap_[NoSharedSecret] = "No shared secret available, DH mode required";
- zrtpMap_[DHErrorWrongPV] = "DH Error: bad pvi or pvr ( == 1, 0, or p-1)";
- zrtpMap_[DHErrorWrongHVI] = "DH Error: hvi != hashed data";
- zrtpMap_[SASuntrustedMiTM] = "Received relayed SAS from untrusted MiTM";
- zrtpMap_[ConfirmHMACWrong] = "Auth. Error: Bad Confirm pkt HMAC";
- zrtpMap_[NonceReused] = "Nonce reuse";
- zrtpMap_[EqualZIDHello] = "Equal ZIDs in Hello";
- zrtpMap_[GoCleatNotAllowed] = "GoClear packet received, but not allowed";
-}
-
-void
-ZrtpSessionCallback::secureOn(std::string cipher)
-{
- RING_DBG("Secure mode is on with cipher %s", cipher.c_str());
- Manager::instance().getClient()->getCallManager()->secureZrtpOn(call_.getCallId(), cipher);
-}
-
-void
-ZrtpSessionCallback::secureOff()
-{
- RING_DBG("Secure mode is off");
- Manager::instance().getClient()->getCallManager()->secureZrtpOff(call_.getCallId());
-}
-
-void
-ZrtpSessionCallback::showSAS(std::string sas, bool verified)
-{
- RING_DBG("SAS is: %s", sas.c_str());
- Manager::instance().getClient()->getCallManager()->showSAS(call_.getCallId(), sas, verified);
-}
-
-void
-ZrtpSessionCallback::zrtpNotSuppOther()
-{
- RING_DBG("Callee does not support ZRTP");
- Manager::instance().getClient()->getCallManager()->zrtpNotSuppOther(call_.getCallId());
-}
-
-void
-ZrtpSessionCallback::showMessage(GnuZrtpCodes::MessageSeverity sev, int32_t subCode)
-{
- if (sev == ZrtpError) {
- if (subCode < 0) { // received an error packet from peer
- RING_DBG("Received an error packet from peer:");
- } else
- RING_DBG("Sent error packet to peer:");
- }
-}
-
-void
-ZrtpSessionCallback::zrtpNegotiationFailed(MessageSeverity severity, int subCode)
-{
- if (severity == ZrtpError) {
- if (subCode < 0) { // received an error packet from peer
- subCode *= -1;
- RING_DBG("Received error packet: ");
- } else
- RING_DBG("Sent error packet: ");
-
- std::map<int32, std::string>::const_iterator iter = zrtpMap_.find(subCode);
-
- if (iter != zrtpMap_.end()) {
- RING_DBG("%s", iter->second.c_str());
- Manager::instance().getClient()->getCallManager()->zrtpNegotiationFailed(call_.getCallId(), iter->second, "ZRTP");
- }
- } else {
- std::map<int32, std::string>::const_iterator iter = severeMap_.find(subCode);
-
- if (iter != severeMap_.end()) {
- RING_DBG("%s", iter->second.c_str());
- Manager::instance().getClient()->getCallManager()->zrtpNegotiationFailed(call_.getCallId(), iter->second, "severe");
- }
- }
-}
-
-void
-ZrtpSessionCallback::confirmGoClear()
-{
- RING_DBG("Received go clear message. Until confirmation, ZRTP won't send any data");
- Manager::instance().getClient()->getCallManager()->zrtpNotSuppOther(call_.getCallId());
-}
-
-std::map<int32, std::string> ZrtpSessionCallback::infoMap_;
-std::map<int32, std::string> ZrtpSessionCallback::warningMap_;
-std::map<int32, std::string> ZrtpSessionCallback::severeMap_;
-std::map<int32, std::string> ZrtpSessionCallback::zrtpMap_;
-}
-
diff --git a/daemon/src/audio/audiortp/zrtp_session_callback.h b/daemon/src/audio/audiortp/zrtp_session_callback.h
deleted file mode 100644
index d0fcedfb697916c84880ff7a4949ef6fa932e790..0000000000000000000000000000000000000000
--- a/daemon/src/audio/audiortp/zrtp_session_callback.h
+++ /dev/null
@@ -1,67 +0,0 @@
-/*
- * Copyright (C) 2004-2015 Savoir-Faire Linux Inc.
- * Author: Pierre-Luc Bacon <pierre-luc.bacon@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- *
- * Additional permission under GNU GPL version 3 section 7:
- *
- * If you modify this program, or any covered work, by linking or
- * combining it with the OpenSSL project's OpenSSL library (or a
- * modified version of that library), containing parts covered by the
- * terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
- * grants you additional permission to convey the resulting work.
- * Corresponding Source for a non-source form of such a combination
- * shall include the source code for the parts of OpenSSL used as well
- * as that of the covered work.
- */
-
-#ifndef ZRTP_SESSION_CALLBACK_H_
-#define ZRTP_SESSION_CALLBACK_H_
-#include <cstddef>
-
-using std::ptrdiff_t;
-
-#include <libzrtpcpp/zrtpccrtp.h>
-#include <libzrtpcpp/ZrtpQueue.h>
-#include <libzrtpcpp/ZrtpUserCallback.h>
-#include <string>
-#include <map>
-
-class SIPCall;
-
-namespace ring {
-
-class ZrtpSessionCallback: public ZrtpUserCallback {
- public:
- ZrtpSessionCallback(SIPCall &call);
-
- void secureOn(std::string cipher);
- void secureOff();
- void showSAS(std::string sas, bool verified);
- void zrtpNotSuppOther();
- void showMessage(GnuZrtpCodes::MessageSeverity sev, int32_t subCode);
- void zrtpNegotiationFailed(GnuZrtpCodes::MessageSeverity severity, int subCode);
- void confirmGoClear();
-
- private:
- SIPCall &call_;
- static std::map<int32, std::string> infoMap_;
- static std::map<int32, std::string> warningMap_;
- static std::map<int32, std::string> severeMap_;
- static std::map<int32, std::string> zrtpMap_;
-};
-}
-#endif // ZRTP_SESSION_CALLBACK_H_
diff --git a/daemon/src/client/callmanager.cpp b/daemon/src/client/callmanager.cpp
index cf8329991e652e242e7868c6c482e5a5bb69103e..718415398af3995b57aed723f635dbc23a3a303c 100644
--- a/daemon/src/client/callmanager.cpp
+++ b/daemon/src/client/callmanager.cpp
@@ -37,10 +37,6 @@
#include "sip/sipcall.h"
#include "sip/sipvoiplink.h"
#include "audio/audiolayer.h"
-#include "audio/audiortp/audio_rtp_factory.h"
-#if HAVE_ZRTP
-#include "audio/audiortp/audio_zrtp_session.h"
-#endif
#include "logger.h"
#include "manager.h"
@@ -303,76 +299,31 @@ CallManager::getAudioZrtpSession(const std::string& callID)
void
CallManager::setSASVerified(const std::string& callID)
{
-#if USE_CCRTP && HAVE_ZRTP
- try {
- ring::AudioZrtpSession * zSession;
- zSession = getAudioZrtpSession(callID);
- zSession->SASVerified();
- } catch (...) {
- }
-#else
- RING_ERR("No zrtp support for %s, please recompile SFLphone with zrtp", callID.c_str());
-#endif
+ RING_ERR("ZRTP not supported");
}
void
