From 1a8d62cde9194611de9cb5c564789f9cbf0edf19 Mon Sep 17 00:00:00 2001 From: Peter Korsgaard Date: Mon, 13 Jan 2014 22:08:46 +0100 Subject: [PATCH] mplayer: add patch to fix building with modern versions of live555 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 --- ... mplayer-0001-disable-install-strip.patch} | 0 ...atch => mplayer-0002-fix-cc-parsing.patch} | 0 ...0003-mpdemux-live555-async-interface.patch | 126 ++++++++++++++++++ 3 files changed, 126 insertions(+) rename package/mplayer/{mplayer-1.1-disable-install-strip.patch => mplayer-0001-disable-install-strip.patch} (100%) rename package/mplayer/{mplayer-1.1-fix-cc-parsing.patch => mplayer-0002-fix-cc-parsing.patch} (100%) create mode 100644 package/mplayer/mplayer-0003-mpdemux-live555-async-interface.patch 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 0000000000..5a62a18710 --- /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 +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 +--- + 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 + -- 2.30.2