public
/
rtphone
2
0
Fork 0
Code Issues Pull Requests Projects Releases Wiki Activity

- more changes to allow CI runs on this project

This commit is contained in:
dmytro.bogovych 2019-06-09 22:00:41 +03:00
parent 228e2d7829
commit 7d4933b066
10 changed files with 328 additions and 250 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

58
src/CMakeLists.txt
View File

@ -1,5 +1,7 @@
project(rtphone)
cmake_minimum_required(VERSION 3.0)
macro(configure_msvc_runtime)
if(MSVC)
# Default to statically-linked runtime.
@ -59,11 +61,25 @@ set (USE_AMR_CODEC OFF CACHE BOOL "Use AMR codec. Requires libraries.")
set (USE_EVS_CODEC OFF CACHE BOOL "Use EVS codec.")
set (OPENSSL_SSL ssl CACHE STRING "Pointer to ssl library")
set (OPENSSL_CRYPTO crypto CACHE STRING "Pointer to crypto library")
set (OPENSSL_INCLUDE "/usr/local/include/openssl" CACHE STRING "Pointer to OpenSSL include files")
set(CMAKE_POSITION_INDEPENDENT_CODE ON)
message ("Using ssl library at ${OPENSSL_SSL}")
message ("Using crypto library at ${OPENSSL_CRYPTO}")
message ("Using OpenSSL include files from ${OPENSSL_INCLUDE}")
if (CMAKE_SYSTEM MATCHES "Windows*")
add_definitions(-DTARGET_WIN)
endif()
if (CMAKE_SYSTEM MATCHES "Linux*")
add_definitions(-DTARGET_LINUX)
endif()
if (CMAKE_SYSTEM MATCHES "Darwin*")
add_definitions(-DTARGET_OSX)
endif()
set (RTPHONE_SOURCES
${rtphone_engine}/media/MT_Statistics.cpp
@ -94,10 +110,39 @@ set (RTPHONE_SOURCES
${rtphone_engine}/endpoint/EP_Session.cpp
)
add_library(rtphone STATIC ${RTPHONE_SOURCES})
set (RTPHONE_HEADERS
${rtphone_engine}/media/MT_Statistics.h
${rtphone_engine}/media/MT_WebRtc.h
${rtphone_engine}/media/MT_Stream.h
${rtphone_engine}/media/MT_SrtpHelper.h
${rtphone_engine}/media/MT_SingleAudioStream.h
${rtphone_engine}/media/MT_NativeRtpSender.h
${rtphone_engine}/media/MT_Dtmf.h
${rtphone_engine}/media/MT_CodecList.h
${rtphone_engine}/media/MT_Codec.h
${rtphone_engine}/media/MT_Box.h
${rtphone_engine}/media/MT_AudioStream.h
${rtphone_engine}/media/MT_AudioReceiver.h
${rtphone_engine}/media/MT_AudioCodec.h
${rtphone_engine}/media/MT_SevanaMos.h
${rtphone_engine}/media/MT_AmrCodec.h
${rtphone_engine}/media/MT_EvsCodec.h
${rtphone_engine}/media/MT_CngHelper.h
${rtphone_engine}/agent/Agent_Impl.h
${rtphone_engine}/agent/Agent_AudioManager.h
${rtphone_engine}/endpoint/EP_Account.h
${rtphone_engine}/endpoint/EP_AudioProvider.h
${rtphone_engine}/endpoint/EP_DataProvider.h
${rtphone_engine}/endpoint/EP_Engine.h
${rtphone_engine}/endpoint/EP_NetworkQueue.h
${rtphone_engine}/endpoint/EP_Observer.h
${rtphone_engine}/endpoint/EP_Session.h
)
add_library(rtphone STATIC ${RTPHONE_SOURCES} ${RTPHONE_HEADERS})
add_subdirectory(${rtphone_engine}/helper)
add_subdirectory(${rtphone_engine}/audio)
add_subdirectory(${rtphone_engine}/media)
add_subdirectory(${rtphone_libs}/resiprocate)
add_subdirectory(${rtphone_libs}/ice)
add_subdirectory(${rtphone_libs}/jrtplib/src)
@ -116,9 +161,9 @@ set(LIBS ice_stack jrtplib g729_codec gsm_codec
if (CMAKE_SYSTEM MATCHES "Win*")
set (LIBS ${LIBS} opus )
else (CMAKE_SYSTEM MATCHES "Win*")
else ()
set (LIBS ${LIBS} dl opus uuid)
endif (CMAKE_SYSTEM MATCHES "Win*")
endif ()
if (USE_AMR_CODEC)
set(LIBS ${LIBS} opencore-amrnb opencore-amrwb)
@ -134,6 +179,11 @@ target_link_libraries(rtphone
target_include_directories(rtphone
PUBLIC $<BUILD_INTERFACE:${CMAKE_CURRENT_SOURCE_DIR}>
$<BUILD_INTERFACE:${CMAKE_CURRENT_SOURCE_DIR}/engine>
PRIVATE ../../libs/speex/include ../../libs ../)
PRIVATE ${CMAKE_CURRENT_SOURCE_DIR}/libs/
${CMAKE_CURRENT_SOURCE_DIR}/libs/speex/include
${CMAKE_CURRENT_SOURCE_DIR}/libs/opus/include/
${CMAKE_CURRENT_SOURCE_DIR}/libs/json
)
configure_msvc_runtime()

