- old live patch replaced by live555-async patch from mplayer ml (port to async RTSPC...

diff --git a/mplayer-live.patch b/mplayer-live.patch
deleted file mode 100644 (file)
index 7ce9fba..0000000
--- a/mplayer-live.patch
+++ /dev/null
@@ -1,10 +0,0 @@
-+++ MPlayer-1.1/configure.orig 2013-01-28 21:25:18.354520436 +0100
---- MPlayer-1.1/configure      2013-01-30 20:06:23.334350773 +0100
-@@ -6686,6 +6686,7 @@
- echocheck "LIVE555 Streaming Media libraries"
- if test "$_live" = auto  && test "$networking" = yes ; then
-   cat > $TMPCPP << EOF
-+#define RTSPCLIENT_SYNCHRONOUS_INTERFACE 1
- #include <liveMedia.hh>
- #if (LIVEMEDIA_LIBRARY_VERSION_INT < 1141257600)
- #error Please upgrade to version 2006.03.03 or later of the "LIVE555 Streaming Media" libraries - available from <www.live555.com/liveMedia/>

diff --git a/mplayer-live555-async.patch b/mplayer-live555-async.patch
new file mode 100644 (file)
index 0000000..b847d78
--- /dev/null
+++ b/mplayer-live555-async.patch
@@ -0,0 +1,110 @@
+origin: https://patches.libav.org/patch/39102/raw/
+diff --git a/libmpdemux/demux_rtp.cpp b/libmpdemux/demux_rtp.cpp
+index 018327c..4fa7efe 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:
+@@ -126,6 +115,19 @@ int rtsp_transport_http = 0;
+ 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;
+@@ -154,13 +156,19 @@ 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,
+@@ -244,8 +252,12 @@ 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"))
+@@ -256,7 +268,11 @@ 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
+     }
+@@ -645,7 +661,8 @@ 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();
+   }

diff --git a/mplayer.spec b/mplayer.spec
index ee4aa86cda791b4ba7c0df154f9e540858f77e42..6d5a5a34886587deb813fdc849975ad770b1b98f 100644 (file)
--- a/mplayer.spec
+++ b/mplayer.spec
@@ -146,7 +146,7 @@ Patch11:    %{name}-altivec.patch
 Patch12:       %{name}-check-byteswap.patch
 Patch13:       %{name}-visibility-hidden-fix.patch
 Patch14:       %{name}-ffmpeg.patch
-Patch15:       %{name}-live.patch
+Patch15:       %{name}-live555-async.patch
 Patch16:       %{name}-libcdio.patch
 Patch17:       %{name}-gsm.patch
 Patch18:       %{name}-openjpeg.patch
@@ -234,9 +234,7 @@ BuildRequires:      libvpx-devel
 %{?with_x264:BuildRequires:    libx264-devel >= 0.1.3}
 BuildRequires: libxslt-progs
 %{?with_lirc:BuildRequires:    lirc-devel}
-%{?with_live:BuildRequires:    live-devel >= 2:2006.03.03}
-# "synchronous" RTSPClient interface removed finally in 2013.05.30
-%{?with_live:BuildRequires:    live-devel < 2:2013.05.30}
+%{?with_live:BuildRequires:    live-devel >= 2:2011.01.10}
 %{?with_lzo:BuildRequires:     lzo-devel >= 2.0}
 %{?with_nas:BuildRequires:     nas-devel}
 BuildRequires: ncurses-devel