Dear All , Kindly I need your help to be able to utilize the HW G729 implementation . During building pjsip 1.12 Symbian_ua_gui / Target = S60 3RD FP1 ( VAS-Direct Activated ) I get the following errors I made sure that I'm activating only VAS in both my config_site and the project mmp file Note : With VAS & VAS-Direct Disabled , I get no errors ( just couple of warnings ) FY reference , both files included below Appreciate your help . undefined reference to `CMdaAudioInputStream::NewL(MMdaAudioInputStreamCallback&)' symb_mda_dev.cpp /Temsa7-03/pjmedia/src/pjmedia-audiodev line 0 C/C++ Problem undefined reference to `CMdaAudioInputStream::Open(TMdaPackage*)' symb_mda_dev.cpp /Temsa7-03/pjmedia/src/pjmedia-audiodev line 0 C/C++ Problem undefined reference to `CMdaAudioInputStream::Stop()' symb_mda_dev.cpp /Temsa7-03/pjmedia/src/pjmedia-audiodev line 0 C/C++ Problem undefined reference to `CMdaAudioInputStream::SetPriority(int, TMdaPriorityPreference)' symb_mda_dev.cpp /Temsa7-03/pjmedia/src/pjmedia-audiodev line 0 C/C++ Problem undefined reference to `CMdaAudioInputStream::ReadL(TDes8&)' symb_mda_dev.cpp /Temsa7-03/pjmedia/src/pjmedia-audiodev line 0 C/C++ Problem undefined reference to `CMdaAudioOutputStream::NewL(MMdaAudioOutputStreamCallback&, CMdaServer*)' symb_mda_dev.cpp /Temsa7-03/pjmedia/src/pjmedia-audiodev line 0 C/C++ Problem undefined reference to `CMdaAudioInputStream::Gain() const' symb_mda_dev.cpp /Temsa7-03/pjmedia/src/pjmedia-audiodev line 0 C/C++ Problem undefined reference to `CMdaAudioInputStream::MaxGain() const' symb_mda_dev.cpp /Temsa7-03/pjmedia/src/pjmedia-audiodev line 0 C/C++ Problem undefined reference to `CMdaAudioInputStream::SetGain(int)' symb_mda_dev.cpp /Temsa7-03/pjmedia/src/pjmedia-audiodev line 0 C/C++ Problem WARNING: When building for ARMV5 platform Compiler RVCT2.2 or later is required. Temsa7-03 Unknown C/C++ Problem enumeral mismatch in conditional expression: `TFalse' vs `TTrue' Temsa7-03 line 242 C/C++ Problem 'g_buddy_id' defined but not used symbian_ua.cpp /Temsa7-03/pjsip-apps/src/symbian_ua_gui/src line 37 C/C++ Problem "/*" within comment Temsa7-03 line 302 C/C++ Problem +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ /* * This file contains several sample settings especially for Windows * Mobile and Symbian targets. You can include this file in your * <pj/config_site.h> file. * * The Windows Mobile and Symbian settings will be activated * automatically if you include this file. * * In addition, you may specify one of these macros (before including * this file) to activate additional settings: * * #define PJ_CONFIG_NOKIA_APS_DIRECT * Use this macro to activate the APS-Direct feature. Please see * http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct for more * info. * * #define PJ_CONFIG_WIN32_WMME_DIRECT * Configuration to activate "APS-Direct" media mode on Windows or * Windows Mobile, useful for testing purposes only. */ /* * Typical configuration for WinCE target. */ #if defined(PJ_WIN32_WINCE) && PJ_WIN32_WINCE!