CallManager::resetSASVerified(const std::string& callID)
{
-#if USE_CCRTP && HAVE_ZRTP
- try {
- ring::AudioZrtpSession * zSession;
- zSession = getAudioZrtpSession(callID);
- zSession->resetSASVerified();
- } catch (...) {
- }
-#else
- RING_ERR("No zrtp support for %s, please recompile SFLphone with zrtp", callID.c_str());
-#endif
+ RING_ERR("ZRTP not supported");
}
void
CallManager::setConfirmGoClear(const std::string& callID)
{
-#if USE_CCRTP && HAVE_ZRTP
- try {
- ring::AudioZrtpSession * zSession;
- zSession = getAudioZrtpSession(callID);
- zSession->goClearOk();
- } catch (...) {
- }
-#else
- RING_ERR("No zrtp support for %s, please recompile SFLphone with zrtp", callID.c_str());
-#endif
+ RING_ERR("ZRTP not supported");
}
void
CallManager::requestGoClear(const std::string& callID)
{
-#if USE_CCRTP && HAVE_ZRTP
- try {
- ring::AudioZrtpSession * zSession;
- zSession = getAudioZrtpSession(callID);
- zSession->requestGoClear();
- } catch (...) {
- }
-#else
- RING_ERR("No zrtp support for %s, please recompile SFLphone with zrtp", callID.c_str());
-#endif
+ RING_ERR("ZRTP not supported");
}
void
CallManager::acceptEnrollment(const std::string& callID, bool accepted)
{
-#if USE_CCRTP && HAVE_ZRTP
- try {
- ring::AudioZrtpSession * zSession;
- zSession = getAudioZrtpSession(callID);
- zSession->acceptEnrollment(accepted);
- } catch (...) {
- }
-#else
- RING_ERR("No zrtp support for %s, please recompile SFLphone with zrtp", callID.c_str());
-#endif
+ RING_ERR("ZRTP not supported");
}
void CallManager::sendTextMessage(const std::string& callID, const std::string& message, const std::string& from)
diff --git a/daemon/src/client/videomanager.cpp b/daemon/src/client/videomanager.cpp
index 4922473a0d29b8e91415fdf2eacd650dd3427079..7b9827103cdb5a1e6975e1933dac988784eb7660 100644
--- a/daemon/src/client/videomanager.cpp
+++ b/daemon/src/client/videomanager.cpp
@@ -31,7 +31,7 @@
*/
#include "videomanager.h"
-#include "video/libav_utils.h"
+#include "libav_utils.h"
#include "video/video_input.h"
#include "video/video_device_monitor.h"
#include "account.h"
diff --git a/daemon/src/media/Makefile.am b/daemon/src/media/Makefile.am
new file mode 100644
index 0000000000000000000000000000000000000000..3cdc6f3b690f49c8712ee871833465bee4d200b1
--- /dev/null
+++ b/daemon/src/media/Makefile.am
@@ -0,0 +1,34 @@
+include $(top_srcdir)/globals.mak
+
+noinst_LTLIBRARIES = libmedia.la
+
+SUBDIRS = audio
+
+if RING_VIDEO
+SUBDIRS += video
+endif
+
+libmedia_la_SOURCES = \
+ libav_utils.cpp
+
+noinst_HEADERS = \
+ libav_utils.h \
+ libav_deps.h
+
+libmedia_la_LIBADD = \
+ ./audio/libaudio.la
+
+if RING_VIDEO
+libmedia_la_libADD = \
+ ./video/libvideo.la
+endif
+
+libmedia_la_LDFLAGS = @LIBAVCODEC_LIBS@ @LIBAVFORMAT_LIBS@ @LIBAVDEVICE_LIBS@ @LIBSWSCALE_LIBS@ @LIBAVUTIL_LIBS@
+
+AM_CFLAGS=@LIBAVCODEC_CFLAGS@ @LIBAVFORMAT_CFLAGS@ @LIBAVDEVICE_CFLAGS@ @LIBSWSCALE_CFLAGS@
+
+AM_CXXFLAGS=@LIBAVCODEC_CFLAGS@ @LIBAVFORMAT_CFLAGS@ @LIBAVDEVICE_CFLAGS@ @LIBSWSCALE_CFLAGS@
+
+if RING_VIDEO
+AM_CXXFLAGS += @UDEV_CFLAGS@
+endif
diff --git a/daemon/src/audio/Makefile.am b/daemon/src/media/audio/Makefile.am
similarity index 98%
rename from daemon/src/audio/Makefile.am
rename to daemon/src/media/audio/Makefile.am
index 363a4c9b012d2a5322c8124aca79787e8685aee1..929b912c572b2778792788be91c65dd02c8b007b 100644
--- a/daemon/src/audio/Makefile.am
+++ b/daemon/src/media/audio/Makefile.am
@@ -56,7 +56,7 @@ noinst_HEADERS = \
audiorecord.h \
audiorecorder.h \
audiolayer.h \
- recordable.h \
+ recordable.h \
$(RING_SPEEXDSP_HEAD) \
dcblocker.h \
resampler.h
diff --git a/daemon/src/audio/alsa/Makefile.am b/daemon/src/media/audio/alsa/Makefile.am
similarity index 100%
rename from daemon/src/audio/alsa/Makefile.am
rename to daemon/src/media/audio/alsa/Makefile.am
diff --git a/daemon/src/audio/alsa/alsalayer.cpp b/daemon/src/media/audio/alsa/alsalayer.cpp
similarity index 100%
rename from daemon/src/audio/alsa/alsalayer.cpp
rename to daemon/src/media/audio/alsa/alsalayer.cpp
diff --git a/daemon/src/audio/alsa/alsalayer.h b/daemon/src/media/audio/alsa/alsalayer.h
similarity index 100%
rename from daemon/src/audio/alsa/alsalayer.h
rename to daemon/src/media/audio/alsa/alsalayer.h
diff --git a/daemon/src/audio/audiobuffer.cpp b/daemon/src/media/audio/audiobuffer.cpp
similarity index 100%
rename from daemon/src/audio/audiobuffer.cpp
rename to daemon/src/media/audio/audiobuffer.cpp
diff --git a/daemon/src/audio/audiobuffer.h b/daemon/src/media/audio/audiobuffer.h
similarity index 100%
rename from daemon/src/audio/audiobuffer.h
rename to daemon/src/media/audio/audiobuffer.h
diff --git a/daemon/src/audio/audiolayer.cpp b/daemon/src/media/audio/audiolayer.cpp
similarity index 100%
rename from daemon/src/audio/audiolayer.cpp
rename to daemon/src/media/audio/audiolayer.cpp
diff --git a/daemon/src/audio/audiolayer.h b/daemon/src/media/audio/audiolayer.h
similarity index 100%
rename from daemon/src/audio/audiolayer.h
rename to daemon/src/media/audio/audiolayer.h
diff --git a/daemon/src/audio/audioloop.cpp b/daemon/src/media/audio/audioloop.cpp
similarity index 100%
rename from daemon/src/audio/audioloop.cpp
rename to daemon/src/media/audio/audioloop.cpp
diff --git a/daemon/src/audio/audioloop.h b/daemon/src/media/audio/audioloop.h
similarity index 99%
rename from daemon/src/audio/audioloop.h
rename to daemon/src/media/audio/audioloop.h
index a6a4f6d6a04784ab22f3c06c902d82ab6cbace74..e257a86658a6df6080a9bfb813d2a8ebcd61fa26 100644
--- a/daemon/src/audio/audioloop.h
+++ b/daemon/src/media/audio/audioloop.h
@@ -92,4 +92,3 @@ class AudioLoop {
}
#endif // __AUDIOLOOP_H__
-
diff --git a/daemon/src/audio/audiorecord.cpp b/daemon/src/media/audio/audiorecord.cpp
similarity index 100%
rename from daemon/src/audio/audiorecord.cpp
rename to daemon/src/media/audio/audiorecord.cpp
diff --git a/daemon/src/audio/audiorecord.h b/daemon/src/media/audio/audiorecord.h
similarity index 98%
rename from daemon/src/audio/audiorecord.h
rename to daemon/src/media/audio/audiorecord.h
