Patch for Libjingle with GCC 4.2.4 on Ubuntu
It is a svn diff result, not a patch, actually.
So, what is Libjingle? Quoted from http://code.google.com/p/libjingle/:
Libjingle, the Google Talk Voice and P2P Interoperability Library, is a set of components we provide to interoperate with Google Talk's peer-to-peer file sharing and voice calling capabilities. The package includes source code for Google's implementation of Jingle and Jingle-Audio, two proposed extensions to the XMPP standard that are currently available in draft form.
You can check out the head revision of Libjingle from its svn repository using command:
svn checkout http://libjingle.googlecode.com/svn/trunk/ libjingle-read-only
Then ``./autogen.sh'' and ``make'' as we usually do for building a *nix software. You will find many errors during ``./autogen.sh'' and ``make''. To fix them, first, some LIBs should be installed:
sudo apt-get install build-essential libexpat1-dev libglib2.0-dev libogg-dev libssl-dev libasound2-dev libspeex-dev openssl libortp7-dev libmediastreamer0-dev libavcodec-dev
I am not very sure if these LIBs are enough. If you have some problem with this, please let me know.
Even if you have all of these LIBs installed, you will still get some errors such as:
../../talk/base/stringutils.h:272: error: extra qualification 'talk_base::Traits::' on member 'empty_str'
../../talk/base/base64.h:26: error: extra qualification ‘talk_base::Base64::’ on member ‘Base64Table’
../../talk/base/base64.h:27: error: extra qualification ‘talk_base::Base64::’ on member ‘DecodeTable’
So here is a patch for source code errors like this. IMPORTANT NOTE: gcc version 4.2.4 on Ubuntu 8.04, libortp7.
Index: talk/p2p/base/sessionmanager.h
===================================================================
--- talk/p2p/base/sessionmanager.h (revision 7)
+++ talk/p2p/base/sessionmanager.h (working copy)
@@ -156,7 +156,7 @@// Creates and returns an error message from the given components. The
// caller is responsible for deleting this.
- buzz::XmlElement* SessionManager::CreateErrorMessage(
+ buzz::XmlElement* CreateErrorMessage(
const buzz::XmlElement* stanza,
const buzz::QName& name,
const std::string& type,
Index: talk/session/phone/linphonemediaengine.cc
===================================================================
--- talk/session/phone/linphonemediaengine.cc (revision 7)
+++ talk/session/phone/linphonemediaengine.cc (working copy)
@@ -80,24 +80,24 @@
}
#endif
#ifdef HAVE_SPEEX
- if (i->name == speex_wb.mime_type && i->clockrate == speex_wb.clock_rate) {
- rtp_profile_set_payload(&av_profile, i->id, &speex_wb);
- } else if (i->name == speex_nb.mime_type && i->clockrate == speex_nb.clock_rate) {
- rtp_profile_set_payload(&av_profile, i->id, &speex_nb);
+ if (i->name == payload_type_speex_wb.mime_type && i->clockrate == payload_type_speex_wb.clock_rate) {
+ rtp_profile_set_payload(&av_profile, i->id, &payload_type_speex_wb);
+ } else if (i->name == payload_type_speex_nb.mime_type && i->clockrate == payload_type_speex_nb.clock_rate) {
+ rtp_profile_set_payload(&av_profile, i->id, &payload_type_speex_nb);
}
#endifif (i->id == 0)
- rtp_profile_set_payload(&av_profile, 0, &pcmu8000);
+ rtp_profile_set_payload(&av_profile, 0, &payload_type_pcmu8000);- if (i->name == telephone_event.mime_type) {
- rtp_profile_set_payload(&av_profile, i->id, &telephone_event);
+ if (i->name == payload_type_telephone_event.mime_type) {
+ rtp_profile_set_payload(&av_profile, i->id, &payload_type_telephone_event);
}
if (first) {
LOG(LS_INFO) << "Using " << i->name << "/" << i->clockrate;
pt_ = i->id;
- audio_stream_ = audio_stream_start(&av_profile, 2000, "127.0.0.1", 3000, i->id, 250);
+ audio_stream_ = audio_stream_start(&av_profile, 2000, (char *)"127.0.0.1", 3000, i->id, 250);
first = false;
}
}
@@ -106,7 +106,7 @@
// We're being asked to set an empty list of codecs. This will only happen when
// working with a buggy client; let's try PCMU.