=0 /* * PJLIB settings. */ /* Disable floating point support */ #define PJ_HAS_FLOATING_POINT 0 /* * PJMEDIA settings */ /* Select codecs to disable */ #define PJMEDIA_HAS_G711_CODEC 0 #define PJMEDIA_HAS_L16_CODEC 0 #define PJMEDIA_HAS_GSM_CODEC 1 #define PJMEDIA_HAS_SPEEX_CODEC 0 #define PJMEDIA_HAS_ILBC_CODEC 0 #define PJMEDIA_HAS_G722_CODEC 0 #define PJMEDIA_HAS_INTEL_IPP 0 /* We probably need more buffers on WM, so increase the limit */ #define PJMEDIA_SOUND_BUFFER_COUNT 32 /* Fine tune Speex's default settings for best performance/quality */ #define PJMEDIA_CODEC_SPEEX_DEFAULT_QUALITY 5 /* For CPU reason, disable speex AEC and use the echo suppressor. */ #define PJMEDIA_HAS_SPEEX_AEC 0 /* Previously, resampling is disabled due to performance reason and * this condition prevented some 'light' wideband codecs (e.g: G722.1) * to work along with narrowband codecs. Lately, some tests showed * that 16kHz <-> 8kHz resampling using libresample small filter was * affordable on ARM9 260 MHz, so here we decided to enable resampling. * Note that it is important to make sure that libresample is created * using small filter. For example PJSUA_DEFAULT_CODEC_QUALITY must * be set to 3 or 4 so pjsua-lib will apply small filter resampling. */ //#define PJMEDIA_RESAMPLE_IMP PJMEDIA_RESAMPLE_NONE #define PJMEDIA_RESAMPLE_IMP PJMEDIA_RESAMPLE_LIBRESAMPLE /* Use the lighter WSOLA implementation */ #define PJMEDIA_WSOLA_IMP PJMEDIA_WSOLA_IMP_WSOLA_LITE /* * PJSIP settings. */ /* Set maximum number of dialog/transaction/calls to minimum to reduce * memory usage */ #define PJSIP_MAX_TSX_COUNT 31 #define PJSIP_MAX_DIALOG_COUNT 31 #define PJSUA_MAX_CALLS 4 /* * PJSUA settings */ /* Default codec quality, previously was set to 5, however it is now * set to 4 to make sure pjsua instantiates resampler with small filter. */ #define PJSUA_DEFAULT_CODEC_QUALITY 4 /* Set maximum number of objects to minimum to reduce memory usage */ #define PJSUA_MAX_ACC 4 #define PJSUA_MAX_PLAYERS 4 #define PJSUA_MAX_RECORDERS 4 #define PJSUA_MAX_CONF_PORTS (PJSUA_MAX_CALLS+2*PJSUA_MAX_PLAYERS) #define PJSUA_MAX_BUDDIES 32 #endif /* PJ_WIN32_WINCE */ /* * Typical configuration for Symbian OS target */ #if defined(PJ_SYMBIAN) && PJ_SYMBIAN!=0 /* * PJLIB settings. */ /* Disable floating point support */ #define PJ_HAS_FLOATING_POINT 0 /* Misc PJLIB setting */ #define PJ_MAXPATH 80 /* This is important for Symbian. Symbian lacks vsnprintf(), so * if the log buffer is not long enough it's possible that * large incoming packet will corrupt memory when the log tries * to log the packet. */ #define PJ_LOG_MAX_SIZE (PJSIP_MAX_PKT_LEN+500) /* Since we don't have threads, log buffer can use static buffer * rather than stack */ #define PJ_LOG_USE_STACK_BUFFER 0 /* Disable check stack since it increases footprint */ #define PJ_OS_HAS_CHECK_STACK 0 /* * PJMEDIA settings */ /* Disable non-Symbian audio devices */ #define PJMEDIA_AUDIO_DEV_HAS_PORTAUDIO 0 #define PJMEDIA_AUDIO_DEV_HAS_WMME 0 /* Select codecs to disable */ #define PJMEDIA_HAS_G711_CODEC 0 #define PJMEDIA_HAS_L16_CODEC 0 #define PJMEDIA_HAS_GSM_CODEC 1 #define PJMEDIA_HAS_SPEEX_CODEC 0 #define PJMEDIA_HAS_ILBC_CODEC 0 #define PJMEDIA_HAS_G722_CODEC 0 #define PJMEDIA_HAS_INTEL_IPP 0 /* Fine tune Speex's default settings for best performance/quality */ #define PJMEDIA_CODEC_SPEEX_DEFAULT_QUALITY 5 /* For CPU reason, disable speex AEC and use the echo suppressor. */ #define PJMEDIA_HAS_SPEEX_AEC 0 /* Previously, resampling is disabled due to performance reason and * this condition prevented some 'light' wideband codecs (e.g: G722.1) * to work along with narrowband codecs. Lately, some tests showed * that 16kHz <-> 8kHz resampling using libresample small filter was * affordable on ARM9 222 MHz, so here we decided to enable resampling. * Note that it is important to make sure that libresample is created * using small filter. For example PJSUA_DEFAULT_CODEC_QUALITY must * be set to 3 or 4 so pjsua-lib will apply small filter resampling. */ //#define PJMEDIA_RESAMPLE_IMP PJMEDIA_RESAMPLE_NONE #define PJMEDIA_RESAMPLE_IMP PJMEDIA_RESAMPLE_LIBRESAMPLE /* Use the lighter WSOLA implementation */ #define PJMEDIA_WSOLA_IMP PJMEDIA_WSOLA_IMP_WSOLA_LITE /* We probably need more buffers especially if MDA audio backend * is used, so increase the limit */ #define PJMEDIA_SOUND_BUFFER_COUNT 32 /* * PJSIP settings. */ /* Disable safe module access, since we don't use multithreading */ #define PJSIP_SAFE_MODULE 0 /* Use large enough packet size */ #define PJSIP_MAX_PKT_LEN 2000 /* Symbian has problem with too many large blocks */ #define PJSIP_POOL_LEN_ENDPT 1000 #define PJSIP_POOL_INC_ENDPT 1000 #define PJSIP_POOL_RDATA_LEN 2000 #define PJSIP_POOL_RDATA_INC 2000 #define PJSIP_POOL_LEN_TDATA 2000 #define PJSIP_POOL_INC_TDATA 512 #define PJSIP_POOL_LEN_UA 2000 #define PJSIP_POOL_INC_UA 1000 #define PJSIP_POOL_TSX_LAYER_LEN 256 #define PJSIP_POOL_TSX_LAYER_INC 256 #define PJSIP_POOL_TSX_LEN 512 #define PJSIP_POOL_TSX_INC 128 /* * PJSUA settings. */ /* Default codec quality, previously was set to 5, however it is now * set to 4 to make sure pjsua instantiates resampler with small filter. */ #define PJSUA_DEFAULT_CODEC_QUALITY 4 /* Set maximum number of dialog/transaction/calls to minimum */ #define PJSIP_MAX_TSX_COUNT 31 #define PJSIP_MAX_DIALOG_COUNT 31 #define PJSUA_MAX_CALLS 4 /* Other pjsua settings */ #define PJSUA_MAX_ACC 4 #define PJSUA_MAX_PLAYERS 4 #define PJSUA_MAX_RECORDERS 4 #define PJSUA_MAX_CONF_PORTS (PJSUA_MAX_CALLS+2*PJSUA_MAX_PLAYERS) #define PJSUA_MAX_BUDDIES 32 #endif /* * Additional configuration to activate APS-Direct feature for * Nokia S60 target * * Please see http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct */ #ifdef PJ_CONFIG_NOKIA_APS_DIRECT /* MUST use switchboard rather than the conference bridge */ #define PJMEDIA_CONF_USE_SWITCH_BOARD 1 /* Enable