index 41f0019c79636f8ed45976a541832b1e8c400f84..b357313a736e74001018b15442fee9ba59bfd7fd 100644
--- a/daemon/src/audio/audiorecord.h
+++ b/daemon/src/media/audio/audiorecord.h
@@ -58,8 +58,8 @@ class AudioRecord {
void initFilename(const std::string &peerNumber);
/**
- * Return the filepath of the recording
- */
+ * Return the filepath of the recording
+ */
std::string getFilename() const;
/**
diff --git a/daemon/src/audio/audiorecorder.cpp b/daemon/src/media/audio/audiorecorder.cpp
similarity index 100%
rename from daemon/src/audio/audiorecorder.cpp
rename to daemon/src/media/audio/audiorecorder.cpp
diff --git a/daemon/src/audio/audiorecorder.h b/daemon/src/media/audio/audiorecorder.h
similarity index 100%
rename from daemon/src/audio/audiorecorder.h
rename to daemon/src/media/audio/audiorecorder.h
diff --git a/daemon/src/media/audio/audiortp/Makefile.am b/daemon/src/media/audio/audiortp/Makefile.am
new file mode 100644
index 0000000000000000000000000000000000000000..e047ab1d89c4b5c80e5645d471d47fadb66f9d0d
--- /dev/null
+++ b/daemon/src/media/audio/audiortp/Makefile.am
@@ -0,0 +1,15 @@
+include $(top_srcdir)/globals.mak
+
+noinst_LTLIBRARIES = libaudiortp.la
+
+libaudiortp_la_SOURCES = \
+ base64.c base64.h
+
+libaudiortp_la_SOURCES += \
+ avformat_rtp_session.cpp \
+ avformat_rtp_session.h
+
+AM_CXXFLAGS = @LIBAVFORMAT_CFLAGS@
+
+# FIXME
+AM_CPPFLAGS += -I$(top_srcdir)/src
diff --git a/daemon/src/audio/audiortp/TODO b/daemon/src/media/audio/audiortp/TODO
similarity index 92%
rename from daemon/src/audio/audiortp/TODO
rename to daemon/src/media/audio/audiortp/TODO
index 06bf60ae98d7519436f509de8c85c2292fe632da..246c72bb6715e494b65a23fad9221afcde801e31 100644
--- a/daemon/src/audio/audiortp/TODO
+++ b/daemon/src/media/audio/audiortp/TODO
@@ -13,7 +13,6 @@ Needs to be implemented
* DTMF over RTP
* RTP + STUN
* Rename Video{Encoder,Decoder} to AV{Encoder,Decoder}
-* Drop CCRTP and its dependencies and its dependents for real
Needs to be fixed:
------------------
diff --git a/daemon/src/audio/audiortp/avformat_rtp_session.cpp b/daemon/src/media/audio/audiortp/avformat_rtp_session.cpp
similarity index 99%
rename from daemon/src/audio/audiortp/avformat_rtp_session.cpp
rename to daemon/src/media/audio/audiortp/avformat_rtp_session.cpp
index 412eae00d8d92553f89e4b9a60270f05f20277cd..09b30db2c0cc3f0e04225def00bd41e1d7f3835d 100644
--- a/daemon/src/audio/audiortp/avformat_rtp_session.cpp
+++ b/daemon/src/media/audio/audiortp/avformat_rtp_session.cpp
@@ -36,7 +36,7 @@
#include "video/video_base.h"
#include "video/video_encoder.h"
#include "video/video_decoder.h"
-#include "video/libav_deps.h"
+#include "libav_deps.h"
#include "audio/audiobuffer.h"
#include "audio/ringbufferpool.h"
#include "audio/resampler.h"
diff --git a/daemon/src/audio/audiortp/avformat_rtp_session.h b/daemon/src/media/audio/audiortp/avformat_rtp_session.h
similarity index 100%
rename from daemon/src/audio/audiortp/avformat_rtp_session.h
rename to daemon/src/media/audio/audiortp/avformat_rtp_session.h
diff --git a/daemon/src/audio/audiortp/base64.c b/daemon/src/media/audio/audiortp/base64.c
similarity index 100%
rename from daemon/src/audio/audiortp/base64.c
rename to daemon/src/media/audio/audiortp/base64.c
diff --git a/daemon/src/audio/audiortp/base64.h b/daemon/src/media/audio/audiortp/base64.h
similarity index 100%
rename from daemon/src/audio/audiortp/base64.h
rename to daemon/src/media/audio/audiortp/base64.h
diff --git a/daemon/src/audio/codecs/Makefile.am b/daemon/src/media/audio/codecs/Makefile.am
similarity index 99%
rename from daemon/src/audio/codecs/Makefile.am
rename to daemon/src/media/audio/codecs/Makefile.am
index 540a07866ef5014a2885da5390198ff225630d58..275a2d523a3f36a84a84c0ec3c8412280e6e4302 100644
--- a/daemon/src/audio/codecs/Makefile.am
+++ b/daemon/src/media/audio/codecs/Makefile.am
@@ -4,7 +4,6 @@ noinst_LTLIBRARIES = libcodecdescriptor.la
libcodecdescriptor_la_SOURCES = audiocodecfactory.cpp audiocodec.cpp audiocodec.h audiocodecfactory.h
libcodecdescriptor_la_CXXFLAGS = -fPIC
-libcodecdescriptor_la_LIBADD = $(CCRTP_LIBS)
if BUILD_GSM
GSM_LIB = libcodec_gsm.so
@@ -119,4 +118,3 @@ uninstall-libcodec_speex_ub_so:
rm -f $(sflcodecdir)/libcodec_speex_ub.so
uninstall-libcodec_ilbc_so:
rm -f $(sflcodecdir)/libcodec_ilbc.so
-
diff --git a/daemon/src/audio/codecs/README b/daemon/src/media/audio/codecs/README
similarity index 100%
rename from daemon/src/audio/codecs/README
rename to daemon/src/media/audio/codecs/README
diff --git a/daemon/src/audio/codecs/alaw.cpp b/daemon/src/media/audio/codecs/alaw.cpp
similarity index 100%
rename from daemon/src/audio/codecs/alaw.cpp
rename to daemon/src/media/audio/codecs/alaw.cpp
diff --git a/daemon/src/audio/codecs/audiocodec.cpp b/daemon/src/media/audio/codecs/audiocodec.cpp
similarity index 100%
rename from daemon/src/audio/codecs/audiocodec.cpp
rename to daemon/src/media/audio/codecs/audiocodec.cpp
diff --git a/daemon/src/audio/codecs/audiocodec.h b/daemon/src/media/audio/codecs/audiocodec.h
similarity index 100%
rename from daemon/src/audio/codecs/audiocodec.h
rename to daemon/src/media/audio/codecs/audiocodec.h
diff --git a/daemon/src/audio/codecs/audiocodecfactory.cpp b/daemon/src/media/audio/codecs/audiocodecfactory.cpp
similarity index 100%
rename from daemon/src/audio/codecs/audiocodecfactory.cpp
rename to daemon/src/media/audio/codecs/audiocodecfactory.cpp
diff --git a/daemon/src/audio/codecs/audiocodecfactory.h b/daemon/src/media/audio/codecs/audiocodecfactory.h
similarity index 100%
rename from daemon/src/audio/codecs/audiocodecfactory.h
rename to daemon/src/media/audio/codecs/audiocodecfactory.h
diff --git a/daemon/src/audio/codecs/g711.h b/daemon/src/media/audio/codecs/g711.h
similarity index 100%
rename from daemon/src/audio/codecs/g711.h
rename to daemon/src/media/audio/codecs/g711.h
diff --git a/daemon/src/audio/codecs/g722.cpp b/daemon/src/media/audio/codecs/g722.cpp
similarity index 100%
rename from daemon/src/audio/codecs/g722.cpp
rename to daemon/src/media/audio/codecs/g722.cpp
