pjaudio and spdlog support

libtgvoip/CMakeLists.txt

@@ -0,0 +1,205 @@
+cmake_minimum_required(VERSION 3.9)
+
+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)
+
+add_library(libtgvoip STATIC
+        BlockingQueue.cpp
+        BlockingQueue.h
+        Buffers.cpp
+        Buffers.h
+        CongestionControl.cpp
+        CongestionControl.h
+        EchoCanceller.cpp
+        EchoCanceller.h
+        JitterBuffer.cpp
+        JitterBuffer.h
+        logging.cpp
+        logging.h
+        MediaStreamItf.cpp
+        MediaStreamItf.h
+        OpusDecoder.cpp
+        OpusDecoder.h
+        OpusEncoder.cpp
+        OpusEncoder.h
+        threading.h
+        VoIPController.cpp
+        VoIPGroupController.cpp
+        VoIPController.h
+        PrivateDefines.h
+        VoIPServerConfig.cpp
+        VoIPServerConfig.h
+        audio/AudioInput.cpp
+        audio/AudioInput.h
+        audio/AudioOutput.cpp
+        audio/AudioOutput.h
+        audio/Resampler.cpp
+        audio/Resampler.h
+        NetworkSocket.cpp
+        NetworkSocket.h
+        PacketReassembler.cpp
+        PacketReassembler.h
+        MessageThread.cpp
+        MessageThread.h
+        audio/AudioIO.cpp
+        audio/AudioIO.h
+
+        # POSIX
+        os/posix/NetworkSocketPosix.cpp
+        os/posix/NetworkSocketPosix.h
+
+        webrtc_dsp/webrtc/base/array_view.h
+        webrtc_dsp/webrtc/base/atomicops.h
+        webrtc_dsp/webrtc/base/basictypes.h
+        webrtc_dsp/webrtc/base/checks.cc
+        webrtc_dsp/webrtc/base/checks.h
+        webrtc_dsp/webrtc/base/constructormagic.h
+        webrtc_dsp/webrtc/base/safe_compare.h
+        webrtc_dsp/webrtc/base/safe_conversions.h
+        webrtc_dsp/webrtc/base/safe_conversions_impl.h
+        webrtc_dsp/webrtc/base/sanitizer.h
+        webrtc_dsp/webrtc/base/stringutils.cc
+        webrtc_dsp/webrtc/base/stringutils.h
+        webrtc_dsp/webrtc/base/type_traits.h
+        webrtc_dsp/webrtc/common_audio/audio_util.cc
+        webrtc_dsp/webrtc/common_audio/channel_buffer.cc
+        webrtc_dsp/webrtc/common_audio/channel_buffer.h
+        webrtc_dsp/webrtc/common_audio/fft4g.c
+        webrtc_dsp/webrtc/common_audio/fft4g.h
+        webrtc_dsp/webrtc/common_audio/include/audio_util.h
+        webrtc_dsp/webrtc/common_audio/ring_buffer.c
+        webrtc_dsp/webrtc/common_audio/ring_buffer.h
+        webrtc_dsp/webrtc/common_audio/signal_processing/auto_corr_to_refl_coef.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/auto_correlation.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/complex_bit_reverse.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/complex_fft.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/complex_fft_tables.h
+        webrtc_dsp/webrtc/common_audio/signal_processing/copy_set_operations.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/cross_correlation.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/division_operations.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/dot_product_with_scale.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/downsample_fast.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/energy.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/filter_ar.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/filter_ma_fast_q12.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/get_hanning_window.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/get_scaling_square.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/ilbc_specific_functions.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/include/real_fft.h