APS sound device backend and disable MDA & VAS */ #define PJMEDIA_AUDIO_DEV_HAS_SYMB_MDA 0 #define PJMEDIA_AUDIO_DEV_HAS_SYMB_APS 1 #define PJMEDIA_AUDIO_DEV_HAS_SYMB_VAS 0 /* Enable passthrough codec framework */ #define PJMEDIA_HAS_PASSTHROUGH_CODECS 1 /* Select codecs to disable */ #define PJMEDIA_HAS_G711_CODEC 0 #define PJMEDIA_HAS_L16_CODEC 0 #define PJMEDIA_HAS_GSM_CODEC 1 #define PJMEDIA_HAS_SPEEX_CODEC 0 #define PJMEDIA_HAS_ILBC_CODEC 0 #define PJMEDIA_HAS_G722_CODEC 0 #define PJMEDIA_HAS_INTEL_IPP 0 /* And selectively enable which codecs are supported by the handset */ #define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMU 0 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMA 0 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_AMR 0 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_G729 1 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_ILBC 0 #endif /* * Additional configuration to activate VAS-Direct feature for * Nokia S60 target * * Please see http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct */ #ifdef PJ_CONFIG_NOKIA_VAS_DIRECT /* MUST use switchboard rather than the conference bridge */ #define PJMEDIA_CONF_USE_SWITCH_BOARD 1 /* Enable VAS sound device backend and disable MDA & APS */ #define PJMEDIA_AUDIO_DEV_HAS_SYMB_MDA 0 #define PJMEDIA_AUDIO_DEV_HAS_SYMB_APS 0 #define PJMEDIA_AUDIO_DEV_HAS_SYMB_VAS 1 /* Enable passthrough codec framework */ #define PJMEDIA_HAS_PASSTHROUGH_CODECS 1 /* Select codecs to disable */ #define PJMEDIA_HAS_G711_CODEC 0 #define PJMEDIA_HAS_L16_CODEC 0 #define PJMEDIA_HAS_GSM_CODEC 1 #define PJMEDIA_HAS_SPEEX_CODEC 0 #define PJMEDIA_HAS_ILBC_CODEC 0 #define PJMEDIA_HAS_G722_CODEC 0 #define PJMEDIA_HAS_INTEL_IPP 0 /* And selectively enable which codecs are supported by the handset */ #define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMU 0 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMA 0 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_AMR 0 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_G729 1 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_ILBC 0 /* Specify Symbian VAS version */ # define PJMEDIA_AUDIO_DEV_SYMB_VAS_VERSION 2 #endif /* * Configuration to activate "APS-Direct" media mode on Windows, * useful for testing purposes only. */ #ifdef PJ_CONFIG_WIN32_WMME_DIRECT /* MUST use switchboard rather than the conference bridge */ #define PJMEDIA_CONF_USE_SWITCH_BOARD 1 /* Only WMME supports the "direct" feature */ #define PJMEDIA_AUDIO_DEV_HAS_PORTAUDIO 0 #define PJMEDIA_AUDIO_DEV_HAS_WMME 1 /* Enable passthrough codec framework */ #define PJMEDIA_HAS_PASSTHROUGH_CODECS 1 /* Select codecs to disable */ #define PJMEDIA_HAS_G711_CODEC 0 #define PJMEDIA_HAS_L16_CODEC 0 #define PJMEDIA_HAS_GSM_CODEC 1 #define PJMEDIA_HAS_SPEEX_CODEC 0 #define PJMEDIA_HAS_ILBC_CODEC 0 #define PJMEDIA_HAS_G722_CODEC 0 #define PJMEDIA_HAS_INTEL_IPP 0 /* Only PCMA and PCMU are supported by WMME-direct */ #define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMU 1 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMA 0 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_AMR 0 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_G729 0 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_ILBC 0 #endif /* * iPhone sample settings. */ #if PJ_CONFIG_IPHONE /* * PJLIB settings. */ /* Disable floating point support */ #define PJ_HAS_FLOATING_POINT 0 /* * PJMEDIA settings */ /* We have our own native CoreAudio backend */ #define PJMEDIA_AUDIO_DEV_HAS_PORTAUDIO 0 #define PJMEDIA_AUDIO_DEV_HAS_WMME 0 #define PJMEDIA_AUDIO_DEV_HAS_COREAUDIO 1 /* The CoreAudio backend has built-in echo canceller! */ #define PJMEDIA_HAS_SPEEX_AEC 0 /* Disable some codecs */ #define PJMEDIA_HAS_L16_CODEC 0 #define PJMEDIA_HAS_G722_CODEC 0 /* Use the built-in CoreAudio's iLBC codec (yay!) */ #define PJMEDIA_HAS_ILBC_CODEC 1 #define PJMEDIA_ILBC_CODEC_USE_COREAUDIO 1 /* Fine tune Speex's default settings for best performance/quality */ #define PJMEDIA_CODEC_SPEEX_DEFAULT_QUALITY 5 /* * PJSIP settings. */ /* Increase allowable packet size, just in case */ //#define PJSIP_MAX_PKT_LEN 2000 /* * PJSUA settings. */ /* Default codec quality, previously was set to 5, however it is now * set to 4 to make sure pjsua instantiates resampler with small filter. */ #define PJSUA_DEFAULT_CODEC_QUALITY 4 /* Set maximum number of dialog/transaction/calls to minimum */ #define PJSIP_MAX_TSX_COUNT 31 #define PJSIP_MAX_DIALOG_COUNT 31 #define PJSUA_MAX_CALLS 4 /* Other pjsua settings */ #define PJSUA_MAX_ACC 4 #define PJSUA_MAX_PLAYERS 4 #define PJSUA_MAX_RECORDERS 4 #define PJSUA_MAX_CONF_PORTS (PJSUA_MAX_CALLS+2*PJSUA_MAX_PLAYERS) #define PJSUA_MAX_BUDDIES 32 #endif /* * Minimum size */ #ifdef PJ_CONFIG_MINIMAL_SIZE # undef PJ_OS_HAS_CHECK_STACK # define PJ_OS_HAS_CHECK_STACK 0 # define PJ_LOG_MAX_LEVEL 0 # define PJ_ENABLE_EXTRA_CHECK 0 # define PJ_HAS_ERROR_STRING 0 # undef PJ_IOQUEUE_MAX_HANDLES /* Putting max handles to lower than 32 will make pj_fd_set_t size smaller * than native fdset_t and will trigger assertion on sock_select.c. */ # define PJ_IOQUEUE_MAX_HANDLES 32 # define PJ_CRC32_HAS_TABLES 0 # define PJSIP_MAX_TSX_COUNT 15 # define PJSIP_MAX_DIALOG_COUNT 15 # define PJSIP_UDP_SO_SNDBUF_SIZE 4000 # define PJSIP_UDP_SO_RCVBUF_SIZE 4000 # define PJMEDIA_HAS_ALAW_ULAW_TABLE 0 #elif defined(PJ_CONFIG_MAXIMUM_SPEED) # define PJ_SCANNER_USE_BITWISE 0 # undef PJ_OS_HAS_CHECK_STACK # define PJ_OS_HAS_CHECK_STACK 0 # define PJ_LOG_MAX_LEVEL 3 # define PJ_ENABLE_EXTRA_CHECK 0 # define PJ_IOQUEUE_MAX_HANDLES 5000 # define PJSIP_MAX_TSX_COUNT ((640*1024)-1) # define PJSIP_MAX_DIALOG_COUNT ((640*1024)-1) # define PJSIP_UDP_SO_SNDBUF_SIZE (24*1024*1024) # define PJSIP_UDP_SO_RCVBUF_SIZE (24*1024*1024) # define PJ_DEBUG 0 # define PJSIP_SAFE_MODULE 0 # define PJ_HAS_STRICMP_ALNUM 0 # define PJ_HASH_USE_OWN_TOLOWER 1 # define PJSIP_UNESCAPE_IN_PLACE 1 # ifdef PJ_WIN32 # define