diff --git a/daemon/src/audio/codecs/g722.h b/daemon/src/media/audio/codecs/g722.h
similarity index 100%
rename from daemon/src/audio/codecs/g722.h
rename to daemon/src/media/audio/codecs/g722.h
diff --git a/daemon/src/audio/codecs/g729.cpp b/daemon/src/media/audio/codecs/g729.cpp
similarity index 100%
rename from daemon/src/audio/codecs/g729.cpp
rename to daemon/src/media/audio/codecs/g729.cpp
diff --git a/daemon/src/audio/codecs/g729.h b/daemon/src/media/audio/codecs/g729.h
similarity index 100%
rename from daemon/src/audio/codecs/g729.h
rename to daemon/src/media/audio/codecs/g729.h
diff --git a/daemon/src/audio/codecs/gsmcodec.cpp b/daemon/src/media/audio/codecs/gsmcodec.cpp
similarity index 100%
rename from daemon/src/audio/codecs/gsmcodec.cpp
rename to daemon/src/media/audio/codecs/gsmcodec.cpp
diff --git a/daemon/src/audio/codecs/ilbc.cpp b/daemon/src/media/audio/codecs/ilbc.cpp
similarity index 100%
rename from daemon/src/audio/codecs/ilbc.cpp
rename to daemon/src/media/audio/codecs/ilbc.cpp
diff --git a/daemon/src/audio/codecs/opuscodec.cpp b/daemon/src/media/audio/codecs/opuscodec.cpp
similarity index 100%
rename from daemon/src/audio/codecs/opuscodec.cpp
rename to daemon/src/media/audio/codecs/opuscodec.cpp
diff --git a/daemon/src/audio/codecs/opuscodec.h b/daemon/src/media/audio/codecs/opuscodec.h
similarity index 100%
rename from daemon/src/audio/codecs/opuscodec.h
rename to daemon/src/media/audio/codecs/opuscodec.h
diff --git a/daemon/src/audio/codecs/speexcodec.cpp b/daemon/src/media/audio/codecs/speexcodec.cpp
similarity index 100%
rename from daemon/src/audio/codecs/speexcodec.cpp
rename to daemon/src/media/audio/codecs/speexcodec.cpp
diff --git a/daemon/src/audio/codecs/speexcodec.h b/daemon/src/media/audio/codecs/speexcodec.h
similarity index 100%
rename from daemon/src/audio/codecs/speexcodec.h
rename to daemon/src/media/audio/codecs/speexcodec.h
diff --git a/daemon/src/audio/codecs/ulaw.cpp b/daemon/src/media/audio/codecs/ulaw.cpp
similarity index 100%
rename from daemon/src/audio/codecs/ulaw.cpp
rename to daemon/src/media/audio/codecs/ulaw.cpp
diff --git a/daemon/src/audio/coreaudio/Makefile.am b/daemon/src/media/audio/coreaudio/Makefile.am
similarity index 85%
rename from daemon/src/audio/coreaudio/Makefile.am
rename to daemon/src/media/audio/coreaudio/Makefile.am
index b600cadcc6071bf5b0f2e9345d51ad362740eeed..1969b70316e52011e4950a219db9ccb840aa921b 100644
--- a/daemon/src/audio/coreaudio/Makefile.am
+++ b/daemon/src/media/audio/coreaudio/Makefile.am
@@ -1,3 +1,5 @@
+include $(top_srcdir)/globals.mak
+
if HAVE_OSX
noinst_LTLIBRARIES = libcoreaudiolayer.la
endif
diff --git a/daemon/src/audio/coreaudio/audiodevice.cpp b/daemon/src/media/audio/coreaudio/audiodevice.cpp
similarity index 100%
rename from daemon/src/audio/coreaudio/audiodevice.cpp
rename to daemon/src/media/audio/coreaudio/audiodevice.cpp
diff --git a/daemon/src/audio/coreaudio/audiodevice.h b/daemon/src/media/audio/coreaudio/audiodevice.h
similarity index 100%
rename from daemon/src/audio/coreaudio/audiodevice.h
rename to daemon/src/media/audio/coreaudio/audiodevice.h
diff --git a/daemon/src/audio/coreaudio/corelayer.cpp b/daemon/src/media/audio/coreaudio/corelayer.cpp
similarity index 100%
rename from daemon/src/audio/coreaudio/corelayer.cpp
rename to daemon/src/media/audio/coreaudio/corelayer.cpp
diff --git a/daemon/src/audio/coreaudio/corelayer.h b/daemon/src/media/audio/coreaudio/corelayer.h
similarity index 100%
rename from daemon/src/audio/coreaudio/corelayer.h
rename to daemon/src/media/audio/coreaudio/corelayer.h
diff --git a/daemon/src/audio/dcblocker.cpp b/daemon/src/media/audio/dcblocker.cpp
similarity index 100%
rename from daemon/src/audio/dcblocker.cpp
rename to daemon/src/media/audio/dcblocker.cpp
diff --git a/daemon/src/audio/dcblocker.h b/daemon/src/media/audio/dcblocker.h
similarity index 100%
rename from daemon/src/audio/dcblocker.h
rename to daemon/src/media/audio/dcblocker.h
diff --git a/daemon/src/audio/dsp.cpp b/daemon/src/media/audio/dsp.cpp
similarity index 100%
rename from daemon/src/audio/dsp.cpp
rename to daemon/src/media/audio/dsp.cpp
diff --git a/daemon/src/audio/dsp.h b/daemon/src/media/audio/dsp.h
similarity index 100%
rename from daemon/src/audio/dsp.h
rename to daemon/src/media/audio/dsp.h
diff --git a/daemon/src/audio/jack/Makefile.am b/daemon/src/media/audio/jack/Makefile.am
similarity index 100%
rename from daemon/src/audio/jack/Makefile.am
rename to daemon/src/media/audio/jack/Makefile.am
diff --git a/daemon/src/audio/jack/jacklayer.cpp b/daemon/src/media/audio/jack/jacklayer.cpp
similarity index 100%
rename from daemon/src/audio/jack/jacklayer.cpp
rename to daemon/src/media/audio/jack/jacklayer.cpp
diff --git a/daemon/src/audio/jack/jacklayer.h b/daemon/src/media/audio/jack/jacklayer.h
similarity index 100%
rename from daemon/src/audio/jack/jacklayer.h
rename to daemon/src/media/audio/jack/jacklayer.h
diff --git a/daemon/src/audio/opensl/Makefile.am b/daemon/src/media/audio/opensl/Makefile.am
similarity index 100%
rename from daemon/src/audio/opensl/Makefile.am
rename to daemon/src/media/audio/opensl/Makefile.am
diff --git a/daemon/src/audio/opensl/opensllayer.cpp b/daemon/src/media/audio/opensl/opensllayer.cpp
similarity index 100%
rename from daemon/src/audio/opensl/opensllayer.cpp
rename to daemon/src/media/audio/opensl/opensllayer.cpp
diff --git a/daemon/src/audio/opensl/opensllayer.h b/daemon/src/media/audio/opensl/opensllayer.h
similarity index 100%
rename from daemon/src/audio/opensl/opensllayer.h
rename to daemon/src/media/audio/opensl/opensllayer.h
diff --git a/daemon/src/audio/pulseaudio/Makefile.am b/daemon/src/media/audio/pulseaudio/Makefile.am
similarity index 100%
rename from daemon/src/audio/pulseaudio/Makefile.am
rename to daemon/src/media/audio/pulseaudio/Makefile.am
diff --git a/daemon/src/audio/pulseaudio/audiostream.cpp b/daemon/src/media/audio/pulseaudio/audiostream.cpp
similarity index 100%