+        webrtc_dsp/webrtc/common_audio/signal_processing/include/signal_processing_library.h
+        webrtc_dsp/webrtc/common_audio/signal_processing/include/spl_inl.h
+        webrtc_dsp/webrtc/common_audio/signal_processing/include/spl_inl_mips.h
+        webrtc_dsp/webrtc/common_audio/signal_processing/levinson_durbin.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/lpc_to_refl_coef.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/min_max_operations.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/randomization_functions.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/real_fft.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/refl_coef_to_lpc.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/resample.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/resample_48khz.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/resample_by_2.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/resample_by_2_internal.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/resample_by_2_internal.h
+        webrtc_dsp/webrtc/common_audio/signal_processing/resample_fractional.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/spl_init.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/spl_inl.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/spl_sqrt.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/spl_sqrt_floor.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/splitting_filter_impl.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
+        webrtc_dsp/webrtc/common_audio/signal_processing/vector_scaling_operations.c
+        webrtc_dsp/webrtc/common_audio/sparse_fir_filter.cc
+        webrtc_dsp/webrtc/common_audio/sparse_fir_filter.h
+        webrtc_dsp/webrtc/common_audio/wav_file.cc
+        webrtc_dsp/webrtc/common_audio/wav_file.h
+        webrtc_dsp/webrtc/common_audio/wav_header.cc
+        webrtc_dsp/webrtc/common_audio/wav_header.h
+        webrtc_dsp/webrtc/modules/audio_processing/aec/aec_common.h
+        webrtc_dsp/webrtc/modules/audio_processing/aec/aec_core.cc
+        webrtc_dsp/webrtc/modules/audio_processing/aec/aec_core.h
+        webrtc_dsp/webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h
+        webrtc_dsp/webrtc/modules/audio_processing/aec/aec_core_sse2.cc
+        webrtc_dsp/webrtc/modules/audio_processing/aec/aec_resampler.cc
+        webrtc_dsp/webrtc/modules/audio_processing/aec/aec_resampler.h
+        webrtc_dsp/webrtc/modules/audio_processing/aec/echo_cancellation.cc
+        webrtc_dsp/webrtc/modules/audio_processing/aec/echo_cancellation.h
+        webrtc_dsp/webrtc/modules/audio_processing/aecm/aecm_core.cc
+        webrtc_dsp/webrtc/modules/audio_processing/aecm/aecm_core.h
+        webrtc_dsp/webrtc/modules/audio_processing/aecm/aecm_core_c.cc
+        webrtc_dsp/webrtc/modules/audio_processing/aecm/aecm_defines.h
+        webrtc_dsp/webrtc/modules/audio_processing/aecm/echo_control_mobile.cc
+        webrtc_dsp/webrtc/modules/audio_processing/aecm/echo_control_mobile.h
+        webrtc_dsp/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
+        webrtc_dsp/webrtc/modules/audio_processing/agc/legacy/analog_agc.h
+        webrtc_dsp/webrtc/modules/audio_processing/agc/legacy/digital_agc.c
+        webrtc_dsp/webrtc/modules/audio_processing/agc/legacy/digital_agc.h
