Patch for Libjingle with GCC 4.2.4 on Ubuntu

目录 Linux, 编程

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);
     }
#endif

     if (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>.

用 Linux 命令行工具自动追踪车票信息

目录 Linux

前一篇博客中说到我买票失败的经历,也充分表达了我想买一张二手座票的意愿。怎么办呢?只好到网上各二手火车票信息平台去找了。心肠不好的人肯定幸灾了祸地在想:“哈哈,这个倒霉的小伙儿该对着浏览器不停地按 F5 了!” 你才 F5 呢,你们全家都 F5。那是典型的 Windows 用户的想法,不要以为 Linux User 跟你一样傻。

前面都是玩笑话 :),本文只是想介绍一下在 Linux 下有什么更方便的方法来追踪网页发布的信息,以展示 Linux 的命令行工具有多强大(也响应一下 Eric 师兄的文章:完全用键盘工作-3:常用的命令行工具)。

我们就拿火车网为例,通常情况下 Windows 用户为了在火车网上找一张二手火车票信息,会不断地到查询页面刷新,看有没有自己需要的车票。而一个 Linux 用户的做法会有何不同呢?一般来讲他会用工具来做这件事情,而不是在那傻刷,浪费时间。

怎么做呢?有很多种方法,我这里来介绍一种比较好玩的方法,用脚本自动跟踪信息,如果有结果就发送一个 Gtalk 消息给自己。

首先,写一个命令行发送 Gtalk 消息的 Python 脚本。其实我本打算用 freetalk 来做这件事的,奈何咱学识浅薄,不懂 freetalk 脚本该怎么写,也不知道 scheme 语言为何物。没办法,只好用 Python 来做了。下面内容就是用 Python 发送 gtalk 消息的脚本(需要 Linux 上装有 python-xmpp):

#!/usr/bin/env python
# -*- encoding: utf-8 -*-
# Usage: gtsent.py "SOMEBODY@gmail.com" "Message"

import xmpp
import sys

login = 'USERNAME' # without @gmail.com
pwd   = 'PASSWORD'

cnx = xmpp.Client('gmail.com', debug=[])
cnx.connect( server=('talk.google.com', 5223) )
cnx.auth(login, pwd, 'python')

cnx.send(xmpp.Message(sys.argv[1], sys.argv[2]))

将以上内容保存为 gtsend.py 文件,chmod u+x gtsend.py,修改 USERNAME 和 PASSWORD 为你的另一个非[常用] gmail 帐户名和密码。这样执行 ./gtsend.py SOMEBODY@gmail.com "Message" 就可以给 SOMEBODY@gmail.com 发送消息了(当然了,前提是 SOMEBODY@gmail.com 好友列表中有 USERNAME@gmail.com,请注意这里大写只是为了方便阅读)。

其次,写一个 Shell 脚本,用来追踪网页,过滤信息并发送 gtalk 消息。这个就更简单了,使用火车网提供的查询表单接口,用 wget 抓下来,再 grep 一下即可,bash 脚本如下:

#!/bin/bash
URL="http://www.huoche.com.cn/piao/2piaoserach.asp?ICheci=T65&type=0"
RESULT=`wget -O - $URL | iconv -f gbk -t utf8 | grep -i -e "t65.*南京.*硬座\
.*2008-9-28"`
if [ $(echo $RESULT | wc -c) -ge 5 ]
then
  /home/solrex/gtsend.py "YOURSELF@gmail.com" "$RESULT $URL"
fi

将该脚本保存为 get_tickt.sh,chmod u+x get_tickt.sh。这个脚本的工作流程是:wget 以 GET 方式提交对 T65 转让车票的查询,得到的结果输出到标准输出,然后将 GBK 编码转换为 UTF8 编码,再 grep 看是否含有“T65 南京 硬座 2008-9-28“关键词。如果有的话,用 gtsend.py 发送一个提醒消息给自己 gtalk 帐户;如果没有结果,什么都不做。

最后,将上面脚本加入 cron 列表每 10 分钟定时执行一次。 执行 crontab -e,添加下面一行即可(注意需要修改到该脚本的路径):

*/10 * * * * /home/solrex/get_ticket.sh

然后呢,你就可以高枕无忧,开着 Gtalk 等消息吧。当然,不一定能等得到 :(,唉,对我们来说, No news is BAD news!

当然,根据不同情况,你可以把追踪的信息换成别的东西。比如追女孩子的时候,可以用上面方法来实时跟踪她的最新博客,实时跟踪她在 BBS 上的留言,永远保持自己沙发的地位,说不定人家就感受到了你的关心,然后...具体方法我就不教了哈...

Google 词典和 Gtalk 翻译机器人

目录 IT杂谈

由于我工作的机器配置实在太低,1.8GHz CPU, 256M 内存,40G 硬盘,跑一个 Ubuntu 也是非常吃力,我平常只敢开三四个程序,这样每次切换程序还要等个十几秒,唉!

虽然我 Ubuntu 里也装了 Stardict 星际译王,但我轻易不敢再开一个程序,太慢了!现在我发现一个非常有意思的东西解决了我的困扰,Gtalk 翻译机器人。其实这是我在尝试另一项服务 Google 词典时无意中发现的,很奇怪的是,我记得曾经看到一个 Google 官方的关于 Gtalk 翻译机器人的页面,怎么再也找不到了?只搜索到一个 Gtalk 开发组的博客上的新闻链接

添加 Gtalk 翻译机器人很简单,就是选择添加好友,好友 email 为:语言缩写2语言缩写@bot.talk.google.com。比如汉英翻译的机器人名字是:zh2en@bot.talk.google.com,英汉翻译的机器人名字是:en2zh@bot.talk.google.com。当然了,还有更多,点上面的新闻链接可以查看。

用这个机器人有什么好处呢?一是方便,直接在聊天软件里就可以查词。就像我用那么落后的机器,打开一个词典软件能让它假死半分钟,而 IM 软件总是会开着的,打开一个聊天窗口显然方便和快很多;还有一个好处是 Google 将你的聊天内容记录到 Gmail 里,那么过一段时间整理一下聊天记录就是一个非常好的生词表 :)。查找聊天记录很简单,只需要在 Gmail 上方的搜索栏中输入:from: en2zh@bot.talk.google.com 再点搜索即可。

Goolge 词典也是非常好用的,但不知道为什么在 Google 首页上点 Language Tools 进去以后却没有词典的链接,只有到 more->even more 中找 Translate 才有。