472
src/engine/endpoint/EP_AudioProvider.cpp
View File

@ -17,16 +17,16 @@
#define LOG_SUBSYSTEM "AudioProvider"
AudioProvider::AudioProvider(UserAgent& agent, MT::Terminal& terminal)
:mUserAgent(agent), mTerminal(terminal), mState(0),
mRemoteTelephoneCodec(0), mRemoteNoSdp(false)
:mUserAgent(agent), mTerminal(terminal), mState(0),
mRemoteTelephoneCodec(0), mRemoteNoSdp(false)
{
mActive = mfActive;
mRemoteState = msSendRecv;
mActiveStream = mTerminal.createStream(MT::Stream::Audio, mUserAgent.config());
if (mUserAgent.config().exists(CONFIG_CODEC_PRIORITY))
mCodecPriority.setupFrom(mUserAgent.config()[CONFIG_CODEC_PRIORITY].asVMap());
mSrtpSuite = SRTP_NONE;
setState((int)StreamState::SipRecv | (int)StreamState::SipSend | (int)StreamState::Receiving | (int)StreamState::Sending);
mActive = mfActive;
mRemoteState = msSendRecv;
mActiveStream = mTerminal.createStream(MT::Stream::Audio, mUserAgent.config());
if (mUserAgent.config().exists(CONFIG_CODEC_PRIORITY))
mCodecPriority.setupFrom(mUserAgent.config()[CONFIG_CODEC_PRIORITY].asVMap());
mSrtpSuite = SRTP_NONE;
setState((int)StreamState::SipRecv | (int)StreamState::SipSend | (int)StreamState::Receiving | (int)StreamState::Sending);
}
AudioProvider::~AudioProvider()
@ -35,346 +35,346 @@ AudioProvider::~AudioProvider()
std::string AudioProvider::streamName()
{
return "audio";
return "audio";
}
std::string AudioProvider::streamProfile()
{
if (mState & (int)StreamState::Srtp)
return "RTP/SAVP";
else
return "RTP/AVP";
if (mState & (int)StreamState::Srtp)
return "RTP/SAVP";
else
return "RTP/AVP";
}
// Sets destination IP address
void AudioProvider::setDestinationAddress(const RtpPair<InternetAddress>& addr)
{
if (!mActiveStream)
return;
if (!mActiveStream)
return;
mActiveStream->setDestination(addr);
mActiveStream->setDestination(addr);
}
void AudioProvider::configureMediaObserver(MT::Stream::MediaObserver *observer, void* userTag)
{
mMediaObserver = observer;
mMediaObserverTag = userTag;
if (mActiveStream)
mActiveStream->configureMediaObserver(observer, userTag);
mMediaObserver = observer;
mMediaObserverTag = userTag;
if (mActiveStream)
mActiveStream->configureMediaObserver(observer, userTag);
}
// Processes incoming data
void AudioProvider::processData(PDatagramSocket s, const void* dataBuffer, int dataSize, InternetAddress& source)
{
if (!mActiveStream)
return;
if (!mActiveStream)
return;
if (RtpHelper::isRtpOrRtcp(dataBuffer, dataSize))
{
ICELogMedia(<<"Adding new data to stream processing");
mActiveStream->dataArrived(s, dataBuffer, dataSize, source);
}
if (RtpHelper::isRtpOrRtcp(dataBuffer, dataSize))
{
ICELogMedia(<<"Adding new data to stream processing");
mActiveStream->dataArrived(s, dataBuffer, dataSize, source);
}
}
// This method is called by user agent to send ICE packet from mediasocket
void AudioProvider::sendData(PDatagramSocket s, InternetAddress& destination, const void* buffer, unsigned int size)
{
s->sendDatagram(destination, buffer, size);
s->sendDatagram(destination, buffer, size);
}
// Create SDP offer
void AudioProvider::updateSdpOffer(resip::SdpContents::Session::Medium& sdp, SdpDirection direction)
{
if (mRemoteNoSdp)
return;
if (mRemoteNoSdp)
return;
if (mState & (int)StreamState::Srtp)
{
// Check if SRTP suite is found already or not
if (mSrtpSuite == SRTP_NONE)