+        webrtc_dsp/webrtc/modules/audio_processing/agc/legacy/gain_control.h
+        webrtc_dsp/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
+        webrtc_dsp/webrtc/modules/audio_processing/logging/apm_data_dumper.h
+        webrtc_dsp/webrtc/modules/audio_processing/ns/defines.h
+        webrtc_dsp/webrtc/modules/audio_processing/ns/noise_suppression.c
+        webrtc_dsp/webrtc/modules/audio_processing/ns/noise_suppression.h
+        webrtc_dsp/webrtc/modules/audio_processing/ns/noise_suppression_x.c
+        webrtc_dsp/webrtc/modules/audio_processing/ns/noise_suppression_x.h
+        webrtc_dsp/webrtc/modules/audio_processing/ns/ns_core.c
+        webrtc_dsp/webrtc/modules/audio_processing/ns/ns_core.h
+        webrtc_dsp/webrtc/modules/audio_processing/ns/nsx_core.c
+        webrtc_dsp/webrtc/modules/audio_processing/ns/nsx_core.h
+        webrtc_dsp/webrtc/modules/audio_processing/ns/nsx_core_c.c
+        webrtc_dsp/webrtc/modules/audio_processing/ns/nsx_defines.h
+        webrtc_dsp/webrtc/modules/audio_processing/ns/windows_private.h
+        webrtc_dsp/webrtc/modules/audio_processing/splitting_filter.cc
+        webrtc_dsp/webrtc/modules/audio_processing/splitting_filter.h
+        webrtc_dsp/webrtc/modules/audio_processing/three_band_filter_bank.cc
+        webrtc_dsp/webrtc/modules/audio_processing/three_band_filter_bank.h
+        webrtc_dsp/webrtc/modules/audio_processing/utility/block_mean_calculator.cc
+        webrtc_dsp/webrtc/modules/audio_processing/utility/block_mean_calculator.h
+        webrtc_dsp/webrtc/modules/audio_processing/utility/delay_estimator.cc
+        webrtc_dsp/webrtc/modules/audio_processing/utility/delay_estimator.h
+        webrtc_dsp/webrtc/modules/audio_processing/utility/delay_estimator_internal.h
+        webrtc_dsp/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.cc
+        webrtc_dsp/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h
+        webrtc_dsp/webrtc/modules/audio_processing/utility/ooura_fft.cc
+        webrtc_dsp/webrtc/modules/audio_processing/utility/ooura_fft.h
+        webrtc_dsp/webrtc/modules/audio_processing/utility/ooura_fft_sse2.cc
+        webrtc_dsp/webrtc/modules/audio_processing/utility/ooura_fft_tables_common.h
+        webrtc_dsp/webrtc/system_wrappers/include/asm_defines.h
+        webrtc_dsp/webrtc/system_wrappers/include/compile_assert_c.h
+        webrtc_dsp/webrtc/system_wrappers/include/cpu_features_wrapper.h
+        webrtc_dsp/webrtc/system_wrappers/include/metrics.h
+        webrtc_dsp/webrtc/system_wrappers/source/cpu_features.cc
+        webrtc_dsp/webrtc/typedefs.h
+
+        #SOFTWARE AUDIO
+        audio/SoftwareAudioInput.h
+        audio/SoftwareAudioInput.cpp
+        audio/SoftwareAudioOutput.h
+        audio/SoftwareAudioOutput.cpp)
+
+set_property(TARGET libtgvoip PROPERTY CXX_STANDARD 11)
+
+target_include_directories(libtgvoip PRIVATE
+        webrtc_dsp
+        ${OPUS_INCLUDE_DIRS}
+        ${OPENSSL_INCLUDE_DIRS}
+        ${PJSIP_INCLUDE_DIRS})
+
+target_compile_definitions(libtgvoip PRIVATE
+        WEBRTC_APM_DEBUG_DUMP=0
+        WEBRTC_POSIX
+        DEFAULT_THREAD_PRIORITY)
+
+target_compile_definitions(libtgvoip PUBLIC
+        TGVOIP_USE_DESKTOP_DSP
+        TGVOIP_USE_SOFTWARE_AUDIO)
+
+if (${spdlog_FOUND})
+    target_compile_definitions(libtgvoip PUBLIC
+            TGVOIP_USE_SPDLOG)
+else ()
+    message(STATUS "Could NOT find spdlog")
+endif ()

