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[packages/trusty/cirros-testvm.git] / cirros-testvm / src-cirros / buildroot-2015.05 / package / mplayer / 0003-mpdemux-live555-async-interface.patch

diff --git a/cirros-testvm/src-cirros/buildroot-2015.05/package/mplayer/0003-mpdemux-live555-async-interface.patch b/cirros-testvm/src-cirros/buildroot-2015.05/package/mplayer/0003-mpdemux-live555-async-interface.patch
new file mode 100644 (file)
index 0000000..5a62a18
--- /dev/null
+++ b/cirros-testvm/src-cirros/buildroot-2015.05/package/mplayer/0003-mpdemux-live555-async-interface.patch
@@ -0,0 +1,126 @@
+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
+