if (mState & (int)StreamState::Srtp)
{
for (int suite = SRTP_AES_128_AUTH_80; suite <= SRTP_LAST; suite++)
sdp.addAttribute("crypto", resip::Data(createCryptoAttribute((SrtpSuite)suite)));
// Check if SRTP suite is found already or not
if (mSrtpSuite == SRTP_NONE)
{
for (int suite = SRTP_AES_128_AUTH_80; suite <= SRTP_LAST; suite++)
sdp.addAttribute("crypto", resip::Data(createCryptoAttribute((SrtpSuite)suite)));
}
else
sdp.addAttribute("crypto", resip::Data(createCryptoAttribute(mSrtpSuite)));
}
// Use CodecListPriority mCodecPriority adapter to work with codec priorities
if (mAvailableCodecs.empty())
{
for (int i=0; i<mCodecPriority.count(mTerminal.codeclist()); i++)
mCodecPriority.codecAt(mTerminal.codeclist(), i).updateSdp(sdp.codecs(), direction);
sdp.addCodec(resip::SdpContents::Session::Codec::TelephoneEvent);
}
else
sdp.addAttribute("crypto", resip::Data(createCryptoAttribute(mSrtpSuite)));
}
// Use CodecListPriority mCodecPriority adapter to work with codec priorities
if (mAvailableCodecs.empty())
{
for (int i=0; i<mCodecPriority.count(mTerminal.codeclist()); i++)
mCodecPriority.codecAt(mTerminal.codeclist(), i).updateSdp(sdp.codecs(), direction);
sdp.addCodec(resip::SdpContents::Session::Codec::TelephoneEvent);
}
else
{
mAvailableCodecs.front().mFactory->updateSdp(sdp.codecs(), direction);
if (mRemoteTelephoneCodec)
sdp.addCodec(resip::SdpContents::Session::Codec::TelephoneEvent);
}
// Publish stream state
const char* attr = nullptr;
switch (mActive)
{
case mfActive:
switch(mRemoteState)
{
case msSendonly: attr = "recvonly"; break;
case msInactive: attr = "recvonly"; break;
mAvailableCodecs.front().mFactory->updateSdp(sdp.codecs(), direction);
if (mRemoteTelephoneCodec)
sdp.addCodec(resip::SdpContents::Session::Codec::TelephoneEvent);
}
break;
case mfPaused:
switch (mRemoteState)
// Publish stream state
const char* attr = nullptr;
switch (mActive)
{
case msRecvonly: attr = "sendonly"; break;
case msSendonly: attr = "inactive"; break;
case msInactive: attr = "inactive"; break;
case msSendRecv: attr = "sendonly"; break;
case mfActive:
switch(mRemoteState)
{
case msSendonly: attr = "recvonly"; break;
case msInactive: attr = "recvonly"; break;
}
break;
case mfPaused:
switch (mRemoteState)
{
case msRecvonly: attr = "sendonly"; break;
case msSendonly: attr = "inactive"; break;
case msInactive: attr = "inactive"; break;
case msSendRecv: attr = "sendonly"; break;
}
break;
}
break;
}
if (attr)
sdp.addAttribute(attr);
if (attr)
sdp.addAttribute(attr);
}
void AudioProvider::sessionDeleted()
{
sessionTerminated();
sessionTerminated();
}
void AudioProvider::sessionTerminated()
{
ICELogDebug(<< "sessionTerminated() for audio provider");
setState(state() & ~((int)StreamState::Sending | (int)StreamState::Receiving));
ICELogDebug(<< "sessionTerminated() for audio provider");
setState(state() & ~((int)StreamState::Sending | (int)StreamState::Receiving));
if (mActiveStream)
{
ICELogDebug(<< "Copy statistics from existing stream before freeing.");
if (mActiveStream)
{
ICELogDebug(<< "Copy statistics from existing stream before freeing.");
// Copy statistics - maybe it will be requested later
mBackupStats = mActiveStream->statistics();
// Copy statistics - maybe it will be requested later
mBackupStats = mActiveStream->statistics();
ICELogDebug(<< "Remove stream from terminal");