libtgvoip/VoIPController.cpp

@@ -123,6 +123,9 @@ CryptoFunctions VoIPController::crypto; // set it yourself upon initialization
 
 
 extern FILE* tgvoipLogFile;
+#ifdef TGVOIP_USE_SPDLOG
+extern std::shared_ptr<spdlog::logger> _voip_logger;
+#endif
 
 VoIPController::VoIPController() : activeNetItfName(""),
 								   currentAudioInput("default"),
@@ -130,6 +133,9 @@ VoIPController::VoIPController() : activeNetItfName(""),
 								   proxyAddress(""),
 								   proxyUsername(""),
 								   proxyPassword(""){
+#ifdef TGVOIP_USE_SPDLOG
+    _voip_logger = spdlog::get("tgvoip");
+#endif
     seq=1;
 	lastRemoteSeq=0;
 	state=STATE_WAIT_INIT;
@@ -245,10 +251,42 @@ VoIPController::VoIPController() : activeNetItfName(""),
 	vstm.codec=CODEC_AVC;
 	vstm.enabled=1;
 	outgoingStreams.push_back(vstm);*/
+
+#if defined(TGVOIP_USE_SOFTWARE_AUDIO)
+	pj_thread_t *tmp_thread = nullptr;
+	pj_thread_desc tmp_thread_desc;
+	if (!pj_thread_is_registered()) {
+		pj_bzero(&tmp_thread_desc, sizeof(tmp_thread_desc));
+		pj_status_t status;
+		status = pj_thread_register("tmp", tmp_thread_desc, &tmp_thread);
+		if (status != PJ_SUCCESS) {
+			LOGE("Temp constructor thread PJ register failed");
+		}
+	}
+
+    softwareMediaInput = new tgvoip::audio::SoftwareAudioInput();
+    softwareMediaOutput = new tgvoip::audio::SoftwareAudioOutput();
+
+    if (tmp_thread) {
+        pj_thread_destroy(tmp_thread);
+    }
+#endif
 }
 
 VoIPController::~VoIPController(){
 	LOGD("Entered VoIPController::~VoIPController");
+#if defined(TGVOIP_USE_SOFTWARE_AUDIO)
+    pj_thread_t *tmp_thread = nullptr;
+    pj_thread_desc tmp_thread_desc;
+    if (!pj_thread_is_registered()) {
+        pj_bzero(&tmp_thread_desc, sizeof(tmp_thread_desc));
+        pj_status_t status;
+        status = pj_thread_register("tmp", tmp_thread_desc, &tmp_thread);
+        if (status != PJ_SUCCESS) {
+            LOGE("Temp destructor thread PJ register failed");
+        }
+    }
+#endif
 	if(!stopping){
 		LOGE("!!!!!!!!!!!!!!!!!!!! CALL controller->Stop() BEFORE DELETING THE CONTROLLER OBJECT !!!!!!!!!!!!!!!!!!!!!!!1");
 		abort();
@@ -289,6 +327,14 @@ VoIPController::~VoIPController(){
 			delete (*stm)->decoder;
 		}
 	}*/
+#if defined(TGVOIP_USE_SOFTWARE_AUDIO)
+    delete softwareMediaInput;
+    delete softwareMediaOutput;
+
+    if(tmp_thread) {
+        pj_thread_destroy(tmp_thread);
+    }
+#endif
 	LOGD("before delete echo canceller");
 	if(echoCanceller){
 		echoCanceller->Stop();
@@ -2648,9 +2694,14 @@ void VoIPController::SendUdpPing(shared_ptr<Endpoint> endpoint){
 void VoIPController::StartAudio(){
 	shared_ptr<Stream>& outgoingAudioStream=outgoingStreams[0];
 	LOGI("before create audio io");
+#if defined(TGVOIP_USE_SOFTWARE_AUDIO)
+    audioInput = softwareMediaInput;
+    audioOutput = softwareMediaOutput;
+#else
 	audioIO=audio::AudioIO::Create();
 	audioInput=audioIO->GetInput();
 	audioOutput=audioIO->GetOutput();
+#endif
 	LOGI("AEC: %d NS: %d AGC: %d", config.enableAEC, config.enableNS, config.enableAGC);
 	echoCanceller=new EchoCanceller(config.enableAEC, config.enableNS, config.enableAGC);
 	encoder=new OpusEncoder(audioInput, peerVersion>=6);

libtgvoip/VoIPController.h

@@ -21,6 +21,11 @@
 #include <unordered_map>
 #include <memory>
 #include <atomic>
+#if defined(TGVOIP_USE_SOFTWARE_AUDIO)
+#include <pjsua2.hpp>
+#include "audio/SoftwareAudioInput.h"
+#include "audio/SoftwareAudioOutput.h"
+#endif
 #include "audio/AudioInput.h"
 #include "BlockingQueue.h"
 #include "audio/AudioOutput.h"
@@ -381,6 +386,11 @@ namespace tgvoip{
 			return 0.0f;
 		};
 
