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