mTerminal.freeStream(mActiveStream);
ICELogDebug(<< "Remove stream from terminal");
mTerminal.freeStream(mActiveStream);
// Retrieve final statistics
MT::AudioStream* audio_stream = dynamic_cast<MT::AudioStream*>(mActiveStream.get());
if (audio_stream)
audio_stream->setFinalStatisticsOutput(&mBackupStats);
// Retrieve final statistics
MT::AudioStream* audio_stream = dynamic_cast<MT::AudioStream*>(mActiveStream.get());
if (audio_stream)
audio_stream->setFinalStatisticsOutput(&mBackupStats);
ICELogDebug(<< "Reset reference to stream.");
mActiveStream.reset();
}
ICELogDebug(<< "Reset reference to stream.");
mActiveStream.reset();
}
}
void AudioProvider::sessionEstablished(int conntype)
{
// Start media streams
setState(state() | (int)StreamState::Receiving | (int)StreamState::Sending);
// Start media streams
setState(state() | (int)StreamState::Receiving | (int)StreamState::Sending);
// Available codec list can be empty in case of no-sdp offers.
if (conntype == EV_SIP && !mAvailableCodecs.empty() && mActiveStream)
{
RemoteCodec& rc = mAvailableCodecs.front();
mActiveStream->setTransmittingCodec(*rc.mFactory, rc.mRemotePayloadType);
dynamic_cast<MT::AudioStream*>(mActiveStream.get())->setTelephoneCodec(mRemoteTelephoneCodec);
}
// Available codec list can be empty in case of no-sdp offers.
if (conntype == EV_SIP && !mAvailableCodecs.empty() && mActiveStream)
{
RemoteCodec& rc = mAvailableCodecs.front();
mActiveStream->setTransmittingCodec(*rc.mFactory, rc.mRemotePayloadType);
dynamic_cast<MT::AudioStream*>(mActiveStream.get())->setTelephoneCodec(mRemoteTelephoneCodec);
}
}
void AudioProvider::setSocket(const RtpPair<PDatagramSocket>& p4, const RtpPair<PDatagramSocket>& p6)
{
mSocket4 = p4;
mSocket6 = p6;
mActiveStream->setSocket(p4);
mSocket4 = p4;
mSocket6 = p6;
mActiveStream->setSocket(p4);
}
RtpPair<PDatagramSocket>& AudioProvider::socket(int family)
{
switch (family)
{
case AF_INET:
return mSocket4;
switch (family)
{
case AF_INET:
return mSocket4;
case AF_INET6:
return mSocket6;
}
return mSocket4;
case AF_INET6:
return mSocket6;
}
return mSocket4;
}
bool AudioProvider::processSdpOffer(const resip::SdpContents::Session::Medium& media, SdpDirection sdpDirection)
{
// Check if there is compatible codec
mAvailableCodecs.clear();
mRemoteTelephoneCodec = 0;
// Check if there is SDP at all
mRemoteNoSdp = media.codecs().empty();
if (mRemoteNoSdp)
// Check if there is compatible codec
mAvailableCodecs.clear();
mRemoteTelephoneCodec = 0;
// Check if there is SDP at all
mRemoteNoSdp = media.codecs().empty();
if (mRemoteNoSdp)
return true;
// Update RFC2833 related information
findRfc2833(media.codecs());
// Use CodecListPriority mCodecPriority to work with codec priorities
int pt;
for (int localIndex=0; localIndex<mCodecPriority.count(mTerminal.codeclist()); localIndex++)
{
MT::Codec::Factory& factory = mCodecPriority.codecAt(mTerminal.codeclist(), localIndex);
if ((pt = factory.processSdp(media.codecs(), sdpDirection)) != -1)
mAvailableCodecs.push_back(RemoteCodec(&factory, pt));
}
if (!mAvailableCodecs.size())
return false;
// Iterate SRTP crypto: attributes
if (media.exists("crypto"))
{
// Find the most strong crypt suite
const std::list<resip::Data>& vl = media.getValues("crypto");
SrtpSuite ss = SRTP_NONE;
ByteBuffer key;