rename from daemon/src/audio/pulseaudio/audiostream.cpp
rename to daemon/src/media/audio/pulseaudio/audiostream.cpp
diff --git a/daemon/src/audio/pulseaudio/audiostream.h b/daemon/src/media/audio/pulseaudio/audiostream.h
similarity index 100%
rename from daemon/src/audio/pulseaudio/audiostream.h
rename to daemon/src/media/audio/pulseaudio/audiostream.h
diff --git a/daemon/src/audio/pulseaudio/pulselayer.cpp b/daemon/src/media/audio/pulseaudio/pulselayer.cpp
similarity index 100%
rename from daemon/src/audio/pulseaudio/pulselayer.cpp
rename to daemon/src/media/audio/pulseaudio/pulselayer.cpp
diff --git a/daemon/src/audio/pulseaudio/pulselayer.h b/daemon/src/media/audio/pulseaudio/pulselayer.h
similarity index 99%
rename from daemon/src/audio/pulseaudio/pulselayer.h
rename to daemon/src/media/audio/pulseaudio/pulselayer.h
index 735f0d2bb37d56b0aee15dd27552606ab93b2fda..812577d07e047b67e7f1fb462ccc2e312f43a326 100644
--- a/daemon/src/audio/pulseaudio/pulselayer.h
+++ b/daemon/src/media/audio/pulseaudio/pulselayer.h
@@ -233,4 +233,3 @@ class PulseLayer : public AudioLayer {
}
#endif // PULSE_LAYER_H_
-
diff --git a/daemon/src/audio/recordable.cpp b/daemon/src/media/audio/recordable.cpp
similarity index 100%
rename from daemon/src/audio/recordable.cpp
rename to daemon/src/media/audio/recordable.cpp
diff --git a/daemon/src/audio/recordable.h b/daemon/src/media/audio/recordable.h
similarity index 100%
rename from daemon/src/audio/recordable.h
rename to daemon/src/media/audio/recordable.h
diff --git a/daemon/src/audio/resampler.cpp b/daemon/src/media/audio/resampler.cpp
similarity index 100%
rename from daemon/src/audio/resampler.cpp
rename to daemon/src/media/audio/resampler.cpp
diff --git a/daemon/src/audio/resampler.h b/daemon/src/media/audio/resampler.h
similarity index 100%
rename from daemon/src/audio/resampler.h
rename to daemon/src/media/audio/resampler.h
diff --git a/daemon/src/audio/ringbuffer.cpp b/daemon/src/media/audio/ringbuffer.cpp
similarity index 100%
rename from daemon/src/audio/ringbuffer.cpp
rename to daemon/src/media/audio/ringbuffer.cpp
diff --git a/daemon/src/audio/ringbuffer.h b/daemon/src/media/audio/ringbuffer.h
similarity index 100%
rename from daemon/src/audio/ringbuffer.h
rename to daemon/src/media/audio/ringbuffer.h
diff --git a/daemon/src/audio/ringbufferpool.cpp b/daemon/src/media/audio/ringbufferpool.cpp
similarity index 100%
rename from daemon/src/audio/ringbufferpool.cpp
rename to daemon/src/media/audio/ringbufferpool.cpp
diff --git a/daemon/src/audio/ringbufferpool.h b/daemon/src/media/audio/ringbufferpool.h
similarity index 100%
rename from daemon/src/audio/ringbufferpool.h
rename to daemon/src/media/audio/ringbufferpool.h
diff --git a/daemon/src/audio/sound/Makefile.am b/daemon/src/media/audio/sound/Makefile.am
similarity index 99%
rename from daemon/src/audio/sound/Makefile.am
rename to daemon/src/media/audio/sound/Makefile.am
index 0680471c72962021e672a1fc78dd13ab0d7765c3..f2ba42881f9dbdd0486d1665e87a603e152a4d9e 100644
--- a/daemon/src/audio/sound/Makefile.am
+++ b/daemon/src/media/audio/sound/Makefile.am
@@ -15,4 +15,3 @@ noinst_HEADERS = \
tonelist.h \
dtmfgenerator.h \
dtmf.h
-
diff --git a/daemon/src/audio/sound/audiofile.cpp b/daemon/src/media/audio/sound/audiofile.cpp
similarity index 100%
rename from daemon/src/audio/sound/audiofile.cpp
rename to daemon/src/media/audio/sound/audiofile.cpp
diff --git a/daemon/src/audio/sound/audiofile.h b/daemon/src/media/audio/sound/audiofile.h
similarity index 100%
rename from daemon/src/audio/sound/audiofile.h
rename to daemon/src/media/audio/sound/audiofile.h
diff --git a/daemon/src/audio/sound/dtmf.cpp b/daemon/src/media/audio/sound/dtmf.cpp
similarity index 100%
rename from daemon/src/audio/sound/dtmf.cpp
rename to daemon/src/media/audio/sound/dtmf.cpp
diff --git a/daemon/src/audio/sound/dtmf.h b/daemon/src/media/audio/sound/dtmf.h
similarity index 100%
rename from daemon/src/audio/sound/dtmf.h
rename to daemon/src/media/audio/sound/dtmf.h
diff --git a/daemon/src/audio/sound/dtmfgenerator.cpp b/daemon/src/media/audio/sound/dtmfgenerator.cpp
similarity index 100%
rename from daemon/src/audio/sound/dtmfgenerator.cpp
rename to daemon/src/media/audio/sound/dtmfgenerator.cpp
diff --git a/daemon/src/audio/sound/dtmfgenerator.h b/daemon/src/media/audio/sound/dtmfgenerator.h
similarity index 100%
rename from daemon/src/audio/sound/dtmfgenerator.h
rename to daemon/src/media/audio/sound/dtmfgenerator.h
diff --git a/daemon/src/audio/sound/tone.cpp b/daemon/src/media/audio/sound/tone.cpp
similarity index 100%
rename from daemon/src/audio/sound/tone.cpp
rename to daemon/src/media/audio/sound/tone.cpp
diff --git a/daemon/src/audio/sound/tone.h b/daemon/src/media/audio/sound/tone.h
similarity index 99%
rename from daemon/src/audio/sound/tone.h
rename to daemon/src/media/audio/sound/tone.h
index d70b64114eb52369d490b92030e815c038d0f0ad..cc75cdb4ebec47f3ded94aee780b8adf14990375 100644
--- a/daemon/src/audio/sound/tone.h
+++ b/daemon/src/media/audio/sound/tone.h
@@ -96,4 +96,3 @@ class Tone : public AudioLoop {
}
#endif // __TONE_H__
-
diff --git a/daemon/src/audio/sound/tonelist.cpp b/daemon/src/media/audio/sound/tonelist.cpp
similarity index 100%
rename from daemon/src/audio/sound/tonelist.cpp
rename to daemon/src/media/audio/sound/tonelist.cpp
diff --git a/daemon/src/audio/sound/tonelist.h b/daemon/src/media/audio/sound/tonelist.h
similarity index 100%
rename from daemon/src/audio/sound/tonelist.h
rename to daemon/src/media/audio/sound/tonelist.h
diff --git a/daemon/src/video/libav_deps.h b/daemon/src/media/libav_deps.h
similarity index 100%
rename from daemon/src/video/libav_deps.h
rename to daemon/src/media/libav_deps.h
diff --git a/daemon/src/video/libav_utils.cpp b/daemon/src/media/libav_utils.cpp
similarity index 99%
rename from daemon/src/video/libav_utils.cpp
rename to daemon/src/media/libav_utils.cpp
index 261fe46d800229190f39358ab9300aba3aad2fbf..261e58c4bdb9d87ef5b9ae72b5e42e2f90de72a3 100644
--- a/daemon/src/video/libav_utils.cpp