+#if defined(TGVOIP_USE_SOFTWARE_AUDIO)
+        pj::AudioMedia * AudioMediaInput() {return softwareMediaInput;};
+        pj::AudioMedia * AudioMediaOutput() {return softwareMediaOutput;};
+#endif
+
 	private:
 		struct Stream;
 		struct UnacknowledgedExtraData;
@@ -631,6 +641,10 @@ namespace tgvoip{
 		double p2pToRelaySwitchThreshold;
 		double relayToP2pSwitchThreshold;
 		double reconnectingTimeout;
+#if defined(TGVOIP_USE_SOFTWARE_AUDIO)
+        tgvoip::audio::SoftwareAudioInput* softwareMediaInput;
+        tgvoip::audio::SoftwareAudioOutput* softwareMediaOutput;
+#endif
 
 	public:
 #ifdef __APPLE__

libtgvoip/audio/AudioIO.cpp

@@ -12,6 +12,7 @@
 #include "config.h"
 #endif
 
+#if !defined(TGVOIP_USE_SOFTWARE_AUDIO)
 #if defined(TGVOIP_USE_CALLBACK_AUDIO_IO)
 #include "AudioIOCallback.h"
 #elif defined(__ANDROID__)
@@ -42,12 +43,14 @@
 #else
 #error "Unsupported operating system"
 #endif
+#endif
 
 using namespace tgvoip;
 using namespace tgvoip::audio;
 using namespace std;
 
 shared_ptr<AudioIO> AudioIO::Create(){
+#if !defined(TGVOIP_USE_SOFTWARE_AUDIO)
 	std::string inputDevice="default", outputDevice="default";
 #if defined(TGVOIP_USE_CALLBACK_AUDIO_IO)
 	return std::make_shared<AudioIOCallback>();
@@ -81,6 +84,7 @@ shared_ptr<AudioIO> AudioIO::Create(){
 	return std::make_shared<AudioPulse>(inputDevice, outputDevice);
 #endif
 #endif
+#endif
 }
 
 bool AudioIO::Failed(){

libtgvoip/audio/AudioInput.cpp

@@ -11,6 +11,7 @@
 #include "config.h"
 #endif
 
+#if !defined(TGVOIP_USE_SOFTWARE_AUDIO)
 #if defined(TGVOIP_USE_CALLBACK_AUDIO_IO)
 // nothing
 #elif defined(__ANDROID__)
@@ -36,6 +37,7 @@
 #else
 #error "Unsupported operating system"
 #endif
+#endif
 
 using namespace tgvoip;
 using namespace tgvoip::audio;
@@ -60,6 +62,7 @@ bool AudioInput::IsInitialized(){
 }
 
 void AudioInput::EnumerateDevices(std::vector<AudioInputDevice>& devs){
+#if !defined(TGVOIP_USE_SOFTWARE_AUDIO)
 #if defined(TGVOIP_USE_CALLBACK_AUDIO_IO)
 	// not supported
 #elif defined(__APPLE__) && TARGET_OS_OSX
@@ -82,6 +85,7 @@ void AudioInput::EnumerateDevices(std::vector<AudioInputDevice>& devs){
 	AudioInputPulse::EnumerateDevices(devs)
 #endif
 #endif
+#endif
 }
 
 std::string AudioInput::GetCurrentDevice(){

libtgvoip/audio/AudioOutput.cpp

@@ -12,6 +12,7 @@
 #include "config.h"
 #endif
 
+#if !defined(TGVOIP_USE_SOFTWARE_AUDIO)
 #if defined(TGVOIP_USE_CALLBACK_AUDIO_IO)
 // nothing
 #elif defined(__ANDROID__)
@@ -40,6 +41,7 @@
 #else
 #error "Unsupported operating system"
 #endif
+#endif
 