PJSIP_MAX_NET_EVENTS 10 # endif # define PJSUA_MAX_CALLS 512 #endif ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ // The part below will be overwritten by automated test // =BEGIN #define SND_HAS_APS 0 #define SND_HAS_VAS 1 #define SND_HAS_MDA 0 // =END TARGET symbian_ua_gui.exe UID 0x100039CE 0xEBD12EE4 VENDORID 0 TARGETTYPE exe EPOCSTACKSIZE 0x8000 MACRO PJ_M_I386=1 MACRO PJ_SYMBIAN=1 MACRO PJ_CONFIG_NOKIA_VAS_DIRECT=1 SYSTEMINCLUDE \epoc32\include SYSTEMINCLUDE \epoc32\include\variant SYSTEMINCLUDE \epoc32\include\ecom SYSTEMINCLUDE \epoc32\include\libc SYSTEMINCLUDE ..\..\..\..\pjlib\include SYSTEMINCLUDE ..\..\..\..\pjlib-util\include SYSTEMINCLUDE ..\..\..\..\pjnath\include SYSTEMINCLUDE ..\..\..\..\pjmedia\include SYSTEMINCLUDE ..\..\..\..\pjsip\include USERINCLUDE ..\inc USERINCLUDE ..\data SOURCEPATH ..\data START RESOURCE symbian_ua_gui.rss HEADER TARGETPATH resource\apps END //RESOURCE START RESOURCE symbian_ua_gui_reg.rss TARGETPATH \private\10003a3f\apps END //RESOURCE // Symbian SDK Libraries LIBRARY euser.lib apparc.lib cone.lib eikcore.lib avkon.lib LIBRARY commonengine.lib efsrv.lib estor.lib eikcoctl.lib eikdlg.lib LIBRARY eikctl.lib bafl.lib fbscli.lib aknnotify.lib aknicon.lib LIBRARY etext.lib gdi.lib egul.lib insock.lib LIBRARY ecom.lib inetprotutil.lib http.lib esock.lib LIBRARY charconv.lib estlib.lib LIBRARY securesocket.lib x509.lib crypto.lib x500.lib LIBRARY hal.lib // PJSIP 1.X Libraries // Ordering static libs based on dependencies, most to least dependent, // this could be necessary for some SDKs, e.g: S60 3rd MR STATICLIBRARY pjsua_lib.lib STATICLIBRARY pjsip_ua.lib pjsip_simple.lib pjsip.lib STATICLIBRARY libgsmcodec.lib libspeexcodec.lib STATICLIBRARY libg7221codec.lib libpassthroughcodec.lib STATICLIBRARY pjmedia.lib STATICLIBRARY pjmedia_audiodev.lib STATICLIBRARY pjsdp.lib STATICLIBRARY pjnath.lib STATICLIBRARY pjlib_util.lib pjlib.lib STATICLIBRARY libsrtp.lib STATICLIBRARY libresample.lib #if SND_HAS_APS LIBRARY APSSession2.lib #endif #if SND_HAS_VAS LIBRARY VoIPAudioIntfc.lib #endif #if SND_HAS_MDA LIBRARY mediaclientaudiostream.lib LIBRARY mediaclientaudioinputstream.lib #endif #if SND_HAS_APS || SND_HAS_VAS CAPABILITY NetworkServices LocalServices ReadUserData WriteUserData UserEnvironment MultimediaDD #else CAPABILITY NetworkServices LocalServices ReadUserData WriteUserData UserEnvironment #endif LANG 01 START BITMAP symbian_ua_gui.mbm HEADER TARGETPATH \resource\apps SOURCEPATH ..\gfx SOURCE c12,1 list_icon.bmp list_icon_mask.bmp END SOURCEPATH ..\src SOURCE symbian_ua_guiContainerView.cpp SOURCE symbian_ua_guiContainer.cpp SOURCE symbian_ua_guiAppUi.cpp SOURCE symbian_ua_guiDocument.cpp SOURCE symbian_ua_guiApplication.cpp SOURCE symbian_ua_guiSettingItemList.cpp SOURCE Symbian_ua_guiSettingItemListSets.cpp SOURCE symbian_ua_guiSettingItemListView.cpp SOURCE symbian_ua.cpp -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20120315/9617b150/attachment-0001.html>