+++ b/daemon/src/media/libav_utils.cpp
@@ -31,7 +31,7 @@
*/
#include "libav_deps.h"
-#include "video_base.h"
+#include "video/video_base.h"
#include "logger.h"
#include <vector>
diff --git a/daemon/src/video/libav_utils.h b/daemon/src/media/libav_utils.h
similarity index 100%
rename from daemon/src/video/libav_utils.h
rename to daemon/src/media/libav_utils.h
diff --git a/daemon/src/video/.gitignore b/daemon/src/media/video/.gitignore
similarity index 100%
rename from daemon/src/video/.gitignore
rename to daemon/src/media/video/.gitignore
diff --git a/daemon/src/video/Makefile.am b/daemon/src/media/video/Makefile.am
similarity index 92%
rename from daemon/src/video/Makefile.am
rename to daemon/src/media/video/Makefile.am
index 2487ab35fc430c3b5c657f6862b1be36546db38b..ab5d484149f8b8cf94958a9b7dc4ceb336687079 100644
--- a/daemon/src/video/Makefile.am
+++ b/daemon/src/media/video/Makefile.am
@@ -17,8 +17,7 @@ libvideo_la_SOURCES = \
video_receive_thread.cpp video_receive_thread.h \
video_sender.cpp video_sender.h \
video_rtp_session.cpp video_rtp_session.h \
- shm_header.h video_provider.h \
- libav_utils.cpp libav_utils.h libav_deps.h
+ shm_header.h video_provider.h
libvideo_la_LIBADD = @LIBAVCODEC_LIBS@ @LIBAVFORMAT_LIBS@ @LIBAVDEVICE_LIBS@ @LIBSWSCALE_LIBS@ @LIBAVUTIL_LIBS@
diff --git a/daemon/src/video/TODO b/daemon/src/media/video/TODO
similarity index 100%
rename from daemon/src/video/TODO
rename to daemon/src/media/video/TODO
diff --git a/daemon/src/video/shm_header.h b/daemon/src/media/video/shm_header.h
similarity index 100%
rename from daemon/src/video/shm_header.h
rename to daemon/src/media/video/shm_header.h
diff --git a/daemon/src/video/shm_sink.cpp b/daemon/src/media/video/shm_sink.cpp
similarity index 100%
rename from daemon/src/video/shm_sink.cpp
rename to daemon/src/media/video/shm_sink.cpp
diff --git a/daemon/src/video/shm_sink.h b/daemon/src/media/video/shm_sink.h
similarity index 100%
rename from daemon/src/video/shm_sink.h
rename to daemon/src/media/video/shm_sink.h
diff --git a/daemon/src/video/socket_pair.cpp b/daemon/src/media/video/socket_pair.cpp
similarity index 100%
rename from daemon/src/video/socket_pair.cpp
rename to daemon/src/media/video/socket_pair.cpp
diff --git a/daemon/src/video/socket_pair.h b/daemon/src/media/video/socket_pair.h
similarity index 100%
rename from daemon/src/video/socket_pair.h
rename to daemon/src/media/video/socket_pair.h
diff --git a/daemon/src/video/test/.gitignore b/daemon/src/media/video/test/.gitignore
similarity index 100%
rename from daemon/src/video/test/.gitignore
rename to daemon/src/media/video/test/.gitignore
diff --git a/daemon/src/video/test/Makefile.am b/daemon/src/media/video/test/Makefile.am
similarity index 70%
rename from daemon/src/video/test/Makefile.am
rename to daemon/src/media/video/test/Makefile.am
index 89f73dd88c769b70060f4429f7ca5f3d787f318a..9f454aaf688177a204d25b582de2767fb1d4d300 100644
--- a/daemon/src/video/test/Makefile.am
+++ b/daemon/src/media/video/test/Makefile.am
@@ -1,19 +1,19 @@
-include ../../../globals.mak
+include $(top_srcdir)/globals.mak
TESTS=test_video_endpoint test_shm test_video_input test_video_rtp
check_PROGRAMS=test_video_endpoint test_video_rtp test_video_input test_shm
test_video_endpoint_SOURCES=test_video_endpoint.cpp test_video_endpoint.h
-test_video_endpoint_LDADD=$(top_builddir)/src/libring.la $(top_builddir)/src/video/libvideo.la $(YAML_LIBS)
+test_video_endpoint_LDADD=$(top_builddir)/src/libring.la $(top_builddir)/src/media/video/libvideo.la $(YAML_LIBS)
test_video_rtp_SOURCES=test_video_rtp.cpp
-test_video_rtp_LDADD=$(top_builddir)/src/libring.la $(top_builddir)/src/video/libvideo.la $(YAML_LIBS)
+test_video_rtp_LDADD=$(top_builddir)/src/libring.la $(top_builddir)/src/media/video/libvideo.la $(YAML_LIBS)
test_video_input_SOURCES=test_video_input.cpp test_video_input.h
-test_video_input_LDADD=$(top_builddir)/src/libring.la $(top_builddir)/src/video/libvideo.la $(YAML_LIBS)
+test_video_input_LDADD=$(top_builddir)/src/libring.la $(top_builddir)/src/media/video/libvideo.la $(YAML_LIBS)
test_shm_SOURCES=test_shm.cpp shm_src.cpp shm_src.h
-test_shm_LDADD=$(top_builddir)/src/libring.la $(top_builddir)/src/video/libvideo.la $(YAML_LIBS) -lrt
+test_shm_LDADD=$(top_builddir)/src/libring.la $(top_builddir)/src/media/video/libvideo.la $(YAML_LIBS) -lrt
test_shm_CXXFLAGS=$(AM_CXXFLAGS)
-AM_CXXFLAGS=-I$(top_srcdir)/src/video -I$(top_srcdir)/src
+AM_CXXFLAGS=-I$(top_srcdir)/src/media/video -I$(top_srcdir)/src
diff --git a/daemon/src/video/test/README b/daemon/src/media/video/test/README
similarity index 100%
rename from daemon/src/video/test/README
rename to daemon/src/media/video/test/README
diff --git a/daemon/src/video/test/make_rtp_stream.sh b/daemon/src/media/video/test/make_rtp_stream.sh
similarity index 100%
rename from daemon/src/video/test/make_rtp_stream.sh
rename to daemon/src/media/video/test/make_rtp_stream.sh
diff --git a/daemon/src/video/test/shm_src.cpp b/daemon/src/media/video/test/shm_src.cpp
similarity index 100%
rename from daemon/src/video/test/shm_src.cpp
rename to daemon/src/media/video/test/shm_src.cpp
diff --git a/daemon/src/video/test/shm_src.h b/daemon/src/media/video/test/shm_src.h
similarity index 100%
rename from daemon/src/video/test/shm_src.h
rename to daemon/src/media/video/test/shm_src.h
diff --git a/daemon/src/video/test/shmclient.cpp b/daemon/src/media/video/test/shmclient.cpp
similarity index 100%
rename from daemon/src/video/test/shmclient.cpp
rename to daemon/src/media/video/test/shmclient.cpp
diff --git a/daemon/src/video/test/test_shm.cpp b/daemon/src/media/video/test/test_shm.cpp
similarity index 100%
rename from daemon/src/video/test/test_shm.cpp
rename to daemon/src/media/video/test/test_shm.cpp
diff --git a/daemon/src/video/test/test_video_endpoint.cpp b/daemon/src/media/video/test/test_video_endpoint.cpp
similarity index 100%
rename from daemon/src/video/test/test_video_endpoint.cpp
rename to daemon/src/media/video/test/test_video_endpoint.cpp