for (std::list<resip::Data>::const_iterator attrIter = vl.begin(); attrIter != vl.end(); attrIter++)
{
const resip::Data& attr = *attrIter;
ByteBuffer tempkey;
SrtpSuite suite = processCryptoAttribute(attr, tempkey);
if (suite > ss)
{
ss = suite;
mSrtpSuite = suite;
key = tempkey;
}
}
// If SRTP suite is agreed
if (ss != SRTP_NONE)
{
ICELogInfo(<< "Found SRTP suite " << ss);
mActiveStream->srtp().open(key, ss);
setState(state() | (int)StreamState::Srtp);
}
else
ICELogInfo(<< "Did not find valid SRTP suite");
}
DataProvider::processSdpOffer(media, sdpDirection);
return true;
// Update RFC2833 related information
findRfc2833(media.codecs());
// Use CodecListPriority mCodecPriority to work with codec priorities
int pt;
for (int localIndex=0; localIndex<mCodecPriority.count(mTerminal.codeclist()); localIndex++)
{
MT::Codec::Factory& factory = mCodecPriority.codecAt(mTerminal.codeclist(), localIndex);
if ((pt = factory.processSdp(media.codecs(), sdpDirection)) != -1)
mAvailableCodecs.push_back(RemoteCodec(&factory, pt));
}
if (!mAvailableCodecs.size())
return false;
// Iterate SRTP crypto: attributes
if (media.exists("crypto"))
{
// Find the most strong crypt suite
const std::list<resip::Data>& vl = media.getValues("crypto");
SrtpSuite ss = SRTP_NONE;
ByteBuffer key;
for (std::list<resip::Data>::const_iterator attrIter = vl.begin(); attrIter != vl.end(); attrIter++)
{
const resip::Data& attr = *attrIter;
ByteBuffer tempkey;
SrtpSuite suite = processCryptoAttribute(attr, tempkey);
if (suite > ss)
{
ss = suite;
mSrtpSuite = suite;
key = tempkey;
}
}
// If SRTP suite is agreed
if (ss != SRTP_NONE)
{
ICELogInfo(<< "Found SRTP suite " << ss);
mActiveStream->srtp().open(key, ss);
setState(state() | (int)StreamState::Srtp);
}
else
ICELogInfo(<< "Did not find valid SRTP suite");
}
DataProvider::processSdpOffer(media, sdpDirection);
return true;
}
void AudioProvider::setState(unsigned state)
{
mState = state;
if (mActiveStream)
mActiveStream->setState(state);
mState = state;
if (mActiveStream)
mActiveStream->setState(state);
}
unsigned AudioProvider::state()
{
return mState;
return mState;
}
MT::Statistics AudioProvider::getStatistics()
{
if (mActiveStream)
return mActiveStream->statistics();
else
return mBackupStats;
if (mActiveStream)
return mActiveStream->statistics();
else
return mBackupStats;
}
MT::PStream AudioProvider::activeStream()
{
return mActiveStream;
return mActiveStream;
}
std::string AudioProvider::createCryptoAttribute(SrtpSuite suite)
{
if (!mActiveStream)
return "";
if (!mActiveStream)
return "";
// Use tag 1 - it is ok, as we use only single crypto attribute
int srtpTag = 1;
// Print key to base64 string
PByteBuffer keyBuffer = mActiveStream->srtp().outgoingKey(suite).first;
resip::Data d(keyBuffer->data(), keyBuffer->size());
resip::Data keyText = d.base64encode();
// Use tag 1 - it is ok, as we use only single crypto attribute
int srtpTag = 1;
// Create "crypto" attribute value
char buffer[512];
const char* suiteName = NULL;
switch (suite)
{
case SRTP_AES_128_AUTH_80: suiteName = SRTP_SUITE_NAME_1; break;
case SRTP_AES_256_AUTH_80: suiteName = SRTP_SUITE_NAME_2; break;
default: assert(0);
}
sprintf(buffer, "%d %s inline:%s", srtpTag, suiteName, keyText.c_str());
// Print key to base64 string
PByteBuffer keyBuffer = mActiveStream->srtp().outgoingKey(suite).first;
