mplayer-live555-async.patch

   1 origin: https://patches.libav.org/patch/39102/raw/
   2 with configure check update added
   3 diff --git a/libmpdemux/demux_rtp.cpp b/libmpdemux/demux_rtp.cpp
   4 index 018327c..4fa7efe 100644
   5 --- a/libmpdemux/demux_rtp.cpp
   6 +++ b/libmpdemux/demux_rtp.cpp
   7 @@ -19,8 +19,6 @@
   8   * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
   9   */
  10
  11 -#define RTSPCLIENT_SYNCHRONOUS_INTERFACE 1
  12 -
  13  extern "C" {
  14  // on MinGW, we must include windows.h before the things it conflicts
  15  #ifdef __MINGW32__    // with.  they are each protected from
  16 @@ -94,15 +92,6 @@ struct RTPState {
  17
  18  extern "C" char* network_username;
  19  extern "C" char* network_password;
  20 -static char* openURL_rtsp(RTSPClient* client, char const* url) {
  21 -  // If we were given a user name (and optional password), then use them:
  22 -  if (network_username != NULL) {
  23 -    char const* password = network_password == NULL ? "" : network_password;
  24 -    return client->describeWithPassword(url, network_username, password);
  25 -  } else {
  26 -    return client->describeURL(url);
  27 -  }
  28 -}
  29
  30  static char* openURL_sip(SIPClient* client, char const* url) {
  31    // If we were given a user name (and optional password), then use them:
  32 @@ -126,6 +115,19 @@ int rtsp_transport_http = 0;
  33  extern AVCodecContext *avcctx;
  34  #endif
  35
  36 +static char fWatchVariableForSyncInterface;
  37 +static char* fResultString;
  38 +static int fResultCode;
  39 +
  40 +static void responseHandlerForSyncInterface(RTSPClient* rtspClient, int responseCode, char* responseString) {
  41 +  // Set result values:
  42 +  fResultCode = responseCode;
  43 +  fResultString = responseString;
  44 +
  45 +  // Signal a break from the event loop (thereby returning from the blocking command):
  46 +  fWatchVariableForSyncInterface = ~0;
  47 +}
  48 +
  49  extern "C" int audio_id, video_id, dvdsub_id;
  50  extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
  51    Boolean success = False;
  52 @@ -154,13 +156,19 @@ extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
  53           rtsp_transport_http = demuxer->stream->streaming_ctrl->url->port;
  54           rtsp_transport_tcp = 1;
  55         }
  56 -       rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer", rtsp_transport_http);
  57 +       rtspClient = RTSPClient::createNew(*env, url, verbose, "MPlayer", rtsp_transport_http);
  58         if (rtspClient == NULL) {
  59           fprintf(stderr, "Failed to create RTSP client: %s\n",
  60                   env->getResultMsg());
  61           break;
  62         }
  63 -       sdpDescription = openURL_rtsp(rtspClient, url);
  64 +       fWatchVariableForSyncInterface = 0;
  65 +       rtspClient->sendDescribeCommand(responseHandlerForSyncInterface);
  66 +       env->taskScheduler().doEventLoop(&fWatchVariableForSyncInterface);
  67 +       if (fResultCode == 0)
  68 +           sdpDescription = fResultString;
  69 +       else
  70 +           delete[] fResultString;
  71        } else { // SIP
  72         unsigned char desiredAudioType = 0; // PCMU (use 3 for GSM)
  73         sipClient = SIPClient::createNew(*env, desiredAudioType, NULL,
  74 @@ -244,8 +252,12 @@ extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
  75
  76         if (rtspClient != NULL) {
  77           // Issue a RTSP "SETUP" command on the chosen subsession:
  78 -         if (!rtspClient->setupMediaSubsession(*subsession, False,
  79 -                                               rtsp_transport_tcp)) break;
  80 +         fWatchVariableForSyncInterface = 0;
  81 +         rtspClient->sendSetupCommand(*subsession, responseHandlerForSyncInterface, False, rtsp_transport_tcp);
  82 +         env->taskScheduler().doEventLoop(&fWatchVariableForSyncInterface);
  83 +         delete[] fResultString;
  84 +         if (fResultCode != 0) break;
  85 +
  86           if (!strcmp(subsession->mediumName(), "audio"))
  87             audiofound = 1;
  88           if (!strcmp(subsession->mediumName(), "video"))
  89 @@ -256,7 +268,11 @@ extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
  90
  91      if (rtspClient != NULL) {
  92        // Issue a RTSP aggregate "PLAY" command on the whole session:
  93 -      if (!rtspClient->playMediaSession(*mediaSession)) break;
  94 +      fWatchVariableForSyncInterface = 0;
  95 +      rtspClient->sendPlayCommand(*mediaSession, responseHandlerForSyncInterface);
  96 +      env->taskScheduler().doEventLoop(&fWatchVariableForSyncInterface);
  97 +      delete[] fResultString;
  98 +      if (fResultCode != 0) break;
  99      } else if (sipClient != NULL) {
 100        sipClient->sendACK(); // to start the stream flowing
 101      }
 102 @@ -645,7 +661,8 @@ static void teardownRTSPorSIPSession(RTPState* rtpState) {
 103    MediaSession* mediaSession = rtpState->mediaSession;
 104    if (mediaSession == NULL) return;
 105    if (rtpState->rtspClient != NULL) {
 106 -    rtpState->rtspClient->teardownMediaSession(*mediaSession);
 107 +    fWatchVariableForSyncInterface = 0;
 108 +    rtpState->rtspClient->sendTeardownCommand(*mediaSession, NULL);
 109    } else if (rtpState->sipClient != NULL) {
 110      rtpState->sipClient->sendBYE();
 111    }