mplayer-live555-async.patch

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