link lib webrtc

CMakeLists.txt

@@ -8,7 +8,7 @@ add_subdirectory(libtgvoip)
 find_package(PkgConfig REQUIRED)
 find_package(Threads REQUIRED)
 find_package(Td 1.8.0 REQUIRED)
-find_package(spdlog 1.11.0 REQUIRED)
+find_package(spdlog 1.9.2 REQUIRED)
 pkg_check_modules(PJSIP libpjproject>=2.8 REQUIRED)
 pkg_check_modules(OPUS opus REQUIRED)
 

libtgvoip/CMakeLists.txt

@@ -4,7 +4,9 @@ find_package(PkgConfig REQUIRED)
 pkg_check_modules(OPUS opus REQUIRED)
 pkg_check_modules(OPENSSL openssl REQUIRED)
 pkg_check_modules(PJSIP libpjproject>=2.8 REQUIRED)
-find_package(spdlog 1.11.0)
+pkg_check_modules(FMT fmt REQUIRED)
+pkg_check_modules(WEBRTC webrtc-audio-processing REQUIRED)
+find_package(spdlog 1.9.2)
 
 add_library(libtgvoip STATIC
         BlockingQueue.cpp
@@ -675,4 +677,10 @@ if (${spdlog_FOUND})
             TGVOIP_USE_SPDLOG)
 else ()
     message(STATUS "Could NOT find spdlog")
-endif ()
+endif ()
+
+target_link_libraries(libtgvoip PRIVATE
+        ${PJSIP_LIBRARIES}
+        ${OPUS_LIBRARIES}
+        ${FMT_LIBRARIES}
+        ${WEBRTC_LIBRARIES})

libtgvoip/os/posix/NetworkSocketPosix.cpp

@@ -29,7 +29,7 @@ extern jclass jniUtilitiesClass;
 
 // fix of undef in pjsip
 #if defined(TGVOIP_USE_SOFTWARE_AUDIO)
-#define s6_addr			__in6_u.__u6_addr8
+// #define s6_addr			__in6_u.__u6_addr8
 #endif
 
 using namespace tgvoip;

tg2sip/sip.cpp

@@ -185,10 +185,10 @@ void Client::init_pj_endpoint(Settings &settings, LogWriter *sip_log_writer) {
     // and TG audio port clock rate so we MUST force
     // using 48kHz codecs for all SIP calls
     std::string codecId = settings.raw_pcm() ? "L16/48000/1" : "opus/48000/2";
-    CodecInfoVector codecVector = ep.codecEnum();
+    CodecInfoVector2 codecVector = ep.codecEnum2();
 
     for (auto const &value : codecVector) {
-        ep.codecSetPriority(value->codecId, (pj_uint8_t) (value->codecId == codecId ? 255 : 0));
+        ep.codecSetPriority(value.codecId, (pj_uint8_t) (value.codecId == codecId ? 255 : 0));
     }
 
     TransportConfig t_cfg;