[Buildroot] [git commit] mplayer: add patch to fix building with modern versions of live555
Peter Korsgaard
peter at korsgaard.com
Mon Jan 13 21:08:46 UTC 2014
commit: http://git.buildroot.net/buildroot/commit/?id=1a8d62cde9194611de9cb5c564789f9cbf0edf19
branch: http://git.buildroot.net/buildroot/commit/?id=refs/heads/master
Patch taken from: https://raw.github.com/sherpya/mplayer-be/master/patches/mp/0025-mpdemux-live555-async-interface.patch
At the same time rename patches to follow the new convention.
Signed-off-by: Peter Korsgaard <peter at korsgaard.com>
---
...ch => mplayer-0001-disable-install-strip.patch} | 0
...ing.patch => mplayer-0002-fix-cc-parsing.patch} | 0
...ayer-0003-mpdemux-live555-async-interface.patch | 126 ++++++++++++++++++++
3 files changed, 126 insertions(+), 0 deletions(-)
diff --git a/package/mplayer/mplayer-1.1-disable-install-strip.patch b/package/mplayer/mplayer-0001-disable-install-strip.patch
similarity index 100%
rename from package/mplayer/mplayer-1.1-disable-install-strip.patch
rename to package/mplayer/mplayer-0001-disable-install-strip.patch
diff --git a/package/mplayer/mplayer-1.1-fix-cc-parsing.patch b/package/mplayer/mplayer-0002-fix-cc-parsing.patch
similarity index 100%
rename from package/mplayer/mplayer-1.1-fix-cc-parsing.patch
rename to package/mplayer/mplayer-0002-fix-cc-parsing.patch
diff --git a/package/mplayer/mplayer-0003-mpdemux-live555-async-interface.patch b/package/mplayer/mplayer-0003-mpdemux-live555-async-interface.patch
new file mode 100644
index 0000000..5a62a18
--- /dev/null
+++ b/package/mplayer/mplayer-0003-mpdemux-live555-async-interface.patch
@@ -0,0 +1,126 @@
+From d3195ea13f4a9aae546ff996e53681349a1a3cdb Mon Sep 17 00:00:00 2001
+From: sherpya <sherpya at 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 at 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
+
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