package/*: rename patches according to the new policy

Autogenerated from rename-patch.py (http://patchwork.ozlabs.org/patch/403345)

Signed-off-by: Samuel Martin <s.martin49@gmail.com>
Signed-off-by: Peter Korsgaard <peter@korsgaard.com>

1 files changed, 0 insertions, 126 deletions

diff --git a/package/mplayer/mplayer-0003-mpdemux-live555-async-interface.patch b/package/mplayer/mplayer-0003-mpdemux-live555-async-interface.patch
deleted file mode 100644
index 5a62a18710..0000000000
--- a/package/mplayer/mplayer-0003-mpdemux-live555-async-interface.patch
+++ /dev/null
@@ -1,126 +0,0 @@
-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
-