for #133, alloc and free rtp port.

trunk/src/app/srs_app_listener.cpp

@@ -77,6 +77,7 @@ SrsUdpListener::SrsUdpListener(ISrsUdpHandler* h, int p)
 
 SrsUdpListener::~SrsUdpListener()
 {
+    // close the stfd to trigger thread to interrupted.
     srs_close_stfd(stfd);
 
     pthread->stop();
@@ -144,7 +145,7 @@ int SrsUdpListener::cycle()
 {
     int ret = ERROR_SUCCESS;
     
-    for (;;) {
+    while (pthread->can_loop()) {
         // TODO: FIXME: support ipv6, @see man 7 ipv6
         sockaddr_in from;
         int nb_from = sizeof(sockaddr_in);
@@ -181,6 +182,7 @@ SrsTcpListener::SrsTcpListener(ISrsTcpHandler* h, int p)
 
 SrsTcpListener::~SrsTcpListener()
 {
+    // close the stfd to trigger thread to interrupted.
     srs_close_stfd(stfd);
 
     pthread->stop();

trunk/src/app/srs_app_rtsp.cpp

@@ -36,21 +36,41 @@ using namespace std;
 
 #ifdef SRS_AUTO_STREAM_CASTER
 
-ISrsRtspHandler::ISrsRtspHandler()
+SrsRtpConn::SrsRtpConn(SrsRtspConn* r, int p)
 {
+    rtsp = r;
+    _port = p;
+    listener = new SrsUdpListener(this, p);
 }
 
-ISrsRtspHandler::~ISrsRtspHandler()
+SrsRtpConn::~SrsRtpConn()
 {
+    srs_freep(listener);
 }
 
-SrsRtspConn::SrsRtspConn(SrsRtspCaster* c, st_netfd_t fd, std::string o, int lpmin, int lpmax)
+int SrsRtpConn::port()
+{
+    return _port;
+}
+
+int SrsRtpConn::listen()
+{
+    return listener->listen();
+}
+
+int SrsRtpConn::on_udp_packet(sockaddr_in* from, char* buf, int nb_buf)
+{
+    int ret = ERROR_SUCCESS;
+    return ret;
+}
+
+SrsRtspConn::SrsRtspConn(SrsRtspCaster* c, st_netfd_t fd, std::string o)
 {
     output = o;
-    local_port_min = lpmin;
-    local_port_max = lpmax;
 
     session = "O9EaZ4bf"; // TODO: FIXME: generate session id.
+    video_rtp = NULL;
+    audio_rtp = NULL;
 
     caster = c;
     stfd = fd;
@@ -64,6 +84,9 @@ SrsRtspConn::~SrsRtspConn()
     srs_close_stfd(stfd);
     trd->stop();
 
+    srs_freep(video_rtp);
+    srs_freep(audio_rtp);
+
     srs_freep(trd);
     srs_freep(skt);
     srs_freep(rtsp);
@@ -103,6 +126,8 @@ int SrsRtspConn::do_cycle()
             }
         } else if (req->is_announce()) {
             srs_assert(req->sdp);
+            video_id = req->sdp->video_stream_id;
+            audio_id = req->sdp->audio_stream_id;
             sps = req->sdp->video_sps;
             pps = req->sdp->video_pps;
             asc = req->sdp->audio_sh;
@@ -119,11 +144,31 @@ int SrsRtspConn::do_cycle()
             }
         } else if (req->is_setup()) {
             srs_assert(req->transport);
+            int lpm = 0;
+            if ((ret = caster->alloc_port(&lpm)) != ERROR_SUCCESS) {
+                srs_error("rtsp: alloc port failed. ret=%d", ret);
+                return ret;
+            }
+
+            SrsRtpConn* rtp = NULL;
+            if (req->stream_id == video_id) {
+                srs_freep(video_rtp);
+                rtp = video_rtp = new SrsRtpConn(this, lpm);
+            } else {
+                srs_freep(audio_rtp);
+                rtp = audio_rtp = new SrsRtpConn(this, lpm);
+            }
+            if ((ret = rtp->listen()) != ERROR_SUCCESS) {
+                srs_error("rtsp: rtp listen at port=%d failed. ret=%d", lpm, ret);
+                return ret;
+            }
+            srs_trace("rtsp: rtp listen at port=%d ok.", lpm);
+
             SrsRtspSetupResponse* res = new SrsRtspSetupResponse(req->seq);
             res->client_port_min = req->transport->client_port_min;
             res->client_port_max = req->transport->client_port_max;
-            res->local_port_min = local_port_min;
-            res->local_port_max = local_port_max;
+            res->local_port_min = lpm;
+            res->local_port_max = lpm + 1;
             res->session = session;
             if ((ret = rtsp->send_message(res)) != ERROR_SUCCESS) {
                 if (!srs_is_client_gracefully_close(ret)) {
@@ -165,6 +210,14 @@ int SrsRtspConn::cycle()
 
 void SrsRtspConn::on_thread_stop()
 {
+    if (video_rtp) {
+        caster->free_port(video_rtp->port(), video_rtp->port() + 1);
+    }
+
+    if (audio_rtp) {
+        caster->free_port(audio_rtp->port(), audio_rtp->port() + 1);
+    }
+
     caster->remove(this);
 }
 
