127 lines
5.0 KiB
Diff
127 lines
5.0 KiB
Diff
From d3195ea13f4a9aae546ff996e53681349a1a3cdb Mon Sep 17 00:00:00 2001
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From: sherpya <sherpya@netfarm.it>
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Date: Fri, 14 Jun 2013 05:25:38 +0200
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Subject: [PATCH 25/27] mpdemux: live555 async interface
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From: https://raw.github.com/sherpya/mplayer-be/master/patches/mp/0025-mpdemux-live555-async-interface.patch
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Adjust live555 interface code for modern versions of live555.
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Signed-off-by: Peter Korsgaard <peter@korsgaard.com>
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---
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libmpdemux/demux_rtp.cpp | 51 ++++++++++++++++++++++++++++++++----------------
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2 files changed, 35 insertions(+), 22 deletions(-)
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diff --git a/libmpdemux/demux_rtp.cpp b/libmpdemux/demux_rtp.cpp
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index ad7a7f1..05d06e0 100644
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--- a/libmpdemux/demux_rtp.cpp
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+++ b/libmpdemux/demux_rtp.cpp
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@@ -19,8 +19,6 @@
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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-#define RTSPCLIENT_SYNCHRONOUS_INTERFACE 1
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-
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extern "C" {
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// on MinGW, we must include windows.h before the things it conflicts
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#ifdef __MINGW32__ // with. they are each protected from
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@@ -94,15 +92,6 @@ struct RTPState {
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extern "C" char* network_username;
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extern "C" char* network_password;
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-static char* openURL_rtsp(RTSPClient* client, char const* url) {
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- // If we were given a user name (and optional password), then use them:
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- if (network_username != NULL) {
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- char const* password = network_password == NULL ? "" : network_password;
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- return client->describeWithPassword(url, network_username, password);
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- } else {
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- return client->describeURL(url);
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- }
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-}
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static char* openURL_sip(SIPClient* client, char const* url) {
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// If we were given a user name (and optional password), then use them:
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@@ -118,6 +107,19 @@ static char* openURL_sip(SIPClient* client, char const* url) {
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extern AVCodecContext *avcctx;
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#endif
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+static char fWatchVariableForSyncInterface;
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+static char* fResultString;
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+static int fResultCode;
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+
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+static void responseHandlerForSyncInterface(RTSPClient* rtspClient, int responseCode, char* responseString) {
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+ // Set result values:
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+ fResultCode = responseCode;
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+ fResultString = responseString;
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+
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+ // Signal a break from the event loop (thereby returning from the blocking command):
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+ fWatchVariableForSyncInterface = ~0;
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+}
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+
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extern "C" int audio_id, video_id, dvdsub_id;
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extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
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Boolean success = False;
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@@ -146,13 +148,19 @@ extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
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rtsp_transport_http = demuxer->stream->streaming_ctrl->url->port;
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rtsp_transport_tcp = 1;
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}
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- rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer", rtsp_transport_http);
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+ rtspClient = RTSPClient::createNew(*env, url, verbose, "MPlayer", rtsp_transport_http);
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if (rtspClient == NULL) {
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fprintf(stderr, "Failed to create RTSP client: %s\n",
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env->getResultMsg());
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break;
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}
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- sdpDescription = openURL_rtsp(rtspClient, url);
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+ fWatchVariableForSyncInterface = 0;
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+ rtspClient->sendDescribeCommand(responseHandlerForSyncInterface);
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+ env->taskScheduler().doEventLoop(&fWatchVariableForSyncInterface);
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+ if (fResultCode == 0)
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+ sdpDescription = fResultString;
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+ else
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+ delete[] fResultString;
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} else { // SIP
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unsigned char desiredAudioType = 0; // PCMU (use 3 for GSM)
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sipClient = SIPClient::createNew(*env, desiredAudioType, NULL,
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@@ -236,8 +244,12 @@ extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
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if (rtspClient != NULL) {
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// Issue a RTSP "SETUP" command on the chosen subsession:
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- if (!rtspClient->setupMediaSubsession(*subsession, False,
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- rtsp_transport_tcp)) break;
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+ fWatchVariableForSyncInterface = 0;
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+ rtspClient->sendSetupCommand(*subsession, responseHandlerForSyncInterface, False, rtsp_transport_tcp);
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+ env->taskScheduler().doEventLoop(&fWatchVariableForSyncInterface);
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+ delete[] fResultString;
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+ if (fResultCode != 0) break;
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+
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if (!strcmp(subsession->mediumName(), "audio"))
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audiofound = 1;
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if (!strcmp(subsession->mediumName(), "video"))
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@@ -248,7 +260,11 @@ extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
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if (rtspClient != NULL) {
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// Issue a RTSP aggregate "PLAY" command on the whole session:
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- if (!rtspClient->playMediaSession(*mediaSession)) break;
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+ fWatchVariableForSyncInterface = 0;
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+ rtspClient->sendPlayCommand(*mediaSession, responseHandlerForSyncInterface);
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+ env->taskScheduler().doEventLoop(&fWatchVariableForSyncInterface);
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+ delete[] fResultString;
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+ if (fResultCode != 0) break;
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} else if (sipClient != NULL) {
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sipClient->sendACK(); // to start the stream flowing
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}
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@@ -637,7 +653,8 @@ static void teardownRTSPorSIPSession(RTPState* rtpState) {
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MediaSession* mediaSession = rtpState->mediaSession;
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if (mediaSession == NULL) return;
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if (rtpState->rtspClient != NULL) {
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- rtpState->rtspClient->teardownMediaSession(*mediaSession);
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+ fWatchVariableForSyncInterface = 0;
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+ rtpState->rtspClient->sendTeardownCommand(*mediaSession, NULL);
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} else if (rtpState->sipClient != NULL) {
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rtpState->sipClient->sendBYE();
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}
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--
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1.8.5.2
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