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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
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