LOG(LS_WARNING) << "Received empty list of codces; using PCMU/8000";
- audio_stream_ = audio_stream_start(&av_profile, 2000, "127.0.0.1", 3000, 0, 250);
+ audio_stream_ = audio_stream_start(&av_profile, 2000, (char *)"127.0.0.1", 3000, 0, 250);
}
}
@@ -114,12 +114,12 @@
bool LinphoneMediaEngine::FindCodec(const Codec &c) {
if (c.id == 0)
return true;
- if (c.name == telephone_event.mime_type)
+ if (c.name == payload_type_telephone_event.mime_type)
return true;
#ifdef HAVE_SPEEX
- if (c.name == speex_wb.mime_type && c.clockrate == speex_wb.clock_rate)
+ if (c.name == payload_type_speex_wb.mime_type && c.clockrate == payload_type_speex_wb.clock_rate)
return true;
- if (c.name == speex_nb.mime_type && c.clockrate == speex_nb.clock_rate)
+ if (c.name == payload_type_speex_nb.mime_type && c.clockrate == payload_type_speex_nb.clock_rate)
return true;
#endif
#ifdef HAVE_ILBC
@@ -171,8 +171,8 @@
#ifdef HAVE_SPEEX
ms_speex_codec_init();- codecs_.push_back(Codec(110, speex_wb.mime_type, speex_wb.clock_rate, 0, 1, 8));
- codecs_.push_back(Codec(111, speex_nb.mime_type, speex_nb.clock_rate, 0, 1, 7));
+ codecs_.push_back(Codec(110, payload_type_speex_wb.mime_type, payload_type_speex_wb.clock_rate, 0, 1, 8));
+ codecs_.push_back(Codec(111, payload_type_speex_nb.mime_type, payload_type_speex_nb.clock_rate, 0, 1, 7));
#endif@@ -181,8 +181,8 @@
codecs_.push_back(Codec(102, payload_type_ilbc.mime_type, payload_type_ilbc.clock_rate, 0, 1, 4));
#endif- codecs_.push_back(Codec(0, pcmu8000.mime_type, pcmu8000.clock_rate, 0, 1, 2));
- codecs_.push_back(Codec(101, telephone_event.mime_type, telephone_event.clock_rate, 0, 1, 1));
+ codecs_.push_back(Codec(0, payload_type_pcmu8000.mime_type, payload_type_pcmu8000.clock_rate, 0, 1, 2));
+ codecs_.push_back(Codec(101, payload_type_telephone_event.mime_type, payload_type_telephone_event.clock_rate, 0, 1, 1));
return true;
}Index: talk/xmpp/xmppclient.h
===================================================================
--- talk/xmpp/xmppclient.h (revision 7)
+++ talk/xmpp/xmppclient.h (working copy)
@@ -138,7 +138,7 @@
}
}- std::string XmppClient::GetStateName(int state) const {
+ std::string GetStateName(int state) const {
switch (state) {
case STATE_PRE_XMPP_LOGIN: return "PRE_XMPP_LOGIN";
case STATE_START_XMPP_LOGIN: return "START_XMPP_LOGIN";
Index: talk/third_party/mediastreamer/msrtprecv.c
===================================================================
--- talk/third_party/mediastreamer/msrtprecv.c (revision 7)
+++ talk/third_party/mediastreamer/msrtprecv.c (working copy)
@@ -26,7 +26,7 @@
MSMessage *msgb_2_ms_message(mblk_t* mp){
MSMessage *msg;
MSBuffer *msbuf;
- if (mp->b_datap->ref_count!=1) return NULL; /* cannot handle properly non-unique buffers*/
+ if (mp->b_datap->db_ref!=1) return NULL; /* cannot handle properly non-unique buffers*/
/* create a MSBuffer using the mblk_t buffer */
msg=ms_message_alloc();
msbuf=ms_buffer_alloc(0);
@@ -120,7 +120,7 @@
gint got=0;
/* we are connected with queues (surely for video)*/
/* use the sync system time to compute a timestamp */
- PayloadType *pt=rtp_profile_get_payload(r->rtpsession->profile,r->rtpsession->payload_type);
+ PayloadType *pt=rtp_profile_get_payload(r->rtpsession->rcv.profile,r->rtpsession->rcv.telephone_events_pt);
if (pt==NULL) {
ms_warning("ms_rtp_recv_process(): NULL RtpPayload- skipping.");
return;
Index: talk/third_party/mediastreamer/audiostream.c
===================================================================
--- talk/third_party/mediastreamer/audiostream.c (revision 7)
+++ talk/third_party/mediastreamer/audiostream.c (working copy)
@@ -112,7 +112,7 @@
RtpSession **recvsend){