 using namespace tgvoip;
 using namespace tgvoip::audio;
@@ -71,6 +73,7 @@ int32_t AudioOutput::GetEstimatedDelay(){
 
 
 void AudioOutput::EnumerateDevices(std::vector<AudioOutputDevice>& devs){
+#if !defined(TGVOIP_USE_SOFTWARE_AUDIO)
 #if defined(TGVOIP_USE_CALLBACK_AUDIO_IO)
 	// not supported
 #elif defined(__APPLE__) && TARGET_OS_OSX
@@ -93,6 +96,7 @@ void AudioOutput::EnumerateDevices(std::vector<AudioOutputDevice>& devs){
 	AudioOutputPulse::EnumerateDevices(devs)
 #endif
 #endif
+#endif
 }
 
 

libtgvoip/audio/SoftwareAudioInput.cpp

@@ -0,0 +1,66 @@
+#include <cassert>
+#include "SoftwareAudioInput.h"
+
+using namespace tgvoip::audio;
+
+SoftwareAudioInput::SoftwareAudioInput() {
+    isActive = false;
+
+    pj_pool = pjsua_pool_create("input%p", 2048, 512);
+    media_port = PJ_POOL_ZALLOC_T(pj_pool, pjmedia_port);
+
+    pj_status_t status;
+    pj_str_t name = pj_str((char *) "input");
+
+    status = pjmedia_port_info_init(&media_port->info,
+                                    &name,
+                                    PJMEDIA_SIG_CLASS_PORT_AUD('S', 'I'), // Software Input SIG
+                                    48000, 1, 16, 960);
+    assert(status == PJ_SUCCESS);
+
+    media_port->port_data.pdata = this;
+    media_port->put_frame = &PutFrameCallback;
+    media_port->get_frame = &GetFrameCallback;
+
+    registerMediaPort(media_port);
+}
+
+SoftwareAudioInput::~SoftwareAudioInput() {
+    unregisterMediaPort();
+    pjmedia_port_destroy(media_port);
+    pj_pool_release(pj_pool);
+}
+
+void SoftwareAudioInput::Start() {
+    isActive = true;
+}
+
+void SoftwareAudioInput::Stop() {
+    isActive = false;
+}
+
+pj_status_t SoftwareAudioInput::PutFrameCallback(pjmedia_port *port, pjmedia_frame *frame) {
+
+    // skip heartbeat frame
+    if (frame->type != PJMEDIA_FRAME_TYPE_AUDIO) {
+        return PJ_SUCCESS;
+    }
+
+    assert(frame->size == 960 * 2);
+
+    auto input = (SoftwareAudioInput *) port->port_data.pdata;
+
+    if (!input->isActive) {
+        return PJ_SUCCESS;
+    }
+
+    input->InvokeCallback((unsigned char *) frame->buf, frame->size);
+
+    return PJ_SUCCESS;
+}
+
+pj_status_t SoftwareAudioInput::GetFrameCallback(pjmedia_port *port, pjmedia_frame *frame) {
+    frame->size = 0;
+    frame->type = PJMEDIA_FRAME_TYPE_NONE;
+    return PJ_SUCCESS;
+}

libtgvoip/audio/SoftwareAudioInput.h

@@ -0,0 +1,33 @@
+#ifndef TG2SIP_SOFTWAREAUDIOINPUT_H
+#define TG2SIP_SOFTWAREAUDIOINPUT_H
+
+#include <pjsua2.hpp>
+#include "AudioInput.h"
+#include "../threading.h"
+
+namespace tgvoip {
+    namespace audio {
+        class SoftwareAudioInput : public AudioInput, public pj::AudioMedia {
+        public:
+            explicit SoftwareAudioInput();
+
+            virtual ~SoftwareAudioInput();
+
+            void Start() override;
+
+            void Stop() override;
+
+        private:
+            static pj_status_t PutFrameCallback(pjmedia_port *port, pjmedia_frame *frame);
+
+            static pj_status_t GetFrameCallback(pjmedia_port *port, pjmedia_frame *frame);
+
+            bool isActive;
+
+            pj_pool_t *pj_pool;
+            pjmedia_port *media_port;
+        };
+    }
+}
+
+#endif //TG2SIP_SOFTWAREAUDIOINPUT_H