@@ -184,16 +237,40 @@ SrsRtspCaster::~SrsRtspCaster()
         srs_freep(conn);
     }
     clients.clear();
+    used_ports.clear();
+}
+
+int SrsRtspCaster::alloc_port(int* pport)
+{
+    int ret = ERROR_SUCCESS;
+
+    // use a pair of port.
+    for (int i = local_port_min; i < local_port_max - 1; i += 2) {
+        if (!used_ports[i]) {
+            used_ports[i] = true;
+            used_ports[i + 1] = true;
+            *pport = i;
+            break;
+        }
+    }
+    srs_info("rtsp: alloc port=%d-%d", *pport, *pport + 1);
+
+    return ret;
+}
+
+void SrsRtspCaster::free_port(int lpmin, int lpmax)
+{
+    for (int i = lpmin; i < lpmax; i++) {
+        used_ports[i] = false;
+    }
+    srs_trace("rtsp: free rtp port=%d-%d", lpmin, lpmax);
 }
 
-int SrsRtspCaster::serve_client(st_netfd_t stfd)
+int SrsRtspCaster::on_tcp_client(st_netfd_t stfd)
 {
     int ret = ERROR_SUCCESS;
 
-    SrsRtspConn* conn = new SrsRtspConn(
-        this, stfd, 
-        output, local_port_min, local_port_max
-    );
+    SrsRtspConn* conn = new SrsRtspConn(this, stfd, output);
 
     if ((ret = conn->serve()) != ERROR_SUCCESS) {
         srs_error("rtsp: serve client failed. ret=%d", ret);

trunk/src/app/srs_app_rtsp.hpp

@@ -32,30 +32,38 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 
 #include <string>
 #include <vector>
+#include <map>
 
 #include <srs_app_st.hpp>
 #include <srs_app_thread.hpp>
+#include <srs_app_listener.hpp>
 
 #ifdef SRS_AUTO_STREAM_CASTER
 
-class SrsConfDirective;
 class SrsStSocket;
+class SrsRtspConn;
 class SrsRtspStack;
 class SrsRtspCaster;
+class SrsConfDirective;
 
 /**
-* the handler for rtsp handler.
+* a rtp connection which transport a stream.
 */
-class ISrsRtspHandler
+class SrsRtpConn: public ISrsUdpHandler
 {
+private:
+    SrsUdpListener* listener;
+    SrsRtspConn* rtsp;
+    int _port;
 public:
-    ISrsRtspHandler();
-    virtual ~ISrsRtspHandler();
+    SrsRtpConn(SrsRtspConn* r, int p);
+    virtual ~SrsRtpConn();
 public:
-    /**
-    * serve the rtsp connection.
-    */
-    virtual int serve_client(st_netfd_t stfd) = 0;
+    virtual int port();
+    virtual int listen();
+// interface ISrsUdpHandler
+public:
+    virtual int on_udp_packet(sockaddr_in* from, char* buf, int nb_buf);
 };
 
 /**
@@ -65,10 +73,14 @@ class SrsRtspConn : public ISrsThreadHandler
 {
 private:
     std::string output;
-    int local_port_min;
-    int local_port_max;
 private:
     std::string session;
+    // video stream.
+    std::string video_id;
+    SrsRtpConn* video_rtp;
+    // audio stream.
+    std::string audio_id;
+    SrsRtpConn* audio_rtp;
     // video sequence header.
     std::string sps;
     std::string pps;
@@ -81,7 +93,7 @@ private:
     SrsRtspCaster* caster;
     SrsThread* trd;
 public:
-    SrsRtspConn(SrsRtspCaster* c, st_netfd_t fd, std::string o, int lpmin, int lpmax);
+    SrsRtspConn(SrsRtspCaster* c, st_netfd_t fd, std::string o);
     virtual ~SrsRtspConn();
 public:
     virtual int serve();
@@ -96,19 +108,32 @@ public:
 /**
 * the caster for rtsp.
 */
-class SrsRtspCaster : public ISrsRtspHandler
+class SrsRtspCaster : public ISrsTcpHandler
 {
 private:
     std::string output;
     int local_port_min;
     int local_port_max;
+    // key: port, value: whether used.
+    std::map<int, bool> used_ports;
 private:
     std::vector<SrsRtspConn*> clients;
 public:
     SrsRtspCaster(SrsConfDirective* c);
     virtual ~SrsRtspCaster();
 public:
-    virtual int serve_client(st_netfd_t stfd);
+    /**
+    * alloc a rtp port from local ports pool.
+    * @param pport output the rtp port.
+    */
+    virtual int alloc_port(int* pport);
+    /**
+    * free the alloced rtp port.
+    */
+    virtual void free_port(int lpmin, int lpmax);
+// interface ISrsTcpHandler
+public:
+    virtual int on_tcp_client(st_netfd_t stfd);
 // internal methods.
 public:
     virtual void remove(SrsRtspConn* conn);

trunk/src/app/srs_app_server.cpp

@@ -220,7 +220,7 @@ int SrsRtspListener::on_tcp_client(st_netfd_t stfd)
 {
     int ret = ERROR_SUCCESS;
     