resip::Data d(keyBuffer->data(), keyBuffer->size());
resip::Data keyText = d.base64encode();
return buffer;
// Create "crypto" attribute value
char buffer[512];
const char* suiteName = NULL;
switch (suite)
{
case SRTP_AES_128_AUTH_80: suiteName = SRTP_SUITE_NAME_1; break;
case SRTP_AES_256_AUTH_80: suiteName = SRTP_SUITE_NAME_2; break;
default: assert(0);
}
sprintf(buffer, "%d %s inline:%s", srtpTag, suiteName, keyText.c_str());
return buffer;
}
SrtpSuite AudioProvider::processCryptoAttribute(const resip::Data& value, ByteBuffer& key)
{
int srtpTag = 0;
char suite[64], keyChunk[256];
int components = sscanf(value.c_str(), "%d %63s inline: %255s", &srtpTag, suite, keyChunk);
if (components != 3)
return SRTP_NONE;
int srtpTag = 0;
char suite[64], keyChunk[256];
int components = sscanf(value.c_str(), "%d %63s inline: %255s", &srtpTag, suite, keyChunk);
if (components != 3)
return SRTP_NONE;
const char* delimiter = strchr(keyChunk, '|');
resip::Data keyText;
if (delimiter)
keyText = resip::Data(keyChunk, delimiter - keyChunk);
else
keyText = resip::Data(keyChunk);
const char* delimiter = strchr(keyChunk, '|');
resip::Data keyText;
if (delimiter)
keyText = resip::Data(keyChunk, delimiter - keyChunk);
else
keyText = resip::Data(keyChunk);
resip::Data rawkey = keyText.base64decode();
key = ByteBuffer(rawkey.c_str(), rawkey.size());
// Open srtp
SrtpSuite result = SRTP_NONE;
if (strcmp(suite, SRTP_SUITE_NAME_1) == 0)
result = SRTP_AES_128_AUTH_80;
else
if (strcmp(suite, SRTP_SUITE_NAME_2) == 0)
result = SRTP_AES_256_AUTH_80;
resip::Data rawkey = keyText.base64decode();
key = ByteBuffer(rawkey.c_str(), rawkey.size());
return result;
// Open srtp
SrtpSuite result = SRTP_NONE;
if (strcmp(suite, SRTP_SUITE_NAME_1) == 0)
result = SRTP_AES_128_AUTH_80;
else
if (strcmp(suite, SRTP_SUITE_NAME_2) == 0)
result = SRTP_AES_256_AUTH_80;
return result;
}
void AudioProvider::findRfc2833(const resip::SdpContents::Session::Medium::CodecContainer& codecs)
{
resip::SdpContents::Session::Medium::CodecContainer::const_iterator codecIter;
for (codecIter = codecs.begin(); codecIter != codecs.end(); codecIter++)
{
if (strcmp("TELEPHONE-EVENT", codecIter->getName().c_str()) == 0 ||
strcmp("telephone-event", codecIter->getName().c_str()) == 0)
mRemoteTelephoneCodec = codecIter->payloadType();
}
resip::SdpContents::Session::Medium::CodecContainer::const_iterator codecIter;
for (codecIter = codecs.begin(); codecIter != codecs.end(); codecIter++)
{
if (strcmp("TELEPHONE-EVENT", codecIter->getName().c_str()) == 0 ||
strcmp("telephone-event", codecIter->getName().c_str()) == 0)
mRemoteTelephoneCodec = codecIter->payloadType();
}
}
void AudioProvider::readFile(const Audio::PWavFileReader& stream, MT::Stream::MediaDirection direction)
{
// Iterate stream list
if (mActiveStream)
mActiveStream->readFile(stream, direction);
// Iterate stream list
if (mActiveStream)
mActiveStream->readFile(stream, direction);
}
void AudioProvider::writeFile(const Audio::PWavFileWriter& stream, MT::Stream::MediaDirection direction)
{
if (mActiveStream)
mActiveStream->writeFile(stream, direction);
if (mActiveStream)
mActiveStream->writeFile(stream, direction);
}
void AudioProvider::setupMirror(bool enable)
{
if (mActiveStream)
mActiveStream->setupMirror(enable);
if (mActiveStream)
mActiveStream->setupMirror(enable);
}