diff --git a/daemon/src/video/test/test_video_endpoint.h b/daemon/src/media/video/test/test_video_endpoint.h
similarity index 100%
rename from daemon/src/video/test/test_video_endpoint.h
rename to daemon/src/media/video/test/test_video_endpoint.h
diff --git a/daemon/src/video/test/test_video_input.cpp b/daemon/src/media/video/test/test_video_input.cpp
similarity index 100%
rename from daemon/src/video/test/test_video_input.cpp
rename to daemon/src/media/video/test/test_video_input.cpp
diff --git a/daemon/src/video/test/test_video_input.h b/daemon/src/media/video/test/test_video_input.h
similarity index 100%
rename from daemon/src/video/test/test_video_input.h
rename to daemon/src/media/video/test/test_video_input.h
diff --git a/daemon/src/video/test/test_video_rtp.cpp b/daemon/src/media/video/test/test_video_rtp.cpp
similarity index 100%
rename from daemon/src/video/test/test_video_rtp.cpp
rename to daemon/src/media/video/test/test_video_rtp.cpp
diff --git a/daemon/src/video/v4l2/Makefile.am b/daemon/src/media/video/v4l2/Makefile.am
similarity index 100%
rename from daemon/src/video/v4l2/Makefile.am
rename to daemon/src/media/video/v4l2/Makefile.am
diff --git a/daemon/src/video/v4l2/video_device_impl.cpp b/daemon/src/media/video/v4l2/video_device_impl.cpp
similarity index 100%
rename from daemon/src/video/v4l2/video_device_impl.cpp
rename to daemon/src/media/video/v4l2/video_device_impl.cpp
diff --git a/daemon/src/video/v4l2/video_device_monitor_impl.cpp b/daemon/src/media/video/v4l2/video_device_monitor_impl.cpp
similarity index 100%
rename from daemon/src/video/v4l2/video_device_monitor_impl.cpp
rename to daemon/src/media/video/v4l2/video_device_monitor_impl.cpp
diff --git a/daemon/src/video/video_base.cpp b/daemon/src/media/video/video_base.cpp
similarity index 100%
rename from daemon/src/video/video_base.cpp
rename to daemon/src/media/video/video_base.cpp
diff --git a/daemon/src/video/video_base.h b/daemon/src/media/video/video_base.h
similarity index 100%
rename from daemon/src/video/video_base.h
rename to daemon/src/media/video/video_base.h
diff --git a/daemon/src/video/video_decoder.cpp b/daemon/src/media/video/video_decoder.cpp
similarity index 100%
rename from daemon/src/video/video_decoder.cpp
rename to daemon/src/media/video/video_decoder.cpp
diff --git a/daemon/src/video/video_decoder.h b/daemon/src/media/video/video_decoder.h
similarity index 100%
rename from daemon/src/video/video_decoder.h
rename to daemon/src/media/video/video_decoder.h
diff --git a/daemon/src/video/video_device.h b/daemon/src/media/video/video_device.h
similarity index 100%
rename from daemon/src/video/video_device.h
rename to daemon/src/media/video/video_device.h
diff --git a/daemon/src/video/video_device_monitor.cpp b/daemon/src/media/video/video_device_monitor.cpp
similarity index 100%
rename from daemon/src/video/video_device_monitor.cpp
rename to daemon/src/media/video/video_device_monitor.cpp
diff --git a/daemon/src/video/video_device_monitor.h b/daemon/src/media/video/video_device_monitor.h
similarity index 100%
rename from daemon/src/video/video_device_monitor.h
rename to daemon/src/media/video/video_device_monitor.h
diff --git a/daemon/src/video/video_encoder.cpp b/daemon/src/media/video/video_encoder.cpp
similarity index 100%
rename from daemon/src/video/video_encoder.cpp
rename to daemon/src/media/video/video_encoder.cpp
diff --git a/daemon/src/video/video_encoder.h b/daemon/src/media/video/video_encoder.h
similarity index 100%
rename from daemon/src/video/video_encoder.h
rename to daemon/src/media/video/video_encoder.h
diff --git a/daemon/src/video/video_input.cpp b/daemon/src/media/video/video_input.cpp
similarity index 100%
rename from daemon/src/video/video_input.cpp
rename to daemon/src/media/video/video_input.cpp
diff --git a/daemon/src/video/video_input.h b/daemon/src/media/video/video_input.h
similarity index 100%
rename from daemon/src/video/video_input.h
rename to daemon/src/media/video/video_input.h
diff --git a/daemon/src/video/video_mixer.cpp b/daemon/src/media/video/video_mixer.cpp
similarity index 100%
rename from daemon/src/video/video_mixer.cpp
rename to daemon/src/media/video/video_mixer.cpp
diff --git a/daemon/src/video/video_mixer.h b/daemon/src/media/video/video_mixer.h
similarity index 100%
rename from daemon/src/video/video_mixer.h
rename to daemon/src/media/video/video_mixer.h
diff --git a/daemon/src/video/video_provider.h b/daemon/src/media/video/video_provider.h
similarity index 100%
rename from daemon/src/video/video_provider.h
rename to daemon/src/media/video/video_provider.h
diff --git a/daemon/src/video/video_receive_thread.cpp b/daemon/src/media/video/video_receive_thread.cpp
similarity index 100%
rename from daemon/src/video/video_receive_thread.cpp
rename to daemon/src/media/video/video_receive_thread.cpp
diff --git a/daemon/src/video/video_receive_thread.h b/daemon/src/media/video/video_receive_thread.h
similarity index 100%
rename from daemon/src/video/video_receive_thread.h
rename to daemon/src/media/video/video_receive_thread.h
diff --git a/daemon/src/video/video_rtp_session.cpp b/daemon/src/media/video/video_rtp_session.cpp
similarity index 100%
rename from daemon/src/video/video_rtp_session.cpp
rename to daemon/src/media/video/video_rtp_session.cpp
diff --git a/daemon/src/video/video_rtp_session.h b/daemon/src/media/video/video_rtp_session.h
similarity index 100%
rename from daemon/src/video/video_rtp_session.h
rename to daemon/src/media/video/video_rtp_session.h
diff --git a/daemon/src/video/video_scaler.cpp b/daemon/src/media/video/video_scaler.cpp
similarity index 100%
rename from daemon/src/video/video_scaler.cpp
rename to daemon/src/media/video/video_scaler.cpp
diff --git a/daemon/src/video/video_scaler.h b/daemon/src/media/video/video_scaler.h
similarity index 100%
rename from daemon/src/video/video_scaler.h
rename to daemon/src/media/video/video_scaler.h
diff --git a/daemon/src/video/video_sender.cpp b/daemon/src/media/video/video_sender.cpp
