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Fix #2355: GB28181: Fix play by RTC bug. 4.0.116

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xialixin 2021-05-17 10:33:55 +08:00 • committed by winlin
parent 4639510f1b
commit 7d8bde3cdb
3 changed files with 36 additions and 5 deletions
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2
README.md
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@ -176,6 +176,7 @@ The ports used by SRS:
## V4 changes ## V4 changes
* v4.0, 2021-05-18, Fix [#2355][bug #2355]: GB28181: Fix play by RTC bug. 4.0.116
* v4.0, 2021-05-15, SRT: Build SRT from source by SRS. 4.0.115 * v4.0, 2021-05-15, SRT: Build SRT from source by SRS. 4.0.115
* v4.0, 2021-05-15, Rename SrsConsumer* to SrsLiveConsumer*. 4.0.114 * v4.0, 2021-05-15, Rename SrsConsumer* to SrsLiveConsumer*. 4.0.114
* v4.0, 2021-05-15, Rename SrsRtcStream* to SrsRtcSource*. 4.0.113 * v4.0, 2021-05-15, Rename SrsRtcStream* to SrsRtcSource*. 4.0.113
@ -1923,6 +1924,7 @@ Winlin
[bug #2188]: https://github.com/ossrs/srs/issues/2188 [bug #2188]: https://github.com/ossrs/srs/issues/2188
[bug #1193]: https://github.com/ossrs/srs/issues/1193 [bug #1193]: https://github.com/ossrs/srs/issues/1193
[bug #2304]: https://github.com/ossrs/srs/issues/2304#issuecomment-826009290 [bug #2304]: https://github.com/ossrs/srs/issues/2304#issuecomment-826009290
[bug #2355]: https://github.com/ossrs/srs/issues/2355
[bug #yyyyyyyyyyyyy]: https://github.com/ossrs/srs/issues/yyyyyyyyyyyyy [bug #yyyyyyyyyyyyy]: https://github.com/ossrs/srs/issues/yyyyyyyyyyyyy
[bug #1631]: https://github.com/ossrs/srs/issues/1631 [bug #1631]: https://github.com/ossrs/srs/issues/1631

37
trunk/src/app/srs_app_gb28181.cpp
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@ -51,6 +51,7 @@ using namespace std;
#include <srs_protocol_utility.hpp> #include <srs_protocol_utility.hpp>
#include <srs_protocol_format.hpp> #include <srs_protocol_format.hpp>
#include <srs_app_sip.hpp> #include <srs_app_sip.hpp>
#include <srs_app_rtc_source.hpp>
//#define W_PS_FILE //#define W_PS_FILE
//#define W_VIDEO_FILE //#define W_VIDEO_FILE
@ -1436,10 +1437,38 @@ srs_error_t SrsGb28181RtmpMuxer::initialize(SrsServer *s, SrsRequest* r)
return srs_error_wrap(err, "create source"); return srs_error_wrap(err, "create source");
} }
//TODO: ??? #ifdef SRS_RTC
// if (!source->can_publish(false)) { SrsRtcSource *rtc = NULL;
// return srs_error_new(ERROR_GB28181_SESSION_IS_EXIST, "stream %s busy", req->get_stream_url().c_str()); bool rtc_server_enabled = _srs_config->get_rtc_server_enabled();
// } bool rtc_enabled = _srs_config->get_rtc_enabled(req->vhost);
if (rtc_server_enabled && rtc_enabled) {
if ((err = _srs_rtc_sources->fetch_or_create(req, &rtc)) != srs_success) {
return srs_error_wrap(err, "create source");
}
if (!rtc->can_publish()) {
return srs_error_new(ERROR_RTC_SOURCE_BUSY, "gb28181 rtc stream %s busy", req->get_stream_url().c_str());
}
}
#endif
// Check whether RTMP stream is busy.
if (!source->can_publish(false)) {
return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "gb28181 rtmp: stream %s is busy", req->get_stream_url().c_str());
}
// Bridge to RTC streaming.
#if defined(SRS_RTC) && defined(SRS_FFMPEG_FIT)
if (rtc) {
SrsRtcFromRtmpBridger *bridger = new SrsRtcFromRtmpBridger(rtc);
if ((err = bridger->initialize(req)) != srs_success) {
srs_freep(bridger);
return srs_error_wrap(err, "bridger init");
}
source->set_bridger(bridger);
}
#endif
if ((err = source->on_publish()) != srs_success) { if ((err = source->on_publish()) != srs_success) {
return srs_error_wrap(err, "on publish"); return srs_error_wrap(err, "on publish");

2
trunk/src/core/srs_core_version4.hpp
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@ -26,6 +26,6 @@
#define VERSION_MAJOR 4 #define VERSION_MAJOR 4
#define VERSION_MINOR 0 #define VERSION_MINOR 0
#define VERSION_REVISION 115 #define VERSION_REVISION 116
#endif #endif