20
src/engine/media/CMakeLists.txt
View File

@ -13,24 +13,28 @@ add_definitions(-DHAVE_STDINT_H -DHAVE_UINT64_T)
if(CMAKE_SYSTEM MATCHES "Linux*")
add_definitions(-DHAVE_NETINET_IN_H)
endif(CMAKE_SYSTEM MATCHES "Linux*")
endif()
if(CMAKE_SYSTEM MATCHES "Darwin*")
# OS X Specific flags
add_definitions(-DHAVE_NETINET_IN_H)
endif(CMAKE_SYSTEM MATCHES "Darwin*")
endif()
if (CMAKE_SYSTEM MATCHES "Win*")
if (CMAKE_SYSTEM MATCHES "Windows*")
# Windows Specific flags - MSVC expected
add_definitions(-D_CRT_SECURE_NO_WARNINGS -DHAVE_WINSOCK2_H
-D_SILENCE_STDEXT_HASH_DEPRECATION_WARNINGS -DUNICODE -D_UNICODE )
endif(CMAKE_SYSTEM MATCHES "Win*")
endif()
add_library(media_lib ${MEDIA_LIB_SOURCES})
target_include_directories(media_lib PRIVATE ../../libs/
../ ../../libs/srtp/include
../../libs/srtp/crypto/include
../../libs/webrtc)
target_include_directories(media_lib
PUBLIC ${CMAKE_CURRENT_SOURCE_DIR}/../../libs/
${CMAKE_CURRENT_SOURCE_DIR}/../
${CMAKE_CURRENT_SOURCE_DIR}/../../libs/srtp/include
${CMAKE_CURRENT_SOURCE_DIR}/../../libs/srtp/crypto/include
${CMAKE_CURRENT_SOURCE_DIR}/../../libs/webrtc
${CMAKE_CURRENT_SOURCE_DIR}/../../libs/opus/include/
)

2
src/libs/libgsm/CMakeLists.txt
View File

@ -24,3 +24,5 @@ set (GSM_SOURCES
)
add_library(gsm_codec ${GSM_SOURCES})
target_include_directories(gsm_codec PRIVATE ${CMAKE_CURRENT_SOURCE_DIR})
target_compile_definitions(gsm_codec PUBLIC HAS_STDLIB_H HAS_STRING_H)

2
src/libs/libgsm/gsm_code.c
View File

@ -7,7 +7,7 @@
*/
#include "config.h"
//#include "config.h"
#ifdef HAS_STDLIB_H

2
src/libs/libgsm/gsm_create.c
View File

@ -7,7 +7,7 @@
*/
#include "config.h"
//#include "config.h"
#ifdef HAS_STRING_H
#include <string.h>

7
src/libs/resiprocate/CMakeLists.txt
View File

@ -307,4 +307,11 @@ SET (RUTIL_SOURCES
add_library(resiprocate ${ARES_SOURCES} ${RUTIL_SOURCES} ${STACK_SOURCES} ${DUM_SOURCES})
target_include_directories(resiprocate PUBLIC
${CMAKE_CURRENT_SOURCE_DIR}
${CMAKE_CURRENT_SOURCE_DIR}/contrib/ares
/usr/local/include
)
target_compile_definitions(resiprocate PUBLIC -DUSE_ARES -DUSE_SSL)
#add_library(resiprocate_lite ${RUTIL_SOURCES} ${STACK_SOURCES})

5
src/libs/speexdsp/CMakeLists.txt
View File

@ -20,3 +20,8 @@ set (SPEEXDSP_SOURCES
)
add_library(speexdsp ${SPEEXDSP_SOURCES})
target_compile_definitions(speexdsp PUBLIC -DUSE_KISS_FFT -DFIXED_POINT)
target_include_directories(speexdsp PRIVATE
${CMAKE_CURRENT_SOURCE_DIR}/libspeexdsp
${CMAKE_CURRENT_SOURCE_DIR}/include)

4
src/libs/srtp/CMakeLists.txt
View File

@ -35,3 +35,7 @@ set (SRTP_SOURCES
)
add_library(srtp ${SRTP_SOURCES})
target_include_directories(srtp PUBLIC
${CMAKE_CURRENT_SOURCE_DIR}/include
${CMAKE_CURRENT_SOURCE_DIR}/crypto/include
)

6
src/libs/webrtc/CMakeLists.txt
View File

@ -173,3 +173,9 @@ set (WEBRTC_SOURCES
)
add_library(webrtc ${WEBRTC_SOURCES})
target_include_directories(webrtc PUBLIC
${CMAKE_CURRENT_SOURCE_DIR}
${CMAKE_CURRENT_SOURCE_DIR}/signal_processing_library
${CMAKE_CURRENT_SOURCE_DIR}/utility
)