-    if ((ret = caster->serve_client(stfd)) != ERROR_SUCCESS) {
+    if ((ret = caster->on_tcp_client(stfd)) != ERROR_SUCCESS) {
         srs_warn("accept client error. ret=%d", ret);
         return ret;
     }

trunk/src/app/srs_app_server.hpp

@@ -47,8 +47,8 @@ class SrsIngester;
 class SrsHttpHeartbeat;
 class SrsKbps;
 class SrsConfDirective;
+class ISrsTcpHandler;
 class ISrsUdpHandler;
-class ISrsRtspHandler;
 class SrsUdpListener;
 class SrsTcpListener;
 
@@ -111,8 +111,7 @@ class SrsRtspListener : virtual public SrsListener, virtual public ISrsTcpHandle
 {
 private:
     SrsTcpListener* listener;
-private:
-    ISrsRtspHandler* caster;
+    ISrsTcpHandler* caster;
 public:
     SrsRtspListener(SrsServer* server, SrsListenerType type, SrsConfDirective* c);
     virtual ~SrsRtspListener();

trunk/src/app/srs_app_thread.cpp

@@ -62,6 +62,7 @@ SrsThread::SrsThread(const char* name, ISrsThreadHandler* thread_handler, int64_
     
     tid = NULL;
     loop = false;
+    really_terminated = true;
     _cid = -1;
     _joinable = joinable;
     
@@ -120,10 +121,27 @@ void SrsThread::stop()
         // which will terminate the cycle thread.
         st_thread_interrupt(tid);
         
-        // wait the thread to exit.
-        int ret = st_thread_join(tid, NULL);
-        // TODO: FIXME: the join maybe failed, should use a variable to ensure thread terminated.
-        srs_assert(ret == 0);
+        // when joinable, wait util quit.
+        if (_joinable) {
+            // wait the thread to exit.
+            int ret = st_thread_join(tid, NULL);
+            if (ret) {
+                srs_warn("core: ignore join thread failed.");
+            }
+
+            // wait the thread actually terminated.
+            // sometimes the thread join return -1, for example, 
+            // when thread use st_recvfrom, the thread join return -1.
+            // so here, we use a variable to ensure the thread stopped.
+            while (!really_terminated) {
+                st_usleep(10 * 1000);
+
+                if (really_terminated) {
+                    break;
+                }
+                srs_warn("core: wait thread to actually terminated");
+            }
+        }
         
         tid = NULL;
     }
@@ -151,6 +169,9 @@ void SrsThread::thread_cycle()
     srs_assert(handler);
     handler->on_thread_start();
 
+    // thread is running now.
+    really_terminated = false;
+    
     // wait for cid to ready, for parent thread to get the cid.
     while (!can_run && loop) {
         st_usleep(10 * 1000);
@@ -191,6 +212,9 @@ failed:
     
     handler->on_thread_stop();
     srs_info("thread %s cycle finished", _name);
+
+    // readly terminated now.
+    really_terminated = true;
 }
 
 void* SrsThread::thread_fun(void* arg)

trunk/src/app/srs_app_thread.hpp

@@ -143,6 +143,7 @@ private:
     int _cid;
     bool loop;
     bool can_run;
+    bool really_terminated;
     bool _joinable;
     const char* _name;
 private:

trunk/src/protocol/srs_rtsp_stack.cpp

@@ -746,6 +746,18 @@ int SrsRtspStack::do_recv_message(SrsRtspRequest* req)
         }
     }
 
+    // for setup, parse the stream id from uri.
+    if (req->is_setup()) {
+        size_t pos = string::npos;
+        if ((pos = req->uri.rfind("/")) != string::npos) {
+            req->stream_id = req->uri.substr(pos + 1);
+        }
+        if ((pos = req->stream_id.find("=")) != string::npos) {
+            req->stream_id = req->stream_id.substr(pos + 1);
+        }
+        srs_info("rtsp: setup stream id=%s", req->stream_id.c_str());
+    }
+
     // parse rdp body.
     long consumed = 0;
     while (consumed < req->content_length) {

trunk/src/protocol/srs_rtsp_stack.hpp

@@ -337,6 +337,7 @@ public:
     * assumed. It is interpreted according to [H14.14].
     */
     long content_length;
+
     /**
     * the sdp in announce, NULL for no sdp.
     */
@@ -345,6 +346,10 @@ public:
     * the transport in setup, NULL for no transport.
     */
     SrsRtspTransport* transport;
+    /**
+    * for setup message, parse the stream id from uri.
+    */
+    std::string stream_id;
 public:
     SrsRtspRequest();
     virtual ~SrsRtspRequest();