similarity index 100%
rename from daemon/src/video/video_sender.cpp
rename to daemon/src/media/video/video_sender.cpp
diff --git a/daemon/src/video/video_sender.h b/daemon/src/media/video/video_sender.h
similarity index 100%
rename from daemon/src/video/video_sender.h
rename to daemon/src/media/video/video_sender.h
diff --git a/daemon/src/ringdht/ringaccount.cpp b/daemon/src/ringdht/ringaccount.cpp
index 61eecb259ed501d056911e38aac7148088ee47b2..c7c884110801285c84e719dedfbe5ef3112b0cb3 100644
--- a/daemon/src/ringdht/ringaccount.cpp
+++ b/daemon/src/ringdht/ringaccount.cpp
@@ -54,7 +54,7 @@
#include "manager.h"
#ifdef RING_VIDEO
-#include "video/libav_utils.h"
+#include "libav_utils.h"
#endif
#include "fileutils.h"
@@ -589,7 +589,7 @@ void RingAccount::doRegister()
username_ = dht_.getId().toString();
- Manager::instance().registerEventHandler((uintptr_t)this, std::bind(&dht::DhtRunner::loop, &dht_));
+ Manager::instance().registerEventHandler((uintptr_t)this, [this](){ dht_.loop(); });
setRegistrationState(RegistrationState::TRYING);
dht_.bootstrap(loadNodes());
diff --git a/daemon/src/ringdht/sip_transport_ice.h b/daemon/src/ringdht/sip_transport_ice.h
index 07ff5d3f7828161636995d4763e7e63445e7fd0d..3665b15f4b77e96344284e1b53762344a6807c0a 100644
--- a/daemon/src/ringdht/sip_transport_ice.h
+++ b/daemon/src/ringdht/sip_transport_ice.h
@@ -35,6 +35,8 @@
#include <pjsip.h>
#include <pj/pool.h>
+
+#include <functional>
#include <memory>
namespace ring {
diff --git a/daemon/src/sip/sdp.cpp b/daemon/src/sip/sdp.cpp
index 4bdaffe53c58b1c1b1581c6ae21e30455817b5e3..1dbd30537b89c3ef6f141d2a53721095335fb5a4 100644
--- a/daemon/src/sip/sdp.cpp
+++ b/daemon/src/sip/sdp.cpp
@@ -40,10 +40,7 @@
#include "sipvoiplink.h"
#include "manager.h"
#include "logger.h"
-
-#ifdef RING_VIDEO
-#include "video/libav_utils.h"
-#endif
+#include "libav_utils.h"
#include <algorithm>
#include <cassert>
@@ -1138,7 +1135,6 @@ bool Sdp::getOutgoingVideoSettings(map<string, string> &args) const
return false;
}
-#ifndef USE_CCRTP
bool Sdp::getOutgoingAudioSettings(map<string, string> &args) const
{
string codec(getOutgoingAudioCodec());
@@ -1180,4 +1176,3 @@ bool Sdp::getOutgoingAudioSettings(map<string, string> &args) const
}
return false;
}
-#endif
diff --git a/daemon/src/sip/sipaccount.cpp b/daemon/src/sip/sipaccount.cpp
index 72f55af2a9d01e1a97b99cd574eb4d40a1f9edb1..b1b8ff4c092a6ce23b133d0d19bdb68de44d86f1 100644
--- a/daemon/src/sip/sipaccount.cpp
+++ b/daemon/src/sip/sipaccount.cpp
@@ -58,7 +58,7 @@
#include "manager.h"
#ifdef RING_VIDEO
-#include "video/libav_utils.h"
+#include "libav_utils.h"
#endif
#include <unistd.h>
diff --git a/daemon/src/sip/sipaccountbase.cpp b/daemon/src/sip/sipaccountbase.cpp
index 042926fc18e12a5bb003c89bc594424b996610a0..91a991c4de7ee551d5a3de05d14e3810c4693479 100644
--- a/daemon/src/sip/sipaccountbase.cpp
+++ b/daemon/src/sip/sipaccountbase.cpp
@@ -33,7 +33,7 @@
#include "sipvoiplink.h"
#ifdef RING_VIDEO
-#include "video/libav_utils.h"
+#include "libav_utils.h"
#endif
#include "account_schema.h"
diff --git a/daemon/src/sip/sipcall.cpp b/daemon/src/sip/sipcall.cpp
index 21c31bbf3a98bc6c03b94eebdf1e5151fb807940..33dc2c3ab78570f9778928f0f601ec0e68bcc37e 100644
--- a/daemon/src/sip/sipcall.cpp
+++ b/daemon/src/sip/sipcall.cpp
@@ -45,12 +45,8 @@
using namespace ring;
-#if USE_CCRTP
-#include "audio/audiortp/audio_rtp_factory.h" // for AudioRtpFactoryException
-#else
#include "audio/audiortp/avformat_rtp_session.h"
-#endif
#include "client/callmanager.h"
#if HAVE_INSTANT_MESSAGING
@@ -124,11 +120,8 @@ dtmfSend(SIPCall &call, char code, const std::string &dtmf)
SIPCall::SIPCall(SIPAccountBase& account, const std::string& id, Call::CallType type)
: Call(account, id, type)
-#if USE_CCRTP
- , audiortp_(this)
-#else
- , avformatrtp_(new ring::AVFormatRtpSession(id, /* FIXME: These are video! */ getSettings()))
-#endif
+ //, avformatrtp_(new ring::AVFormatRtpSession(id, /* FIXME: These are video! */ getSettings()))
+ , avformatrtp_(new ring::AVFormatRtpSession(id, *new std::map<std::string, std::string>))
#ifdef RING_VIDEO
// The ID is used to associate video streams to calls
, videortp_(id, getSettings())
@@ -377,11 +370,7 @@ SIPCall::refuse()
if (!isIncoming() or getConnectionState() == Call::CONNECTED or !inv)
return;
-#if USE_CCRTP
- getAudioRtp().stop();
-#else
avformatrtp_->stop();
-#endif
pjsip_tx_data *tdata;
@@ -575,12 +564,7 @@ SIPCall::onhold()
if (not setState(Call::HOLD))
return;
-#if USE_CCRTP
- audiortp_.saveLocalContext();
- audiortp_.stop();
-#else
avformatrtp_->stop();
-#endif
#ifdef RING_VIDEO
videortp_.stop();
#endif
@@ -821,15 +805,6 @@ void
SIPCall::startAllMedia()
{
auto& remoteIP = sdp_->getRemoteIP();
-#if USE_CCRTP
- try {
- audiortp_.updateDestinationIpAddress();
- } catch (const AudioRtpFactoryException &e) {
- RING_ERR("%s", e.what());
- }
-
- audiortp_.setDtmfPayloadType(sdp_->getTelephoneEventType());
-#else
avformatrtp_->updateSDP(*sdp_);
avformatrtp_->updateDestination(remoteIP, sdp_->getRemoteAudioPort());
if (isIceRunning()) {
@@ -840,7 +815,6 @@ SIPCall::startAllMedia()
const auto localAudioPort = sdp_->getLocalAudioPort();
avformatrtp_->start(localAudioPort ? localAudioPort : sdp_->getRemoteAudioPort());
}
-#endif
#ifdef RING_VIDEO
auto remoteVideoPort = sdp_->getRemoteVideoPort();
@@ -956,11 +930,7 @@ void
SIPCall::stopAllMedias()
{
RING_DBG("SIPCall %s: stopping all medias", getCallId().c_str());
-#if USE_CCRTP
- audiortp_.stop();
-#else
avformatrtp_->stop();
-#endif
#ifdef RING_VIDEO
videortp_.stop();
#endif
diff --git a/daemon/src/sip/sipvoiplink.cpp b/daemon/src/sip/sipvoiplink.cpp
index c901ddd0ceae9b34061443deacfa75e24ff4562f..349b49145b9cb689fc06127d08d19d3eead5fe67 100644
--- a/daemon/src/sip/sipvoiplink.cpp
+++ b/daemon/src/sip/sipvoiplink.cpp
@@ -55,9 +55,7 @@
#include "im/instant_messaging.h"
#endif
-#ifndef USE_CCRTP
#include "audio/audiortp/avformat_rtp_session.h"
-#endif
#ifdef RING_VIDEO
#include "video/video_rtp_session.h"