libtgvoip/audio/SoftwareAudioOutput.cpp

@@ -0,0 +1,81 @@
+#include <cassert>
+#include "SoftwareAudioOutput.h"
+
+using namespace tgvoip::audio;
+
+SoftwareAudioOutput::SoftwareAudioOutput() {
+    isActive = false;
+
+    pj_pool = pjsua_pool_create("output%p", 2048, 512);
+    media_port = PJ_POOL_ZALLOC_T(pj_pool, pjmedia_port);
+
+    pj_status_t status;
+    pj_str_t name = pj_str((char *) "output");
+
+    // ptime = 10ms -> 480 samples per frame
+    status = pjmedia_port_info_init(&media_port->info,
+                                    &name,
+                                    PJMEDIA_SIG_CLASS_PORT_AUD('S', 'O'), // Software Output SIG
+                                    48000, 1, 16, 480);
+    assert(status == PJ_SUCCESS);
+
+    media_port->port_data.pdata = this;
+    media_port->put_frame = &PutFrameCallback;
+    media_port->get_frame = &GetFrameCallback;
+
+    registerMediaPort(media_port);
+
+    // 48kHz * 20 ms * 16 bits per sample
+    buffer = new unsigned char[1920];
+}
+
+SoftwareAudioOutput::~SoftwareAudioOutput() {
+    unregisterMediaPort();
+    pjmedia_port_destroy(media_port);
+    pj_pool_release(pj_pool);
+    if (buffer) {
+        delete buffer;
+    }
+}
+
+void SoftwareAudioOutput::Start() {
+    isActive = true;
+}
+
+void SoftwareAudioOutput::Stop() {
+    isActive = false;
+}
+
+bool SoftwareAudioOutput::IsPlaying() {
+    return isActive;
+}
+
+pj_status_t SoftwareAudioOutput::PutFrameCallback(pjmedia_port *port, pjmedia_frame *frame) {
+
+    frame->size = 0;
+    frame->type = PJMEDIA_FRAME_TYPE_NONE;
+    return PJ_SUCCESS;
+}
+
+pj_status_t SoftwareAudioOutput::GetFrameCallback(pjmedia_port *port, pjmedia_frame *frame) {
+
+    auto output = (SoftwareAudioOutput *) port->port_data.pdata;
+
+    if (!output->isActive) {
+        frame->type = PJMEDIA_FRAME_TYPE_NONE;
+        frame->size = 0;
+        return PJ_SUCCESS;
+    }
+
+    // libtgvoip ptime = 20ms
+    if (output->read_buffer) {
+        output->InvokeCallback(output->buffer, 1920);
+        memcpy(frame->buf, output->buffer, 960);
+    } else {
+        memcpy(frame->buf, output->buffer + 960, 960);
+    }
+
+    output->read_buffer = !output->read_buffer;
+
+    return PJ_SUCCESS;
+}

libtgvoip/audio/SoftwareAudioOutput.h

@@ -0,0 +1,38 @@
+#ifndef TG2SIP_SOFTWAREAUDIOOUTPUT_H
+#define TG2SIP_SOFTWAREAUDIOOUTPUT_H
+
+#include <pjsua2.hpp>
+#include "AudioOutput.h"
+#include "../threading.h"
+
+namespace tgvoip {
+    namespace audio {
+        class SoftwareAudioOutput : public AudioOutput, public pj::AudioMedia {
+        public:
+            explicit SoftwareAudioOutput();
+
+            virtual ~SoftwareAudioOutput();
+
+            void Start() override;
+
+            void Stop() override;
+
+            virtual bool IsPlaying();
+
+        private:
+            static pj_status_t PutFrameCallback(pjmedia_port *port, pjmedia_frame *frame);
+
+            static pj_status_t GetFrameCallback(pjmedia_port *port, pjmedia_frame *frame);
+
+            bool isActive;
+
+            pj_pool_t *pj_pool;
+            pjmedia_port *media_port;
+
+            bool read_buffer{true};
+            unsigned char *buffer;
+        };
+    }
+}
+
+#endif //TG2SIP_SOFTWAREAUDIOOUTPUT_H

