tg2sip-alpine/alpine.patch

135 lines
4.1 KiB
Diff

diff --git a/CMakeLists.txt b/CMakeLists.txt
index 8507dbe..8577efb 100644
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -7,8 +7,8 @@ add_subdirectory(libtgvoip)
find_package(PkgConfig REQUIRED)
find_package(Threads REQUIRED)
-find_package(Td 1.7.10 REQUIRED)
-find_package(spdlog 0.17 REQUIRED)
+find_package(Td 1.8.0 REQUIRED)
+find_package(spdlog 1.9.2 REQUIRED)
pkg_check_modules(PJSIP libpjproject>=2.8 REQUIRED)
pkg_check_modules(OPUS opus REQUIRED)
diff --git a/include/boost/sml.hpp b/include/boost/sml.hpp
index a28bddf..3245862 100644
--- a/include/boost/sml.hpp
+++ b/include/boost/sml.hpp
@@ -2525,7 +2525,7 @@ BOOST_SML_NAMESPACE_END
#if defined(__clang__)
#pragma clang diagnostic pop
#elif defined(__GNUC__)
-#undef __has_builtin
+//#undef __has_builtin
#pragma GCC diagnostic pop
#elif defined(_MSC_VER)
#undef __has_builtin
diff --git a/libtgvoip/CMakeLists.txt b/libtgvoip/CMakeLists.txt
index a9658fb..237bcd7 100644
--- a/libtgvoip/CMakeLists.txt
+++ b/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 0.17)
+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 ()
\ No newline at end of file
+endif ()
+
+target_link_libraries(libtgvoip PRIVATE
+ ${PJSIP_LIBRARIES}
+ ${OPUS_LIBRARIES}
+ ${FMT_LIBRARIES}
+ ${WEBRTC_LIBRARIES})
diff --git a/libtgvoip/audio/AudioIO.cpp b/libtgvoip/audio/AudioIO.cpp
index ad2ec82..23faf89 100644
--- a/libtgvoip/audio/AudioIO.cpp
+++ b/libtgvoip/audio/AudioIO.cpp
@@ -85,6 +85,7 @@ AudioIO* AudioIO::Create(std::string inputDevice, std::string outputDevice){
#endif
#endif
#endif
+ return NULL;
}
bool AudioIO::Failed(){
diff --git a/libtgvoip/os/posix/NetworkSocketPosix.cpp b/libtgvoip/os/posix/NetworkSocketPosix.cpp
index 26289d5..16e28b1 100644
--- a/libtgvoip/os/posix/NetworkSocketPosix.cpp
+++ b/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;
diff --git a/tg2sip/logging.cpp b/tg2sip/logging.cpp
index 8a51353..287fb03 100755
--- a/tg2sip/logging.cpp
+++ b/tg2sip/logging.cpp
@@ -18,6 +18,7 @@
#include <iostream>
#include <td/telegram/Log.h>
#include "logging.h"
+#include <spdlog/sinks/rotating_file_sink.h>
void init_logging(Settings &settings) {
diff --git a/tg2sip/queue.h b/tg2sip/queue.h
index bfe1fa0..3e9c555 100755
--- a/tg2sip/queue.h
+++ b/tg2sip/queue.h
@@ -20,6 +20,7 @@
#include <mutex>
#include <queue>
+#include <optional>
#include <condition_variable>
template<typename T>
diff --git a/tg2sip/sip.cpp b/tg2sip/sip.cpp
index 06672a6..306f650 100755
--- a/tg2sip/sip.cpp
+++ b/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;
diff --git a/tg2sip/utils.cpp b/tg2sip/utils.cpp
index c8da02b..09a330f 100644
--- a/tg2sip/utils.cpp
+++ b/tg2sip/utils.cpp
@@ -17,5 +17,6 @@
#include <algorithm>
#include "utils.h"
+#include <string>
bool is_digits(const std::string &str) { return std::all_of(str.begin(), str.end(), ::isdigit); };
\ No newline at end of file