mplayer: add patch to fix building with modern versions of live555

From d3195ea13f4a9aae546ff996e53681349a1a3cdb Mon Sep 17 00:00:00 2001

From: sherpya <sherpya@netfarm.it>

Date: Fri, 14 Jun 2013 05:25:38 +0200

Subject: [PATCH 25/27] mpdemux: live555 async interface

From: https://raw.github.com/sherpya/mplayer-be/master/patches/mp/0025-mpdemux-live555-async-interface.patch

Adjust live555 interface code for modern versions of live555.

Signed-off-by: Peter Korsgaard <peter@korsgaard.com>

---

 libmpdemux/demux_rtp.cpp | 51 ++++++++++++++++++++++++++++++++----------------

 2 files changed, 35 insertions(+), 22 deletions(-)

diff --git a/libmpdemux/demux_rtp.cpp b/libmpdemux/demux_rtp.cpp

index ad7a7f1..05d06e0 100644

--- a/libmpdemux/demux_rtp.cpp

+++ b/libmpdemux/demux_rtp.cpp

@@ -19,8 +19,6 @@

  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.

  */

-#define RTSPCLIENT_SYNCHRONOUS_INTERFACE 1

-

 extern "C" {

 // on MinGW, we must include windows.h before the things it conflicts

 #ifdef __MINGW32__    // with.  they are each protected from

@@ -94,15 +92,6 @@ struct RTPState {

 extern "C" char* network_username;

 extern "C" char* network_password;

-static char* openURL_rtsp(RTSPClient* client, char const* url) {

-  // If we were given a user name (and optional password), then use them:

-  if (network_username != NULL) {

-    char const* password = network_password == NULL ? "" : network_password;

-    return client->describeWithPassword(url, network_username, password);

-  } else {

-    return client->describeURL(url);

-  }

-}

 static char* openURL_sip(SIPClient* client, char const* url) {

   // If we were given a user name (and optional password), then use them:

@@ -118,6 +107,19 @@ static char* openURL_sip(SIPClient* client, char const* url) {

 extern AVCodecContext *avcctx;

 #endif

+static char fWatchVariableForSyncInterface;

+static char* fResultString;

+static int fResultCode;

+

+static void responseHandlerForSyncInterface(RTSPClient* rtspClient, int responseCode, char* responseString) {

+  // Set result values:

+  fResultCode = responseCode;

+  fResultString = responseString;

+

+  // Signal a break from the event loop (thereby returning from the blocking command):

+  fWatchVariableForSyncInterface = ~0;

+}

+

 extern "C" int audio_id, video_id, dvdsub_id;

 extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {

   Boolean success = False;

@@ -146,13 +148,19 @@ extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {

 	  rtsp_transport_http = demuxer->stream->streaming_ctrl->url->port;

 	  rtsp_transport_tcp = 1;

 	}

-	rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer", rtsp_transport_http);

+	rtspClient = RTSPClient::createNew(*env, url, verbose, "MPlayer", rtsp_transport_http);

 	if (rtspClient == NULL) {

 	  fprintf(stderr, "Failed to create RTSP client: %s\n",

 		  env->getResultMsg());

 	  break;

 	}

-	sdpDescription = openURL_rtsp(rtspClient, url);

+	fWatchVariableForSyncInterface = 0;

+	rtspClient->sendDescribeCommand(responseHandlerForSyncInterface);

+	env->taskScheduler().doEventLoop(&fWatchVariableForSyncInterface);

+	if (fResultCode == 0)

+	    sdpDescription = fResultString;

+	else

+	    delete[] fResultString;

       } else { // SIP

 	unsigned char desiredAudioType = 0; // PCMU (use 3 for GSM)

 	sipClient = SIPClient::createNew(*env, desiredAudioType, NULL,

@@ -236,8 +244,12 @@ extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {

 	if (rtspClient != NULL) {

 	  // Issue a RTSP "SETUP" command on the chosen subsession:

-	  if (!rtspClient->setupMediaSubsession(*subsession, False,

-						rtsp_transport_tcp)) break;

+	  fWatchVariableForSyncInterface = 0;

+	  rtspClient->sendSetupCommand(*subsession, responseHandlerForSyncInterface, False, rtsp_transport_tcp);

+	  env->taskScheduler().doEventLoop(&fWatchVariableForSyncInterface);

+	  delete[] fResultString;

+	  if (fResultCode != 0) break;

+

 	  if (!strcmp(subsession->mediumName(), "audio"))

 	    audiofound = 1;

 	  if (!strcmp(subsession->mediumName(), "video"))

@@ -248,7 +260,11 @@ extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {

     if (rtspClient != NULL) {

       // Issue a RTSP aggregate "PLAY" command on the whole session:

-      if (!rtspClient->playMediaSession(*mediaSession)) break;

+      fWatchVariableForSyncInterface = 0;

+      rtspClient->sendPlayCommand(*mediaSession, responseHandlerForSyncInterface);

+      env->taskScheduler().doEventLoop(&fWatchVariableForSyncInterface);

+      delete[] fResultString;

+      if (fResultCode != 0) break;

     } else if (sipClient != NULL) {

       sipClient->sendACK(); // to start the stream flowing

     }

@@ -637,7 +653,8 @@ static void teardownRTSPorSIPSession(RTPState* rtpState) {

   MediaSession* mediaSession = rtpState->mediaSession;

   if (mediaSession == NULL) return;

   if (rtpState->rtspClient != NULL) {

-    rtpState->rtspClient->teardownMediaSession(*mediaSession);

+    fWatchVariableForSyncInterface = 0;

+    rtpState->rtspClient->sendTeardownCommand(*mediaSession, NULL);

   } else if (rtpState->sipClient != NULL) {

     rtpState->sipClient->sendBYE();

   }

-- 

1.8.5.2