libtgvoip/logging.cpp

@@ -22,7 +22,48 @@
 #include "os/darwin/DarwinSpecific.h"
 #endif
 
+#ifdef TGVOIP_USE_SPDLOG
+#include <memory>
+#include "spdlog/spdlog.h"
+#endif
+
 FILE* tgvoipLogFile=NULL;
+#ifdef TGVOIP_USE_SPDLOG
+std::shared_ptr<spdlog::logger> _voip_logger;
+
+void tgvoip_log_spdlog(char level, const char* msg, ...) {
+    if (_voip_logger) {
+        spdlog::level::level_enum lvl;
+        switch (level) {
+            case 'V':
+                lvl = spdlog::level::level_enum::trace;
+                break;
+            case 'D':
+                lvl = spdlog::level::level_enum::debug;
+                break;
+            case 'I':
+                lvl = spdlog::level::level_enum::info;
+                break;
+            case 'W':
+                lvl = spdlog::level::level_enum::warn;
+                break;
+            case 'E':
+                lvl = spdlog::level::level_enum::err;
+                break;
+            default:
+                return;
+        }
+
+        va_list argptr;
+        va_start(argptr, msg);
+
+        char buf[1024];
+        vsprintf(buf, msg, argptr);
+		_voip_logger->log(lvl, buf);
+    }
+}
+
+#endif
 
 void tgvoip_log_file_printf(char level, const char* msg, ...){
 	if(tgvoipLogFile){

libtgvoip/logging.h

@@ -13,11 +13,19 @@
 #include <TargetConditionals.h>
 #endif
 
+#ifdef TGVOIP_USE_SPDLOG
+#include "spdlog/spdlog.h"
+#endif
+
 #include <stdio.h>
 
 void tgvoip_log_file_printf(char level, const char* msg, ...);
 void tgvoip_log_file_write_header(FILE* file);
 
+#ifdef TGVOIP_USE_SPDLOG
+void tgvoip_log_spdlog(char level, const char* msg, ...);
+#endif
+
 #if defined(__ANDROID__)
 
 #include <android/log.h>
@@ -58,7 +66,11 @@ void tgvoip_log_file_write_header(FILE* file);
 
 #include <stdio.h>
 
+#ifdef TGVOIP_USE_SPDLOG
+#define _TGVOIP_LOG_PRINT(verb, msg, ...) {tgvoip_log_spdlog(verb, msg, ##__VA_ARGS__);}
+#else
 #define _TGVOIP_LOG_PRINT(verb, msg, ...) {printf("%c/tgvoip: " msg "\n", verb, ##__VA_ARGS__); tgvoip_log_file_printf(verb, msg, ##__VA_ARGS__);}
+#endif
 
 #define LOGV(msg, ...) _TGVOIP_LOG_PRINT('V', msg, ##__VA_ARGS__)
 #define LOGD(msg, ...) _TGVOIP_LOG_PRINT('D', msg, ##__VA_ARGS__)

libtgvoip/os/posix/NetworkSocketPosix.cpp

@@ -26,6 +26,11 @@ extern jclass jniUtilitiesClass;
 #include <ifaddrs.h>
 #endif
 
+// fix of undef in pjsip
+#if defined(TGVOIP_USE_SOFTWARE_AUDIO)
+#define s6_addr			__in6_u.__u6_addr8
+#endif
+
 using namespace tgvoip;