RtpSession *rtpr;
rtpr=rtp_session_new(RTP_SESSION_SENDRECV);
- rtp_session_max_buf_size_set(rtpr,MAX_RTP_SIZE);
+ rtp_session_set_recv_buf_size(rtpr,MAX_RTP_SIZE);
rtp_session_set_profile(rtpr,profile);
rtp_session_set_local_addr(rtpr,get_local_addr_for(remip),locport);
if (remport>0) rtp_session_set_remote_addr(rtpr,remip,remport);
@@ -133,7 +133,7 @@
/* creates two rtp filters to recv send streams (remote part)*/
rtps=rtp_session_new(RTP_SESSION_SENDONLY);
- rtp_session_max_buf_size_set(rtps,MAX_RTP_SIZE);
+ rtp_session_set_recv_buf_size(rtps,MAX_RTP_SIZE);
rtp_session_set_profile(rtps,profile);
#ifdef INET6
rtp_session_set_local_addr(rtps,"::",locport+2);
@@ -147,7 +147,7 @@
rtp_session_set_jitter_compensation(rtps,jitt_comp);
rtpr=rtp_session_new(RTP_SESSION_RECVONLY);
- rtp_session_max_buf_size_set(rtpr,MAX_RTP_SIZE);
+ rtp_session_set_recv_buf_size(rtpr,MAX_RTP_SIZE);
rtp_session_set_profile(rtpr,profile);
#ifdef INET6
rtp_session_set_local_addr(rtpr,"::",locport);
@@ -217,8 +217,8 @@
ms_filter_set_property(stream->decoder,MS_FILTER_PROPERTY_FREQ,&pt->clock_rate);
ms_filter_set_property(stream->decoder,MS_FILTER_PROPERTY_BITRATE,&pt->normal_bitrate);
- ms_filter_set_property(stream->encoder,MS_FILTER_PROPERTY_FMTP, (void*)pt->fmtp);
- ms_filter_set_property(stream->decoder,MS_FILTER_PROPERTY_FMTP,(void*)pt->fmtp);
+ ms_filter_set_property(stream->encoder,MS_FILTER_PROPERTY_FMTP, (void*)pt->send_fmtp);
+ ms_filter_set_property(stream->decoder,MS_FILTER_PROPERTY_FMTP,(void*)pt->send_fmtp);
/* create the synchronisation source */
stream->timer=ms_timer_new();
Index: talk/third_party/mediastreamer/msrtpsend.c
===================================================================
--- talk/third_party/mediastreamer/msrtpsend.c (revision 7)
+++ talk/third_party/mediastreamer/msrtpsend.c (working copy)
@@ -85,7 +85,7 @@
{
guint32 clockts;
/* use the sync system time to compute a timestamp */
- PayloadType *pt=rtp_profile_get_payload(r->rtpsession->profile,r->rtpsession->payload_type);
+ PayloadType *pt=rtp_profile_get_payload(r->rtpsession->snd.profile,r->rtpsession->snd.telephone_events_pt);
g_return_val_if_fail(pt!=NULL,0);
clockts=(guint32)(((double)synctime * (double)pt->clock_rate)/1000.0);
ms_trace("ms_rtp_send_process: sync->time=%i clock=%i",synctime,clockts);
Index: talk/base/base64.h
===================================================================
--- talk/base/base64.h (revision 7)
+++ talk/base/base64.h (working copy)
@@ -23,8 +23,8 @@
static std::string decode(const std::string & data);
static std::string encodeFromArray(const char * data, size_t len);
private:
- static const std::string Base64::Base64Table;
- static const std::string::size_type Base64::DecodeTable[];
+ static const std::string Base64Table;
+ static const std::string::size_type DecodeTable[];
};} // namespace talk_base
Index: talk/base/stringutils.h
===================================================================
--- talk/base/stringutils.h (revision 7)
+++ talk/base/stringutils.h (working copy)
@@ -269,7 +269,7 @@
template<>
struct Traits<char> {
typedef std::string string;
- inline static const char* Traits<char>::empty_str() { return ""; }
+ inline static const char* empty_str() { return ""; }
};///////////////////////////////////////////////////////////////////////////////
You killed all these errors? Congratulations! You can start talking with your gtalk friends with command ``call'' in talk/examples/call/ !
PS: If you are working with GCC 4.3.x, more strict checking is applied on the code. However, most errors can be fixed by adding some C headers into the #include fields, such as: <cstdlib>, <cstring>.