package/mplayer/0002-mpdemux-live555-async-interface.patch

   1 From d3195ea13f4a9aae546ff996e53681349a1a3cdb Mon Sep 17 00:00:00 2001
   2 From: sherpya <sherpya@netfarm.it>
   3 Date: Fri, 14 Jun 2013 05:25:38 +0200
   4 Subject: [PATCH 25/27] mpdemux: live555 async interface
   5
   6 From: https://raw.github.com/sherpya/mplayer-be/master/patches/mp/0025-mpdemux-live555-async-interface.patch
   7
   8 Adjust live555 interface code for modern versions of live555.
   9
  10 Signed-off-by: Peter Korsgaard <peter@korsgaard.com>
  11 ---
  12  libmpdemux/demux_rtp.cpp | 51 ++++++++++++++++++++++++++++++++----------------
  13  2 files changed, 35 insertions(+), 22 deletions(-)
  14
  15 diff --git a/libmpdemux/demux_rtp.cpp b/libmpdemux/demux_rtp.cpp
  16 index ad7a7f1..05d06e0 100644
  17 --- a/libmpdemux/demux_rtp.cpp
  18 +++ b/libmpdemux/demux_rtp.cpp
  19 @@ -19,8 +19,6 @@
  20   * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  21   */
  22
  23 -#define RTSPCLIENT_SYNCHRONOUS_INTERFACE 1
  24 -
  25  extern "C" {
  26  // on MinGW, we must include windows.h before the things it conflicts
  27  #ifdef __MINGW32__    // with.  they are each protected from
  28 @@ -94,15 +92,6 @@ struct RTPState {
  29
  30  extern "C" char* network_username;
  31  extern "C" char* network_password;
  32 -static char* openURL_rtsp(RTSPClient* client, char const* url) {
  33 -  // If we were given a user name (and optional password), then use them:
  34 -  if (network_username != NULL) {
  35 -    char const* password = network_password == NULL ? "" : network_password;
  36 -    return client->describeWithPassword(url, network_username, password);
  37 -  } else {
  38 -    return client->describeURL(url);
  39 -  }
  40 -}
  41
  42  static char* openURL_sip(SIPClient* client, char const* url) {
  43    // If we were given a user name (and optional password), then use them:
  44 @@ -118,6 +107,19 @@ static char* openURL_sip(SIPClient* client, char const* url) {
  45  extern AVCodecContext *avcctx;
  46  #endif
  47
  48 +static char fWatchVariableForSyncInterface;
  49 +static char* fResultString;
  50 +static int fResultCode;
  51 +
  52 +static void responseHandlerForSyncInterface(RTSPClient* rtspClient, int responseCode, char* responseString) {
  53 +  // Set result values:
  54 +  fResultCode = responseCode;
  55 +  fResultString = responseString;
  56 +
  57 +  // Signal a break from the event loop (thereby returning from the blocking command):
  58 +  fWatchVariableForSyncInterface = ~0;
  59 +}
  60 +
  61  extern "C" int audio_id, video_id, dvdsub_id;
  62  extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
  63    Boolean success = False;
  64 @@ -146,13 +148,19 @@ extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
  65           rtsp_transport_http = demuxer->stream->streaming_ctrl->url->port;
  66           rtsp_transport_tcp = 1;
  67         }
  68 -       rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer", rtsp_transport_http);
  69 +       rtspClient = RTSPClient::createNew(*env, url, verbose, "MPlayer", rtsp_transport_http);
  70         if (rtspClient == NULL) {
  71           fprintf(stderr, "Failed to create RTSP client: %s\n",
  72                   env->getResultMsg());
  73           break;
  74         }
  75 -       sdpDescription = openURL_rtsp(rtspClient, url);
  76 +       fWatchVariableForSyncInterface = 0;
  77 +       rtspClient->sendDescribeCommand(responseHandlerForSyncInterface);
  78 +       env->taskScheduler().doEventLoop(&fWatchVariableForSyncInterface);
  79 +       if (fResultCode == 0)
  80 +           sdpDescription = fResultString;
  81 +       else
  82 +           delete[] fResultString;
  83        } else { // SIP
  84         unsigned char desiredAudioType = 0; // PCMU (use 3 for GSM)
  85         sipClient = SIPClient::createNew(*env, desiredAudioType, NULL,
  86 @@ -236,8 +244,12 @@ extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
  87
  88         if (rtspClient != NULL) {
  89           // Issue a RTSP "SETUP" command on the chosen subsession:
  90 -         if (!rtspClient->setupMediaSubsession(*subsession, False,
  91 -                                               rtsp_transport_tcp)) break;
  92 +         fWatchVariableForSyncInterface = 0;
  93 +         rtspClient->sendSetupCommand(*subsession, responseHandlerForSyncInterface, False, rtsp_transport_tcp);
  94 +         env->taskScheduler().doEventLoop(&fWatchVariableForSyncInterface);
  95 +         delete[] fResultString;
  96 +         if (fResultCode != 0) break;
  97 +
  98           if (!strcmp(subsession->mediumName(), "audio"))
  99             audiofound = 1;
 100           if (!strcmp(subsession->mediumName(), "video"))
 101 @@ -248,7 +260,11 @@ extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
 102
 103      if (rtspClient != NULL) {
 104        // Issue a RTSP aggregate "PLAY" command on the whole session:
 105 -      if (!rtspClient->playMediaSession(*mediaSession)) break;
 106 +      fWatchVariableForSyncInterface = 0;
 107 +      rtspClient->sendPlayCommand(*mediaSession, responseHandlerForSyncInterface);
 108 +      env->taskScheduler().doEventLoop(&fWatchVariableForSyncInterface);
 109 +      delete[] fResultString;
 110 +      if (fResultCode != 0) break;
 111      } else if (sipClient != NULL) {
 112        sipClient->sendACK(); // to start the stream flowing
 113      }
 114 @@ -637,7 +653,8 @@ static void teardownRTSPorSIPSession(RTPState* rtpState) {
 115    MediaSession* mediaSession = rtpState->mediaSession;
 116    if (mediaSession == NULL) return;
 117    if (rtpState->rtspClient != NULL) {
 118 -    rtpState->rtspClient->teardownMediaSession(*mediaSession);
 119 +    fWatchVariableForSyncInterface = 0;
 120 +    rtpState->rtspClient->sendTeardownCommand(*mediaSession, NULL);
 121    } else if (rtpState->sipClient != NULL) {
 122      rtpState->sipClient->sendBYE();
 123    }
 124 --
 125 1.8.5.2
 126